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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 library 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 of the License, or (at your option) any later version.
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 *
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 * This library 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 this library; if not, write to the Free Software
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 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
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 *
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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 distorions expecially
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 * 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>
35

    
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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"
40

    
41
#ifdef CONFIG_MPEGAUDIO_HP
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#define USE_HIGHPRECISION
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#endif
44

    
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#include "mpegaudio.h"
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#include "qdm2data.h"
48

    
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#undef NDEBUG
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#include <assert.h>
51

    
52

    
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#define SOFTCLIP_THRESHOLD 27600
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#define HARDCLIP_THRESHOLD 35716
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56

    
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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))
69

    
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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)])
75

    
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#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
77

    
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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");
80

    
81
#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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84

    
85
typedef int8_t sb_int8_array[2][30][64];
86

    
87
/**
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 * Subpacket
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 */
90
typedef struct {
91
    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;
95

    
96
/**
97
 * A node in subpacket list
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 */
99
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;
103

    
104
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;
112
    short time_index;
113
    short cutoff;
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} FFTTone;
115

    
116
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;
121
    uint8_t phase;
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} FFTCoefficient;
123

    
124
typedef struct {
125
    float re;
126
    float im;
127
} QDM2Complex;
128

    
129
typedef struct {
130
    QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
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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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135
/**
136
 * QDM2 decoder context
137
 */
138
typedef struct {
139
    /// Parameters from codec header, do not change during playback
140
    int nb_channels;         ///< number of channels
141
    int channels;            ///< number of channels
142
    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
157
    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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164
    /// FFT and tones
165
    FFTTone fft_tones[1000];
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    int fft_tone_start;
167
    int fft_tone_end;
168
    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
178
    uint8_t *compressed_data;
179
    int compressed_size;
180
    float output_buffer[1024];
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182
    /// Synthesis filter
183
    MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
184
    int synth_buf_offset[MPA_MAX_CHANNELS];
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    int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
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    /// Mixed temporary data used in decoding
188
    float tone_level[MPA_MAX_CHANNELS][30][64];
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    int8_t coding_method[MPA_MAX_CHANNELS][30][64];
190
    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
199
    int has_errors;         ///< packet have errors
200
    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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203
    int sub_packet;
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    int noise_idx; ///< Index for dithering noise table
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} QDM2Context;
206

    
207

    
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static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
209

    
210
static VLC vlc_tab_level;
211
static VLC vlc_tab_diff;
212
static VLC vlc_tab_run;
213
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;
220
static VLC vlc_tab_type30;
221
static VLC vlc_tab_type34;
222
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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230
static MPA_INT mpa_window[512] __attribute__((aligned(16)));
231

    
232

    
233
static void softclip_table_init() {
234
    int i;
235
    double dfl = SOFTCLIP_THRESHOLD - 32767;
236
    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);
239
}
240

    
241

    
242
// random generated table
243
static void rnd_table_init() {
244
    int i,j;
245
    uint32_t ldw,hdw;
246
    uint64_t tmp64_1;
247
    uint64_t random_seed = 0;
248
    float delta = 1.0 / 16384.0;
249
    for(i = 0; i < 4096 ;i++) {
250
        random_seed = random_seed * 214013 + 2531011;
251
        noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
252
    }
253

    
254
    for (i = 0; i < 256 ;i++) {
255
        random_seed = 81;
256
        ldw = i;
257
        for (j = 0; j < 5 ;j++) {
258
            random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
259
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
260
            tmp64_1 = (random_seed * 0x55555556);
261
            hdw = (uint32_t)(tmp64_1 >> 32);
262
            random_seed = (uint64_t)(hdw + (ldw >> 31));
263
        }
264
    }
265
    for (i = 0; i < 128 ;i++) {
266
        random_seed = 25;
267
        ldw = i;
268
        for (j = 0; j < 3 ;j++) {
269
            random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
270
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
271
            tmp64_1 = (random_seed * 0x66666667);
272
            hdw = (uint32_t)(tmp64_1 >> 33);
273
            random_seed = hdw + (ldw >> 31);
274
        }
275
    }
276
}
277

    
278

    
279
static void init_noise_samples() {
280
    int i;
281
    int random_seed = 0;
282
    float delta = 1.0 / 16384.0;
283
    for (i = 0; i < 128;i++) {
284
        random_seed = random_seed * 214013 + 2531011;
285
        noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
286
    }
287
}
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289

    
290
static void qdm2_init_vlc()
291
{
292
    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);
295

    
296
    init_vlc (&vlc_tab_diff, 8, 37,
297
        vlc_tab_diff_huffbits, 1, 1,
298
        vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
299

    
300
    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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304
    init_vlc (&fft_level_exp_alt_vlc, 8, 28,
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        fft_level_exp_alt_huffbits, 1, 1,
306
        fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
307

    
308
    init_vlc (&fft_level_exp_vlc, 8, 20,
309
        fft_level_exp_huffbits, 1, 1,
310
        fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
311

    
312
    init_vlc (&fft_stereo_exp_vlc, 6, 7,
313
        fft_stereo_exp_huffbits, 1, 1,
314
        fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
315

    
316
    init_vlc (&fft_stereo_phase_vlc, 6, 9,
317
        fft_stereo_phase_huffbits, 1, 1,
318
        fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
319

    
320
    init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
321
        vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
322
        vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
323

    
324
    init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
325
        vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
326
        vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
327

    
328
    init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
329
        vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
330
        vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
331

    
332
    init_vlc (&vlc_tab_type30, 6, 9,
333
        vlc_tab_type30_huffbits, 1, 1,
334
        vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
335

    
336
    init_vlc (&vlc_tab_type34, 5, 10,
337
        vlc_tab_type34_huffbits, 1, 1,
338
        vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
339

    
340
    init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
341
        vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
342
        vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
343

    
344
    init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
345
        vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
346
        vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
347

    
348
    init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
349
        vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
350
        vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
351

    
352
    init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
353
        vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
354
        vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
355

    
356
    init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
357
        vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
358
        vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
359
}
360

    
361

    
362
/* for floating point to fixed point conversion */
363
static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
364

    
365

    
366
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
367
{
368
    int value;
369

    
370
    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
371

    
372
    /* stage-2, 3 bits exponent escape sequence */
373
    if (value-- == 0)
374
        value = get_bits (gb, get_bits (gb, 3) + 1);
375

    
376
    /* stage-3, optional */
377
    if (flag) {
378
        int tmp = vlc_stage3_values[value];
379

    
380
        if ((value & ~3) > 0)
381
            tmp += get_bits (gb, (value >> 2));
382
        value = tmp;
383
    }
384

    
385
    return value;
386
}
387

    
388

    
389
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
390
{
391
    int value = qdm2_get_vlc (gb, vlc, 0, depth);
392

    
393
    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
394
}
395

    
396

    
397
/**
398
 * QDM2 checksum
399
 *
400
 * @param data      pointer to data to be checksum'ed
401
 * @param length    data length
402
 * @param value     checksum value
403
 *
404
 * @return          0 if checksum is ok
405
 */
406
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
407
    int i;
408

