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ffmpeg / libavcodec / qdm2.c @ bb54f6ab

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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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/**
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 * @file qdm2.c
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 * QDM2 decoder
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 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
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 * The decoder is not perfect yet, there are still some distortions
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 * especially on files encoded with 16 or 8 subbands.
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 */
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#include <math.h>
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#include <stddef.h>
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#include <stdio.h>
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#define ALT_BITSTREAM_READER_LE
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#include "avcodec.h"
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#include "bitstream.h"
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#include "dsputil.h"
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#ifdef CONFIG_MPEGAUDIO_HP
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#define USE_HIGHPRECISION
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#endif
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#include "mpegaudio.h"
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#include "qdm2data.h"
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#undef NDEBUG
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#include <assert.h>
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54

    
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#define SOFTCLIP_THRESHOLD 27600
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#define HARDCLIP_THRESHOLD 35716
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58

    
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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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#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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86

    
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typedef int8_t sb_int8_array[2][30][64];
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/**
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 * Subpacket
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 */
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typedef struct {
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    int type;            ///< subpacket type
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    unsigned int size;   ///< subpacket size
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    const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
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} QDM2SubPacket;
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/**
99
 * A node in the subpacket list
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 */
101
typedef struct _QDM2SubPNode {
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    QDM2SubPacket *packet;      ///< packet
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    struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
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} QDM2SubPNode;
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typedef struct {
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    float level;
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    float *samples_im;
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    float *samples_re;
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    float *table;
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    int   phase;
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    int   phase_shift;
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    int   duration;
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    short time_index;
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    short cutoff;
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} FFTTone;
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typedef struct {
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    int16_t sub_packet;
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    uint8_t channel;
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    int16_t offset;
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    int16_t exp;
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    uint8_t phase;
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} FFTCoefficient;
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126
typedef struct {
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    float re;
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    float im;
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} QDM2Complex;
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131
typedef struct {
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    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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/**
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 * QDM2 decoder context
139
 */
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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
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    QDM2SubPacket sub_packets[16];      ///< the packets themselves
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    QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
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    QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
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    int sub_packets_B;                  ///< number of packets on 'B' list
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    QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
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    QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
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    /// FFT and tones
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    FFTTone fft_tones[1000];
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    int fft_tone_start;
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    int fft_tone_end;
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    FFTCoefficient fft_coefs[1000];
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    int fft_coefs_index;
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    int fft_coefs_min_index[5];
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    int fft_coefs_max_index[5];
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    int fft_level_exp[6];
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    FFTContext fft_ctx;
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    FFTComplex exptab[128];
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    QDM2FFT fft;
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    /// I/O data
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    uint8_t *compressed_data;
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    int compressed_size;
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    float output_buffer[1024];
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    /// Synthesis filter
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    MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
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    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
190
    float tone_level[MPA_MAX_CHANNELS][30][64];
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    int8_t coding_method[MPA_MAX_CHANNELS][30][64];
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    int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
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    int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
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    int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
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    int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
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    int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
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    int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
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    int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
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    // Flags
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    int has_errors;         ///< packet has errors
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    int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
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    int do_synth_filter;    ///< used to perform or skip synthesis filter
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    int sub_packet;
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    int noise_idx; ///< index for dithering noise table
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} QDM2Context;
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static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
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static VLC vlc_tab_level;
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static VLC vlc_tab_diff;
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static VLC vlc_tab_run;
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static VLC fft_level_exp_alt_vlc;
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static VLC fft_level_exp_vlc;
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static VLC fft_stereo_exp_vlc;
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static VLC fft_stereo_phase_vlc;
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static VLC vlc_tab_tone_level_idx_hi1;
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static VLC vlc_tab_tone_level_idx_mid;
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static VLC vlc_tab_tone_level_idx_hi2;
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static VLC vlc_tab_type30;
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static VLC vlc_tab_type34;
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static VLC vlc_tab_fft_tone_offset[5];
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static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
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static float noise_table[4096];
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static uint8_t random_dequant_index[256][5];
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static uint8_t random_dequant_type24[128][3];
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static float noise_samples[128];
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static MPA_INT mpa_window[512] __attribute__((aligned(16)));
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234

    
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static void softclip_table_init(void) {
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    int i;
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    double dfl = SOFTCLIP_THRESHOLD - 32767;
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    float delta = 1.0 / -dfl;
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    for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
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        softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
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}
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243

    
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// random generated table
245
static void rnd_table_init(void) {
246
    int i,j;
247
    uint32_t ldw,hdw;
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    uint64_t tmp64_1;
249
    uint64_t random_seed = 0;
250
    float delta = 1.0 / 16384.0;
251
    for(i = 0; i < 4096 ;i++) {
252
        random_seed = random_seed * 214013 + 2531011;
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        noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
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    }
255