    
409
    for (i=0; i < length; i++)
410
        value -= data[i];
411

    
412
    return (uint16_t)(value & 0xffff);
413
}
414

    
415

    
416
/**
417
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer
418
 *
419
 * @param gb            bitreader context
420
 * @param sub_packet    packet under analysis
421
 */
422
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
423
{
424
    sub_packet->type = get_bits (gb, 8);
425

    
426
    if (sub_packet->type == 0) {
427
        sub_packet->size = 0;
428
        sub_packet->data = NULL;
429
    } else {
430
        sub_packet->size = get_bits (gb, 8);
431

    
432
      if (sub_packet->type & 0x80) {
433
          sub_packet->size <<= 8;
434
          sub_packet->size  |= get_bits (gb, 8);
435
          sub_packet->type  &= 0x7f;
436
      }
437

    
438
      if (sub_packet->type == 0x7f)
439
          sub_packet->type |= (get_bits (gb, 8) << 8);
440

    
441
      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
442
    }
443

    
444
    av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n",
445
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
446
}
447

    
448

    
449
/**
450
 * Return node pointer to first packet of requested type in list
451
 *
452
 * @param list    list of subpacket to be scanned
453
 * @param type    type of searched subpacket
454
 * @return        node pointer for subpacket if found, else NULL
455
 */
456
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
457
{
458
    while (list != NULL && list->packet != NULL) {
459
        if (list->packet->type == type)
460
            return list;
461
        list = list->next;
462
    }
463
    return NULL;
464
}
465

    
466

    
467
/**
468
 * Replaces 8 elements with their average value
469
 * Called by qdm2_decode_superblock before starting subblocks decoding
470
 *
471
 * @param q       context
472
 */
473
static void average_quantized_coeffs (QDM2Context *q)
474
{
475
    int i, j, n, ch, sum;
476

    
477
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
478

    
479
    for (ch = 0; ch < q->nb_channels; ch++)
480
        for (i = 0; i < n; i++) {
481
            sum = 0;
482

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

    
486
            sum /= 8;
487
            if (sum > 0)
488
                sum--;
489

    
490
            for (j=0; j < 8; j++)
491
                q->quantized_coeffs[ch][i][j] = sum;
492
        }
493
}
494

    
495

    
496
/**
497
 * Build subband samples with noise weighted by q->tone_level
498
 * Called by synthfilt_build_sb_samples
499
 *
500
 * @param q     context
501
 * @param sb    subband index
502
 */
503
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
504
{
505
    int ch, j;
506

    
507
    FIX_NOISE_IDX(q->noise_idx);
508

    
509
    if (!q->nb_channels)
510
        return;
511

    
512
    for (ch = 0; ch < q->nb_channels; ch++)
513
        for (j = 0; j < 64; j++) {
514
            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);
515
            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);
516
        }
517
}
518

    
519

    
520
/**
521
 * Called while processing data from subpackets 11 and 12
522
 * Used after making changes to coding_method array
523
 *
524
 * @param sb               subband index
525
 * @param channels         number of channels
526
 * @param coding_method    q->coding_method[0][0][0]
527
 */
528
 void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
529
{
530
    int j,k;
531
    int ch;
532
    int run, case_val;
533
    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};
534

    
535
    for (ch = 0; ch < channels; ch++) {
536
        for (j = 0; j < 64; ) {
537
            if((coding_method[ch][sb][j] - 8) > 22) {
538
                run = 1;
539
                case_val = 8;
540
            } else {
541
                switch (switchtable[coding_method[ch][sb][j]]) {
542
                    case 0: run = 10; case_val = 10; break;
543
                    case 1: run = 1; case_val = 16; break;
544
                    case 2: run = 5; case_val = 24; break;
545
                    case 3: run = 3; case_val = 30; break;
546
                    case 4: run = 1; case_val = 30; break;
547
                    case 5: run = 1; case_val = 8; break;
548
                    default: run = 1; case_val = 8; break;
549
                }
550
            }
551
            for (k = 0; k < run; k++)
552
                if (j + k < 128)
553
                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
554
                        if (k > 0) {
555
                           SAMPLES_NEEDED
556
                            //not debugged, almost never used
557
                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
558
                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
559
                        }
560
            j += run;
561
        }
562
    }
563
}
564

    
565

    
566
/**
567
 * Related to synthesis filter
568
 * Called by process_subpacket_10
569
 *
570
 * @param q       context
571
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
572
 */
573
static void fill_tone_level_array (QDM2Context *q, int flag)
574
{
575
    int i, sb, ch, sb_used;
576
    int tmp, tab;
577

    
578
    // This should never happen
579
    if (q->nb_channels <= 0)
580
        return;
581

    
582
    for (ch = 0; ch < q->nb_channels; ch++)
583
        for (sb = 0; sb < 30; sb++)
584
            for (i = 0; i < 8; i++) {
585
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
586
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
587
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
588
                else
589
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
590
                if(tmp < 0)
591
                    tmp += 0xff;
592
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
593
            }
594

    
595
    sb_used = QDM2_SB_USED(q->sub_sampling);
596

    
597
    if ((q->superblocktype_2_3 != 0) && !flag) {
598
        for (sb = 0; sb < sb_used; sb++)
599
            for (ch = 0; ch < q->nb_channels; ch++)
600
                for (i = 0; i < 64; i++) {
601
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
602
                    if (q->tone_level_idx[ch][sb][i] < 0)
603
                        q->tone_level[ch][sb][i] = 0;
604
                    else
605
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
606
                }
607
    } else {
608
        tab = q->superblocktype_2_3 ? 0 : 1;
609
        for (sb = 0; sb < sb_used; sb++) {
610
            if ((sb >= 4) && (sb <= 23)) {
611
                for (ch = 0; ch < q->nb_channels; ch++)
612
                    for (i = 0; i < 64; i++) {
613
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
614
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
615
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
616
                              q->tone_level_idx_hi2[ch][sb - 4];
617
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
618
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
619
                            q->tone_level[ch][sb][i] = 0;
620
                        else
621
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
622
                }
623
            } else {
624
                if (sb > 4) {
625
                    for (ch = 0; ch < q->nb_channels; ch++)
626
                        for (i = 0; i < 64; i++) {
627
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
628
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
629
                                  q->tone_level_idx_hi2[ch][sb - 4];
630
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
631
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
632
                                q->tone_level[ch][sb][i] = 0;
633
                            else
634
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
635
                    }
636
                } else {
637
                    for (ch = 0; ch < q->nb_channels; ch++)
638
                        for (i = 0; i < 64; i++) {
639
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
640
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
641
                                q->tone_level[ch][sb][i] = 0;
642
                            else
643
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
644
                        }
645
                }
646
            }
647
        }
648
    }
649

    
650
    return;
651
}
652

    
653

    
654
/**
655
 * Related to synthesis filter
656
 * Called by process_subpacket_11
657
 * c is built with data from subpacket 11
658
 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
659
 *
660
 * @param tone_level_idx
661
 * @param tone_level_idx_temp
662
 * @param coding_method        q->coding_method[0][0][0]
663
 * @param nb_channels          number of channels
664
 * @param c                    coming from subpacket 11, passed as 8*c
665
 * @param superblocktype_2_3   flag based on superblock packet type
666
 * @param cm_table_select      q->cm_table_select
667
 */
668
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
669
                sb_int8_array coding_method, int nb_channels,
670
                int c, int superblocktype_2_3, int cm_table_select)
671
{
672
    int ch, sb, j;
673
    int tmp, acc, esp_40, comp;
674
    int add1, add2, add3, add4;
675
    int64_t multres;
676

    
677
    // This should never happen
678
    if (nb_channels <= 0)
679
        return;
680

    
681
    if (!superblocktype_2_3) {
682
        /* This case is untested, no samples available */
683
        SAMPLES_NEEDED
684
        for (ch = 0; ch < nb_channels; ch++)
685
            for (sb = 0; sb < 30; sb++) {
686
                for (j = 1; j < 64; j++) {
687
                    add1 = tone_level_idx[ch][sb][j] - 10;
688
                    if (add1 < 0)
689
                        add1 = 0;
690
                    add2 = add3 = add4 = 0;
691
                    if (sb > 1) {
692
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
693
                        if (add2 < 0)
694
                            add2 = 0;
695
                    }
696
                    if (sb > 0) {
697
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
698
                        if (add3 < 0)
699
                            add3 = 0;
700
                    }
701
                    if (sb < 29) {
702
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
703
                        if (add4 < 0)
704
                            add4 = 0;
705
                    }
706
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
707
                    if (tmp < 0)
708
                        tmp = 0;
709
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
710
                }
711
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
712
            }
713
            acc = 0;
714
            for (ch = 0; ch < nb_channels; ch++)
715
                for (sb = 0; sb < 30; sb++)
716
                    for (j = 0; j < 64; j++)
717
                        acc += tone_level_idx_temp[ch][sb][j];
718
            if (acc)
719
                tmp = c * 256 / (acc & 0xffff);
720
            multres = 0x66666667 * (acc * 10);
721
            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
722
            for (ch = 0;  ch < nb_channels; ch++)
723
                for (sb = 0; sb < 30; sb++)
724
                    for (j = 0; j < 64; j++) {
725
                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
726
                        if (comp < 0)
727
                            comp += 0xff;
728
                        comp /= 256; // signed shift
729
                        switch(sb) {
730
                            case 0:
731
                                if (comp < 30)
732
                                    comp = 30;
733
                                comp += 15;
734
                                break;
735
                            case 1:
736
                                if (comp < 24)
737
                                    comp = 24;
738
                                comp += 10;
739
                                break;
740
                            case 2:
741
                            case 3:
742
                            case 4:
743
                                if (comp < 16)
744
                                    comp = 16;
745
                        }
746
                        if (comp <= 5)
747
                            tmp = 0;
748
                        else if (comp <= 10)
749
                            tmp = 10;
750
                        else if (comp <= 16)
751
                            tmp = 16;
752
                        else if (comp <= 24)
753
                            tmp = -1;
754
                        else
755
                            tmp = 0;
756
                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
757
                    }
758
            for (sb = 0; sb < 30; sb++)
759
                fix_coding_method_array(sb, nb_channels, coding_method);
760
            for (ch = 0; ch < nb_channels; ch++)
761
                for (sb = 0; sb < 30; sb++)
762
                    for (j = 0; j < 64; j++)
763
                        if (sb >= 10) {
764
                            if (coding_method[ch][sb][j] < 10)
765
                                coding_method[ch][sb][j] = 10;
766
                        } else {
767
                            if (sb >= 2) {
768
                                if (coding_method[ch][sb][j] < 16)
769
                                    coding_method[ch][sb][j] = 16;
770
                            } else {
771
                                if (coding_method[ch][sb][j] < 30)
772
                                    coding_method[ch][sb][j] = 30;
773
                            }
774
                        }
775
    } else { // superblocktype_2_3 != 0
776
        for (ch = 0; ch < nb_channels; ch++)
777
            for (sb = 0; sb < 30; sb++)
778
                for (j = 0; j < 64; j++)
779
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
780
    }
781

    
782
    return;
783
}
784

    
785

    
786
/**
787
 *
788
 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
789
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
790
 *
791
 * @param q         context
792
 * @param gb        bitreader context
793
 * @param length    packet length in bit
794
 * @param sb_min    lower subband processed (sb_min included)
795
 * @param sb_max    higher subband processed (sb_max excluded)
796
 */
797
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
798
{
799
    int sb, j, k, n, ch, run, channels;
800
    int joined_stereo, zero_encoding, chs;
801
    int type34_first;
802
    float type34_div = 0;
803
    float type34_predictor;
804
    float samples[10], sign_bits[16];
805

    
806
    if (length == 0) {
807
        // If no data use noise
808
        for (sb=sb_min; sb < sb_max; sb++)
809
            build_sb_samples_from_noise (q, sb);
810

    
811
        return;
812
    }
813

    
814
    for (sb = sb_min; sb < sb_max; sb++) {
815
        FIX_NOISE_IDX(q->noise_idx);
816

    
817
        channels = q->nb_channels;
818

    
819
        if (q->nb_channels <= 1 || sb < 12)
820
            joined_stereo = 0;
821
        else if (sb >= 24)
822
            joined_stereo = 1;
823
        else
824
            joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
825

    
826
        if (joined_stereo) {
827
            if (BITS_LEFT(length,gb) >= 16)
828
                for (j = 0; j < 16; j++)
829
                    sign_bits[j] = get_bits1 (gb);
830

    
831
            for (j = 0; j < 64; j++)
832
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
833
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
834

    
835
            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
836
            channels = 1;
837
        }
838

    
839
        for (ch = 0; ch < channels; ch++) {
840
            zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
841
            type34_predictor = 0.0;
842
            type34_first = 1;
843

    
844
            for (j = 0; j < 128; ) {
845
                switch (q->coding_method[ch][sb][j / 2]) {
846
                    case 8:
847
                        if (BITS_LEFT(length,gb) >= 10) {
848
                            if (zero_encoding) {
849
                                for (k = 0; k < 5; k++) {
850
                                    if ((j + 2 * k) >= 128)
851
                                        break;
852
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
853
                                }
854
                            } else {
855
                                n = get_bits(gb, 8);
856
                                for (k = 0; k < 5; k++)
857
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
858
                            }
859
                            for (k = 0; k < 5; k++)
860
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
861
                        } else {
862
                            for (k = 0; k < 10; k++)
863
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
864
                        }
865
                        run = 10;
866
                        break;
867

    
868
                    case 10:
869
                        if (BITS_LEFT(length,gb) >= 1) {
870
                            float f = 0.81;
871

    
872
                            if (get_bits1(gb))
873
                                f = -f;
874
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
875
                            samples[0] = f;
876
                        } else {
877
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
878
                        }
879
                        run = 1;
880
                        break;
881

    
882
                    case 16:
883
                        if (BITS_LEFT(length,gb) >= 10) {
884
                            if (zero_encoding) {
885
                                for (k = 0; k < 5; k++) {
886
                                    if ((j + k) >= 128)
887
                                        break;
888
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
889
                                }
890
                            } else {
891
                                n = get_bits (gb, 8);
892
                                for (k = 0; k < 5; k++)
893
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
894
                            }
895
                        } else {
896
                            for (k = 0; k < 5; k++)
897
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
898
                        }
899
                        run = 5;
900
                        break;
901

    
902
                    case 24:
903
                        if (BITS_LEFT(length,gb) >= 7) {
904
                            n = get_bits(gb, 7);
905
                            for (k = 0; k < 3; k++)
906
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
907
                        } else {
908
                            for (k = 0; k < 3; k++)
909
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
910
                        }
911
                        run = 3;
912
                        break;
913

    
914
                    case 30:
915
                        if (BITS_LEFT(length,gb) >= 4)
916
                            samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
917
                        else
918
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
919

    
920
                        run = 1;
921
                        break;
922

    
923
                    case 34:
924
                        if (BITS_LEFT(length,gb) >= 7) {
925
                            if (type34_first) {
926
                                type34_div = (float)(1 << get_bits(gb, 2));
927
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
928
                                type34_predictor = samples[0];
929
                                type34_first = 0;
930
                            } else {
931
                                samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
932
                                type34_predictor = samples[0];
933
                            }
934
                        } else {
935
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
936
                        }
937
                        run = 1;
938
                        break;
939

    
940
                    default:
941
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
942
                        run = 1;
943
                        break;
944
                }
945

    
946
                if (joined_stereo) {
947
                    float tmp[10][MPA_MAX_CHANNELS];
948

    
949
                    for (k = 0; k < run; k++) {
950
                        tmp[k][0] = samples[k];
951
                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
952
                    }
953
                    for (chs = 0; chs < q->nb_channels; chs++)
954
                        for (k = 0; k < run; k++)
955
                            if ((j + k) < 128)
956
                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
957
                } else {
958
                    for (k = 0; k < run; k++)
959
                        if ((j + k) < 128)
960
                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
961
                }
962

    
963
                j += run;
964
            } // j loop
965
        } // channel loop
966
    } // subband loop
967
}
968

    
969

    
970
/**
971
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0])
972
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
973
 * same VLC tables as process_subpacket_9 are used
974
 *
975
 * @param q         context
976
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
977
 * @param gb        bitreader context
978
 * @param length    packet length in bit
979
 */
980
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
981
{
982
    int i, k, run, level, diff;
983

    
984
    if (BITS_LEFT(length,gb) < 16)
985
        return;
986
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
987

    
988
    quantized_coeffs[0] = level;
989

    
990
    for (i = 0; i < 7; ) {
991
        if (BITS_LEFT(length,gb) < 16)
992
            break;
993
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
994

    
995
        if (BITS_LEFT(length,gb) < 16)
996
            break;
997
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
998

    
999
        for (k = 1; k <= run; k++)
1000
            quantized_coeffs[i + k] = (level + ((k * diff) / run));
1001

    
1002
        level += diff;
1003
        i += run;
1004
    }
1005
}
1006

    
1007

    
1008
/**
1009
 * Related to synthesis filter, process data from packet 10
1010
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1011
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1012
 *
1013
 * @param q         context
1014
 * @param gb        bitreader context
1015
 * @param length    packet length in bit
1016
 */
1017
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1018
{
1019
    int sb, j, k, n, ch;
1020

    
1021
    for (ch = 0; ch < q->nb_channels; ch++) {
1022
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1023

    
1024
        if (BITS_LEFT(length,gb) < 16) {
1025
            memset(q->quantized_coeffs[ch][0], 0, 8);
1026
            break;
1027
        }
1028
    }
1029

    
1030
    n = q->sub_sampling + 1;
1031

    
1032
    for (sb = 0; sb < n; sb++)
1033
        for (ch = 0; ch < q->nb_channels; ch++)
1034
            for (j = 0; j < 8; j++) {
1035
                if (BITS_LEFT(length,gb) < 1)
1036
                    break;
1037
                if (get_bits1(gb)) {
1038
                    for (k=0; k < 8; k++) {
1039
                        if (BITS_LEFT(length,gb) < 16)
1040
                            break;
1041
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1042
                    }
1043
                } else {
1044
                    for (k=0; k < 8; k++)
1045
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1046
                }
1047
            }
1048

    
1049
    n = QDM2_SB_USED(q->sub_sampling) - 4;
1050

    
1051
    for (sb = 0; sb < n; sb++)
1052
        for (ch = 0; ch < q->nb_channels; ch++) {
1053
            if (BITS_LEFT(length,gb) < 16)
1054
                break;
1055
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1056
            if (sb > 19)
1057
                q->tone_level_idx_hi2[ch][sb] -= 16;
1058
            else
1059
                for (j = 0; j < 8; j++)
1060
                    q->tone_level_idx_mid[ch][sb][j] = -16;
1061
        }
1062

    
1063
    n = QDM2_SB_USED(q->sub_sampling) - 5;
1064

    
1065
    for (sb = 0; sb < n; sb++)
1066
        for (ch = 0; ch < q->nb_channels; ch++)
1067
            for (j = 0; j < 8; j++) {
1068
                if (BITS_LEFT(length,gb) < 16)
1069
                    break;
1070
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1071
            }
1072
}
1073

    
1074
/**
1075
 * Process subpacket 9, init quantized_coeffs with data from it
1076
 *
1077
 * @param q       context
1078
 * @param node    pointer to node with packet
1079
 */
1080
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1081
{
1082
    GetBitContext gb;
1083
    int i, j, k, n, ch, run, level, diff;
1084

    
1085
    init_get_bits(&gb, node->packet->data, node->packet->size*8);
1086

    
1087
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1088

    
1089
    for (i = 1; i < n; i++)
1090
        for (ch=0; ch < q->nb_channels; ch++) {
1091
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1092
            q->quantized_coeffs[ch][i][0] = level;
1093

    
1094
            for (j = 0; j < (8 - 1); ) {
1095
                run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1096
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1097

    
1098
                for (k = 1; k <= run; k++)
1099
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1100

    
1101
                level += diff;
1102
                j += run;
1103
            }
1104
        }
1105

    
1106
    for (ch = 0; ch < q->nb_channels; ch++)
1107
        for (i = 0; i < 8; i++)
1108
            q->quantized_coeffs[ch][0][i] = 0;
1109
}
1110

    
1111

    
1112
/**
1113
 * Process subpacket 10 if not null, else
1114
 *
1115
 * @param q         context
1116
 * @param node      pointer to node with packet
1117
 * @param length    packet length in bit
1118
 */
1119
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1120
{
1121
    GetBitContext gb;
1122

    
1123
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1124

    
1125
    if (length != 0) {
1126
        init_tone_level_dequantization(q, &gb, length);
1127
        fill_tone_level_array(q, 1);
1128
    } else {
1129
        fill_tone_level_array(q, 0);
1130
    }
1131
}
1132

    
1133

    
1134
/**
1135
 * Process subpacket 11
1136
 *
1137
 * @param q         context
1138
 * @param node      pointer to node with packet
1139
 * @param length    packet length in bit
1140
 */
1141
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1142
{
1143
    GetBitContext gb;
1144

    
1145
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1146
    if (length >= 32) {
1147
        int c = get_bits (&gb, 13);
1148

    
1149
        if (c > 3)
1150
            fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1151
                                      q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1152
    }
1153

    
1154
    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1155
}
1156

    
1157

    
1158
/**
1159
 * Process subpacket 12
1160
 *
1161
 * @param q         context
1162
 * @param node      pointer to node with packet
1163
 * @param length    packet length in bit
1164
 */
1165
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1166
{
1167
    GetBitContext gb;
1168

    
1169
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1170
    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1171
}
1172

    
1173
/*
1174
 * Process new subpackets for synthesis filter
1175
 *
1176
 * @param q       context
1177
 * @param list    list with synthesis filter packets (list D)
1178
 */
1179
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1180
{
1181
    QDM2SubPNode *nodes[4];
1182

    
1183
    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1184
    if (nodes[0] != NULL)
1185
        process_subpacket_9(q, nodes[0]);
1186

    
1187
    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1188
    if (nodes[1] != NULL)
1189
        process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1190
    else
1191
        process_subpacket_10(q, NULL, 0);
1192

    
1193
    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1194
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1195
        process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1196
    else
1197
        process_subpacket_11(q, NULL, 0);
1198

    
1199
    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1200
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1201
        process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1202
    else
1203
        process_subpacket_12(q, NULL, 0);
1204
}
1205

    
1206

    
1207
/*
1208
 * Decode superblock, fill packet lists
1209
 *
1210
 * @param q    context
1211
 */
1212
static void qdm2_decode_super_block (QDM2Context *q)
1213
{
1214
    GetBitContext gb;
1215
    QDM2SubPacket header, *packet;
1216
    int i, packet_bytes, sub_packet_size, sub_packets_D;
1217
    unsigned int next_index = 0;
1218

    
1219
    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1220
    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1221
    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1222

    
1223
    q->sub_packets_B = 0;
1224
    sub_packets_D = 0;
1225

    
1226
    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1227

    
1228
    init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1229
    qdm2_decode_sub_packet_header(&gb, &header);
1230

    
1231
    if (header.type < 2 || header.type >= 8) {
1232
        q->has_errors = 1;
1233
        av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1234
        return;
1235
    }
1236

    
1237
    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1238
    packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1239

    
1240
    init_get_bits(&gb, header.data, header.size*8);
1241

    
1242
    if (header.type == 2 || header.type == 4 || header.type == 5) {
1243
        int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1244

    
1245
        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1246

    
1247
        if (csum != 0) {
1248
            q->has_errors = 1;
1249
            av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1250
            return;
1251
        }
1252
    }
1253

    
1254
    q->sub_packet_list_B[0].packet = NULL;
1255
    q->sub_packet_list_D[0].packet = NULL;
1256

    
1257
    for (i = 0; i < 6; i++)
1258
        if (--q->fft_level_exp[i] < 0)
1259
            q->fft_level_exp[i] = 0;
1260

    
1261
    for (i = 0; packet_bytes > 0; i++) {
1262
        int j;
1263

    
1264
        q->sub_packet_list_A[i].next = NULL;
1265

    
1266
        if (i > 0) {
1267
            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1268

    
1269
            /* seek to next block */
1270
            init_get_bits(&gb, header.data, header.size*8);
1271
            skip_bits(&gb, next_index*8);
1272

    
1273
            if (next_index >= header.size)
1274
                break;
1275
        }
1276

    
1277
        /* decode sub packet */
1278
        packet = &q->sub_packets[i];
1279
        qdm2_decode_sub_packet_header(&gb, packet);
1280
        next_index = packet->size + get_bits_count(&gb) / 8;
1281
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1282

    
1283
        if (packet->type == 0)
1284
            break;
1285

    
1286
        if (sub_packet_size > packet_bytes) {
1287
            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1288
                break;
1289
            packet->size += packet_bytes - sub_packet_size;
1290
        }
1291

    
1292
        packet_bytes -= sub_packet_size;
1293

    
1294
        /* add sub packet to 'all sub packets' list */
1295
        q->sub_packet_list_A[i].packet = packet;
1296

    
1297
        /* add sub packet to related list */
1298
        if (packet->type == 8) {
1299
            SAMPLES_NEEDED_2("packet type 8");
1300
            return;
1301
        } else if (packet->type >= 9 && packet->type <= 12) {
1302
            /* packets for MPEG Audio like Synthesis Filter */
1303
            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1304
        } else if (packet->type == 13) {
1305
            for (j = 0; j < 6; j++)
1306
                q->fft_level_exp[j] = get_bits(&gb, 6);
1307
        } else if (packet->type == 14) {
1308
            for (j = 0; j < 6; j++)
1309
                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1310
        } else if (packet->type == 15) {
1311
            SAMPLES_NEEDED_2("packet type 15")
1312
            return;
1313
        } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1314
            /* packets for FFT */
1315
            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1316
        }
1317
    } // Packet bytes loop
1318

    
1319
/* **************************************************************** */
1320
    if (q->sub_packet_list_D[0].packet != NULL) {
1321
        process_synthesis_subpackets(q, q->sub_packet_list_D);
1322
        q->do_synth_filter = 1;
1323
    } else if (q->do_synth_filter) {
1324
        process_subpacket_10(q, NULL, 0);
1325
        process_subpacket_11(q, NULL, 0);
1326
        process_subpacket_12(q, NULL, 0);
1327
    }
1328
/* **************************************************************** */
1329
}
1330

    
1331

    
1332
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1333
                       int offset, int duration, int channel,
1334
                       int exp, int phase)
1335
{
1336
    if (q->fft_coefs_min_index[duration] < 0)
1337
        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1338

    
1339
    q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1340
    q->fft_coefs[q->fft_coefs_index].channel = channel;
1341
    q->fft_coefs[q->fft_coefs_index].offset = offset;
1342
    q->fft_coefs[q->fft_coefs_index].exp = exp;
1343
    q->fft_coefs[q->fft_coefs_index].phase = phase;
1344
    q->fft_coefs_index++;
1345
}
1346

    
1347

    
1348
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1349
{
1350
    int channel, stereo, phase, exp;
1351
    int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
1352
    int local_int_14, stereo_exp, local_int_20, local_int_28;
1353
    int n, offset;
1354

    
1355
    local_int_4 = 0;
1356
    local_int_28 = 0;
1357
    local_int_20 = 2;
1358
    local_int_8 = (4 - duration);
1359
    local_int_10 = 1 << (q->group_order - duration - 1);
1360
    offset = 1;
1361

    
1362
    while (1) {
1363
        if (q->superblocktype_2_3) {
1364
            while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1365
                offset = 1;
1366
                if (n == 0) {
1367
                    local_int_4 += local_int_10;
1368
                    local_int_28 += (1 << local_int_8);
1369
                } else {
1370
                    local_int_4 += 8*local_int_10;
1371
                    local_int_28 += (8 << local_int_8);
1372
                }
1373
            }
1374
            offset += (n - 2);
1375
        } else {
1376
            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1377
            while (offset >= (local_int_10 - 1)) {
1378
                offset += (1 - (local_int_10 - 1));
1379
                local_int_4  += local_int_10;
1380
                local_int_28 += (1 << local_int_8);
1381
            }
1382
        }
1383

    
1384
        if (local_int_4 >= q->group_size)
1385
            return;
1386

    
1387
        local_int_14 = (offset >> local_int_8);
1388

    
1389
        if (q->nb_channels > 1) {
1390
            channel = get_bits1(gb);
1391
            stereo = get_bits1(gb);
1392
        } else {
1393
            channel = 0;
1394
            stereo = 0;
1395
        }
1396

    
1397
        exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1398
        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1399
        exp = (exp < 0) ? 0 : exp;
1400

    
1401
        phase = get_bits(gb, 3);
1402
        stereo_exp = 0;
1403
        stereo_phase = 0;
1404

    
1405
        if (stereo) {
1406
            stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1407
            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1408
            if (stereo_phase < 0)
1409
                stereo_phase += 8;
1410
        }
1411

    
1412
        if (q->frequency_range > (local_int_14 + 1)) {
1413
            int sub_packet = (local_int_20 + local_int_28);
1414

    
1415
            qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1416
            if (stereo)
1417
                qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1418
        }
1419

    
1420
        offset++;
1421
    }
1422
}
1423

    
1424

    
1425
static void qdm2_decode_fft_packets (QDM2Context *q)
1426
{
1427
    int i, j, min, max, value, type, unknown_flag;
1428
    GetBitContext gb;
1429

    
1430
    if (q->sub_packet_list_B[0].packet == NULL)
1431
        return;
1432

    
1433
    /* reset minimum indices for FFT coefficients */
1434
    q->fft_coefs_index = 0;
1435
    for (i=0; i < 5; i++)
1436
        q->fft_coefs_min_index[i] = -1;
1437

    
1438
    /* process sub packets ordered by type, largest type first */
1439
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1440
        QDM2SubPacket *packet;
1441

    
1442
        /* find sub packet with largest type less than max */
1443
        for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1444
            value = q->sub_packet_list_B[j].packet->type;
1445
            if (value > min && value < max) {
1446
                min = value;
1447
                packet = q->sub_packet_list_B[j].packet;
1448
            }
1449
        }
1450

    
1451
        max = min;
1452

    
1453
        /* check for errors (?) */
1454
        if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1455
            return;
1456

    
1457
        /* decode FFT tones */
1458
        init_get_bits (&gb, packet->data, packet->size*8);
1459

    
1460
        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1461
            unknown_flag = 1;
1462
        else
1463
            unknown_flag = 0;
1464

    
1465
        type = packet->type;
1466

    
1467
        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1468
            int duration = q->sub_sampling + 5 - (type & 15);
1469

    
1470
            if (duration >= 0 && duration < 4)
1471
                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1472
        } else if (type == 31) {
1473
            for (i=0; i < 4; i++)
1474
                qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
1475
        } else if (type == 46) {
1476
            for (i=0; i < 6; i++)
1477
                q->fft_level_exp[i] = get_bits(&gb, 6);
1478
            for (i=0; i < 4; i++)
1479
            qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
1480
        }
1481
    } // Loop on B packets
1482

    
1483
    /* calculate maximum indices for FFT coefficients */
1484
    for (i = 0, j = -1; i < 5; i++)
1485
        if (q->fft_coefs_min_index[i] >= 0) {
1486
            if (j >= 0)
1487
                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1488
            j = i;
1489
        }
1490
    if (j >= 0)
1491
        q->fft_coefs_max_index[j] = q->fft_coefs_index;
1492
}
1493

    
1494

    
1495
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1496
{
1497
   float level, f[6];
1498
   int i;
1499
   QDM2Complex c;
1500
   const double iscale = 2.0*M_PI / 512.0;
1501

    
1502
    tone->phase += tone->phase_shift;
1503

    
1504
    /* calculate current level (maximum amplitude) of tone */
1505
    level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1506
    c.im = level * sin(tone->phase*iscale);
1507
    c.re = level * cos(tone->phase*iscale);
1508

    
1509
    /* generate FFT coefficients for tone */
1510
    if (tone->duration >= 3 || tone->cutoff >= 3) {
1511
        tone->samples_im[0] += c.im;
1512
        tone->samples_re[0] += c.re;
1513
        tone->samples_im[1] -= c.im;
1514
        tone->samples_re[1] -= c.re;
1515
    } else {
1516
        f[1] = -tone->table[4];
1517
        f[0] =  tone->table[3] - tone->table[0];
1518
        f[2] =  1.0 - tone->table[2] - tone->table[3];
1519
        f[3] =  tone->table[1] + tone->table[4] - 1.0;
1520
        f[4] =  tone->table[0] - tone->table[1];
1521
        f[5] =  tone->table[2];
1522
        for (i = 0; i < 2; i++) {
1523
            tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1524
            tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1525
        }
1526
        for (i = 0; i < 4; i++) {
1527
            tone->samples_re[i] += c.re * f[i+2];
1528
            tone->samples_im[i] += c.im * f[i+2];
1529
        }
1530
    }
1531

    
1532
    /* copy the tone if it has not yet died out */
1533
    if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1534
      memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1535
      q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1536
    }
1537
}
1538

    
1539

    
1540
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1541
{
1542
    int i, j, ch;
1543
    const double iscale = 0.25 * M_PI;
1544

    
1545
    for (ch = 0; ch < q->channels; ch++) {
1546
        memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1547
        memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1548
    }
1549

    
1550

    
1551
    /* apply FFT tones with duration 4 (1 FFT period) */
1552
    if (q->fft_coefs_min_index[4] >= 0)
1553
        for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1554
            float level;
1555
            QDM2Complex c;
1556

    
1557
            if (q->fft_coefs[i].sub_packet != sub_packet)
1558
                break;
1559

    
1560
            ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1561
            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];
1562

    
1563
            c.re = level * cos(q->fft_coefs[i].phase * iscale);
1564
            c.im = level * sin(q->fft_coefs[i].phase * iscale);
1565
            q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1566
            q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1567
            q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1568
            q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1569
        }
1570

    
1571
    /* generate existing FFT tones */
1572
    for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1573
        qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1574
        q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1575
    }
1576

    
1577
    /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1578
    for (i = 0; i < 4; i++)
1579
        if (q->fft_coefs_min_index[i] >= 0) {
1580
            for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1581
                int offset, four_i;
1582
                FFTTone tone;
1583

    
1584
                if (q->fft_coefs[j].sub_packet != sub_packet)
1585
                    break;
1586

    
1587
                four_i = (4 - i);
1588
                offset = q->fft_coefs[j].offset >> four_i;
1589
                ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1590

    
1591
                if (offset < q->frequency_range) {
1592
                    if (offset < 2)
1593
                        tone.cutoff = offset;
1594
                    else
1595
                        tone.cutoff = (offset >= 60) ? 3 : 2;
1596

    
1597
                    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];
1598
                    tone.samples_im = &q->fft.samples_im[ch][offset];
1599
                    tone.samples_re = &q->fft.samples_re[ch][offset];
1600
                    tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1601
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1602
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1603
                    tone.duration = i;
1604
                    tone.time_index = 0;
1605

    
1606
                    qdm2_fft_generate_tone(q, &tone);
1607
                }
1608
            }
1609
            q->fft_coefs_min_index[i] = j;
1610
        }
1611
}
1612

    
1613

    
1614
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1615
{
1616
    const int n = 1 << (q->fft_order - 1);
1617
    const int n2 = n >> 1;
1618
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1619
    float c, s, f0, f1, f2, f3;
1620
    int i, j;
1621

    
1622
    /* pre rotation (or something like that) */
1623
    for (i=1; i < n2; i++) {
1624
        j  = (n - i);
1625
        c = q->exptab[i].re;
1626
        s = -q->exptab[i].im;
1627
        f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1628
        f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1629
        f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1630
        f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1631
        q->fft.complex[i].re =  s * f0 - c * f1 + f2;
1632
        q->fft.complex[i].im =  c * f0 + s * f1 + f3;
1633
        q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1634
        q->fft.complex[j].im =  c * f0 + s * f1 - f3;
1635
    }
1636

    
1637
    q->fft.complex[ 0].re =  q->fft.samples_re[channel][ 0] * gain * 2.0;
1638
    q->fft.complex[ 0].im =  q->fft.samples_re[channel][ 0] * gain * 2.0;
1639
    q->fft.complex[n2].re =  q->fft.samples_re[channel][n2] * gain * 2.0;
1640
    q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1641

    
1642
    ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1643
    ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1644
    /* add samples to output buffer */
1645
    for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1646
        q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1647
}
1648

    
1649

    
1650
/**
1651
 * @param q        context
1652
 * @param index    subpacket number
1653
 */
1654
static void qdm2_synthesis_filter (QDM2Context *q, int index)
1655
{
1656
    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1657
    int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1658

    
1659
    /* copy sb_samples */
1660
    sb_used = QDM2_SB_USED(q->sub_sampling);
1661

    
1662
    for (ch = 0; ch < q->channels; ch++)
1663
        for (i = 0; i < 8; i++)
1664
            for (k=sb_used; k < SBLIMIT; k++)
1665
                q->sb_samples[ch][(8 * index) + i][k] = 0;
1666

    
1667
    for (ch = 0; ch < q->nb_channels; ch++) {
1668
        OUT_INT *samples_ptr = samples + ch;
1669

    
1670
        for (i = 0; i < 8; i++) {
1671
            ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1672
                mpa_window, &dither_state,
1673
                samples_ptr, q->nb_channels,
1674
                q->sb_samples[ch][(8 * index) + i]);
1675
            samples_ptr += 32 * q->nb_channels;
1676
        }
1677
    }
1678

    
1679
    /* add samples to output buffer */
1680
    sub_sampling = (4 >> q->sub_sampling);
1681

    
1682
    for (ch = 0; ch < q->channels; ch++)
1683
        for (i = 0; i < q->frame_size; i++)
1684
            q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1685
}
1686

    
1687

    
1688
/**
1689
 * Init static data (does not depend on specific file)
1690
 *
1691
 * @param q    context
1692
 */
1693
void qdm2_init(QDM2Context *q) {
1694
    static int inited = 0;
1695

    
1696
    if (inited != 0)
1697
        return;
1698
    inited = 1;
1699

    
1700
    qdm2_init_vlc();
1701
    ff_mpa_synth_init(mpa_window);
1702
    softclip_table_init();
1703
    rnd_table_init();
1704
    init_noise_samples();
1705

    
1706
    av_log(NULL, AV_LOG_DEBUG, "init done\n");
1707
}
1708

    
1709

    
1710
#if 0
1711
static void dump_context(QDM2Context *q)
1712
{
1713
    int i;
1714
#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1715
    PRINT("compressed_data",q->compressed_data);
1716
    PRINT("compressed_size",q->compressed_size);
1717
    PRINT("frame_size",q->frame_size);
1718
    PRINT("checksum_size",q->checksum_size);
1719
    PRINT("channels",q->channels);
1720
    PRINT("nb_channels",q->nb_channels);
1721
    PRINT("fft_frame_size",q->fft_frame_size);
1722
    PRINT("fft_size",q->fft_size);
1723
    PRINT("sub_sampling",q->sub_sampling);
1724
    PRINT("fft_order",q->fft_order);
1725
    PRINT("group_order",q->group_order);
1726
    PRINT("group_size",q->group_size);
1727
    PRINT("sub_packet",q->sub_packet);
1728
    PRINT("frequency_range",q->frequency_range);
1729
    PRINT("has_errors",q->has_errors);
1730
    PRINT("fft_tone_end",q->fft_tone_end);
1731
    PRINT("fft_tone_start",q->fft_tone_start);
1732
    PRINT("fft_coefs_index",q->fft_coefs_index);
1733
    PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1734
    PRINT("cm_table_select",q->cm_table_select);
1735
    PRINT("noise_idx",q->noise_idx);
1736

1737
    for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1738
    {
1739
    FFTTone *t = &q->fft_tones[i];
1740

1741
    av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1742
    av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
1743
//  PRINT(" level", t->level);
1744
    PRINT(" phase", t->phase);
1745
    PRINT(" phase_shift", t->phase_shift);
1746
    PRINT(" duration", t->duration);
1747
    PRINT(" samples_im", t->samples_im);
1748
    PRINT(" samples_re", t->samples_re);
1749
    PRINT(" table", t->table);
1750
    }
1751

1752
}
1753
#endif
1754

    
1755

    
1756
/**
1757
 * Init parameters from codec extradata
1758
 */
1759
static int qdm2_decode_init(AVCodecContext *avctx)
1760
{
1761
    QDM2Context *s = avctx->priv_data;
1762
    uint8_t *extradata;
1763
    int extradata_size;
1764
    int tmp_val, tmp, size;
1765
    int i;
1766
    float alpha;
1767

    
1768
    /* extradata parsing
1769

1770
    Structure:
1771
    wave {
1772
        frma (QDM2)
1773
        QDCA
1774
        QDCP
1775
    }
1776

1777
    32  size (including this field)
1778
    32  tag (=frma)
1779
    32  type (=QDM2 or QDMC)
1780

1781
    32  size (including this field, in bytes)
1782
    32  tag (=QDCA) // maybe mandatory parameters
1783
    32  unknown (=1)
1784
    32  channels (=2)
1785
    32  samplerate (=44100)
1786
    32  bitrate (=96000)
1787
    32  block size (=4096)
1788
    32  frame size (=256) (for one channel)
1789
    32  packet size (=1300)
1790

1791
    32  size (including this field, in bytes)
1792
    32  tag (=QDCP) // maybe some tuneable parameters
1793
    32  float1 (=1.0)
1794
    32  zero ?
1795
    32  float2 (=1.0)
1796
    32  float3 (=1.0)
1797
    32  unknown (27)
1798
    32  unknown (8)
1799
    32  zero ?
1800
    */
1801

    
1802
    if (!avctx->extradata || (avctx->extradata_size < 48)) {
1803
        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1804
        return -1;
1805
    }
1806

    
1807
    extradata = avctx->extradata;
1808
    extradata_size = avctx->extradata_size;
1809

    
1810
    while (extradata_size > 7) {
1811
        if (!memcmp(extradata, "frmaQDM", 7))
1812
            break;
1813
        extradata++;
1814
        extradata_size--;
1815
    }
1816

    
1817
    if (extradata_size < 12) {
1818
        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1819
               extradata_size);
1820
        return -1;
1821
    }
1822

    
1823
    if (memcmp(extradata, "frmaQDM", 7)) {
1824
        av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1825
        return -1;
1826
    }
1827

    
1828
    if (extradata[7] == 'C') {
1829
//        s->is_qdmc = 1;
1830
        av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1831
        return -1;
1832
    }
1833

    
1834
    extradata += 8;
1835
    extradata_size -= 8;
1836

    
1837
    size = BE_32(extradata);
1838

    
1839
    if(size > extradata_size){
1840
        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1841
               extradata_size, size);
1842
        return -1;
1843
    }
1844

    
1845
    extradata += 4;
1846
    av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1847
    if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
1848
        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1849
        return -1;
1850
    }
1851

    
1852
    extradata += 8;
1853

    
1854
    avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
1855
    extradata += 4;
1856

    
1857
    avctx->sample_rate = BE_32(extradata);
1858
    extradata += 4;
1859

    
1860
    avctx->bit_rate = BE_32(extradata);
1861
    extradata += 4;
1862

    
1863
    s->group_size = BE_32(extradata);
1864
    extradata += 4;
1865

    
1866
    s->fft_size = BE_32(extradata);
1867
    extradata += 4;
1868

    
1869
    s->checksum_size = BE_32(extradata);
1870
    extradata += 4;
1871

    
1872
    s->fft_order = av_log2(s->fft_size) + 1;
1873
    s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1874

    
1875
    // something like max decodable tones
1876
    s->group_order = av_log2(s->group_size) + 1;
1877
    s->frame_size = s->group_size / 16; // 16 iterations per super block
1878

    
1879
    s->sub_sampling = s->fft_order - 7;
1880
    s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1881

    
1882
    switch ((s->sub_sampling * 2 + s->channels - 1)) {
1883
        case 0: tmp = 40; break;
1884
        case 1: tmp = 48; break;
1885
        case 2: tmp = 56; break;
1886
        case 3: tmp = 72; break;
1887
        case 4: tmp = 80; break;
1888
        case 5: tmp = 100;break;
1889
        default: tmp=s->sub_sampling; break;
1890
    }
1891
    tmp_val = 0;
1892
    if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
1893
    if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
1894
    if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
1895
    if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
1896
    s->cm_table_select = tmp_val;
1897

    
1898
    if (s->sub_sampling == 0)
1899
        tmp = 7999;
1900
    else
1901
        tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1902
    /*
1903
    0: 7999 -> 0
1904
    1: 20000 -> 2
1905
    2: 28000 -> 2
1906
    */
1907
    if (tmp < 8000)
1908
        s->coeff_per_sb_select = 0;
1909
    else if (tmp <= 16000)
1910
        s->coeff_per_sb_select = 1;
1911
    else
1912
        s->coeff_per_sb_select = 2;
1913

    
1914
    // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
1915
    if ((s->fft_order < 7) || (s->fft_order > 9)) {
1916
        av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1917
        return -1;
1918
    }
1919

    
1920
    ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1921

    
1922
    for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1923
        alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1924
        s->exptab[i].re = cos(alpha);
1925
        s->exptab[i].im = sin(alpha);
1926
    }
1927

    
1928
    qdm2_init(s);
1929

    
1930
//    dump_context(s);
1931
    return 0;
1932
}
1933

    
1934

    
1935
static int qdm2_decode_close(AVCodecContext *avctx)
1936
{
1937
    QDM2Context *s = avctx->priv_data;
1938

    
1939
    ff_fft_end(&s->fft_ctx);
1940

    
1941
    return 0;
1942
}
1943

    
1944

    
1945
void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
1946
{
1947
    int ch, i;
1948
    const int frame_size = (q->frame_size * q->channels);
1949

    
1950
    /* select input buffer */
1951
    q->compressed_data = in;
1952
    q->compressed_size = q->checksum_size;
1953

    
1954
//  dump_context(q);
1955

    
1956
    /* copy old block, clear new block of output samples */
1957
    memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1958
    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1959

    
1960
    /* decode block of QDM2 compressed data */
1961
    if (q->sub_packet == 0) {
1962
        q->has_errors = 0; // zero it for a new super block
1963
        av_log(NULL,AV_LOG_DEBUG,"Super block follows\n");
1964
        qdm2_decode_super_block(q);
1965
    }
1966

    
1967
    /* parse sub packets */
1968
    if (!q->has_errors) {
1969
        if (q->sub_packet == 2)
1970
            qdm2_decode_fft_packets(q);
1971

    
1972
        qdm2_fft_tone_synthesizer(q, q->sub_packet);
1973
    }
1974

    
1975
    /* sound synthesis stage 1 (FFT) */
1976
    for (ch = 0; ch < q->channels; ch++) {
1977
        qdm2_calculate_fft(q, ch, q->sub_packet);
1978

    
1979
        if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1980
            SAMPLES_NEEDED_2("has errors, and C list is not empty")
1981
            return;
1982
        }
1983
    }
1984

    
1985
    /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1986
    if (!q->has_errors && q->do_synth_filter)
1987
        qdm2_synthesis_filter(q, q->sub_packet);
1988

    
1989
    q->sub_packet = (q->sub_packet + 1) % 16;
1990

    
1991
    /* clip and convert output float[] to 16bit signed samples */
1992
    for (i = 0; i < frame_size; i++) {
1993
        int value = (int)q->output_buffer[i];
1994

    
1995
        if (value > SOFTCLIP_THRESHOLD)
1996
            value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
1997
        else if (value < -SOFTCLIP_THRESHOLD)
1998
            value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1999

    
2000
        out[i] = value;
2001
    }
2002
}
2003

    
2004

    
2005
static int qdm2_decode_frame(AVCodecContext *avctx,
2006
            void *data, int *data_size,
2007
            uint8_t *buf, int buf_size)
2008
{
2009
    QDM2Context *s = avctx->priv_data;
2010

    
2011
    if((buf == NULL) || (buf_size < s->checksum_size))
2012
        return 0;
2013

    
2014
    *data_size = s->channels * s->frame_size * sizeof(int16_t);
2015

    
2016
    av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2017
       buf_size, buf, s->checksum_size, data, *data_size);
2018

    
2019
    qdm2_decode(s, buf, data);
2020

    
2021
    // reading only when next superblock found
2022
    if (s->sub_packet == 0) {
2023
        return s->checksum_size;
2024
    }
2025

    
2026
    return 0;
2027
}
2028

    
2029
AVCodec qdm2_decoder =
2030
{
2031
    .name = "qdm2",
2032
    .type = CODEC_TYPE_AUDIO,
2033
    .id = CODEC_ID_QDM2,
2034
    .priv_data_size = sizeof(QDM2Context),
2035
    .init = qdm2_decode_init,
2036
    .close = qdm2_decode_close,
2037
    .decode = qdm2_decode_frame,
2038
};