    
256
    for (i = 0; i < 256 ;i++) {
257
        random_seed = 81;
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        ldw = i;
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        for (j = 0; j < 5 ;j++) {
260
            random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
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            ldw = (uint32_t)ldw % (uint32_t)random_seed;
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            tmp64_1 = (random_seed * 0x55555556);
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            hdw = (uint32_t)(tmp64_1 >> 32);
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            random_seed = (uint64_t)(hdw + (ldw >> 31));
265
        }
266
    }
267
    for (i = 0; i < 128 ;i++) {
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        random_seed = 25;
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        ldw = i;
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        for (j = 0; j < 3 ;j++) {
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            random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
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            ldw = (uint32_t)ldw % (uint32_t)random_seed;
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            tmp64_1 = (random_seed * 0x66666667);
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            hdw = (uint32_t)(tmp64_1 >> 33);
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            random_seed = hdw + (ldw >> 31);
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        }
277
    }
278
}
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280

    
281
static void init_noise_samples(void) {
282
    int i;
283
    int random_seed = 0;
284
    float delta = 1.0 / 16384.0;
285
    for (i = 0; i < 128;i++) {
286
        random_seed = random_seed * 214013 + 2531011;
287
        noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
288
    }
289
}
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static void qdm2_init_vlc(void)
293
{
294
    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);
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298
    init_vlc (&vlc_tab_diff, 8, 37,
299
        vlc_tab_diff_huffbits, 1, 1,
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        vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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302
    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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306
    init_vlc (&fft_level_exp_alt_vlc, 8, 28,
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        fft_level_exp_alt_huffbits, 1, 1,
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        fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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310
    init_vlc (&fft_level_exp_vlc, 8, 20,
311
        fft_level_exp_huffbits, 1, 1,
312
        fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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314
    init_vlc (&fft_stereo_exp_vlc, 6, 7,
315
        fft_stereo_exp_huffbits, 1, 1,
316
        fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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318
    init_vlc (&fft_stereo_phase_vlc, 6, 9,
319
        fft_stereo_phase_huffbits, 1, 1,
320
        fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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322
    init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
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        vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
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        vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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326
    init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
327
        vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
328
        vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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330
    init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
331
        vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
332
        vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
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334
    init_vlc (&vlc_tab_type30, 6, 9,
335
        vlc_tab_type30_huffbits, 1, 1,
336
        vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
337

    
338
    init_vlc (&vlc_tab_type34, 5, 10,
339
        vlc_tab_type34_huffbits, 1, 1,
340
        vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
341

    
342
    init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
343
        vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
344
        vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
345

    
346
    init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
347
        vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
348
        vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
349

    
350
    init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
351
        vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
352
        vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
353

    
354
    init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
355
        vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
356
        vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
357

    
358
    init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
359
        vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
360
        vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
361
}
362

    
363

    
364
/* for floating point to fixed point conversion */
365
static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
366

    
367

    
368
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
369
{
370
    int value;
371

    
372
    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
373

    
374
    /* stage-2, 3 bits exponent escape sequence */
375
    if (value-- == 0)
376
        value = get_bits (gb, get_bits (gb, 3) + 1);
377

    
378
    /* stage-3, optional */
379
    if (flag) {
380
        int tmp = vlc_stage3_values[value];
381

    
382
        if ((value & ~3) > 0)
383
            tmp += get_bits (gb, (value >> 2));
384
        value = tmp;
385
    }
386

    
387
    return value;
388
}
389

    
390

    
391
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
392
{
393
    int value = qdm2_get_vlc (gb, vlc, 0, depth);
394

    
395
    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
396
}
397

    
398

    
399
/**
400
 * QDM2 checksum
401
 *
402
 * @param data      pointer to data to be checksum'ed
403
 * @param length    data length
404
 * @param value     checksum value
405
 *
406
 * @return          0 if checksum is OK
407
 */
408
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
409
    int i;
410

    
411
    for (i=0; i < length; i++)
412
        value -= data[i];
413

    
414
    return (uint16_t)(value & 0xffff);
415
}
416

    
417

    
418
/**
419
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
420
 *
421
 * @param gb            bitreader context
422
 * @param sub_packet    packet under analysis
423
 */
424
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
425
{
426
    sub_packet->type = get_bits (gb, 8);
427

    
428
    if (sub_packet->type == 0) {
429
        sub_packet->size = 0;
430
        sub_packet->data = NULL;
431
    } else {
432
        sub_packet->size = get_bits (gb, 8);
433

    
434
      if (sub_packet->type & 0x80) {
435
          sub_packet->size <<= 8;
436
          sub_packet->size  |= get_bits (gb, 8);
437
          sub_packet->type  &= 0x7f;
438
      }
439

    
440
      if (sub_packet->type == 0x7f)
441
          sub_packet->type |= (get_bits (gb, 8) << 8);
442

    
443
      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
444
    }
445

    
446
    av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
447
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
448
}
449

    
450

    
451
/**
452
 * Return node pointer to first packet of requested type in list.
453
 *
454
 * @param list    list of subpackets to be scanned
455
 * @param type    type of searched subpacket
456
 * @return        node pointer for subpacket if found, else NULL
457
 */
458
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
459
{
460
    while (list != NULL && list->packet != NULL) {
461
        if (list->packet->type == type)
462
            return list;
463
        list = list->next;
464
    }
465
    return NULL;
466
}
467

    
468

    
469
/**
470
 * Replaces 8 elements with their average value.
471
 * Called by qdm2_decode_superblock before starting subblock decoding.
472
 *
473
 * @param q       context
474
 */
475
static void average_quantized_coeffs (QDM2Context *q)
476
{
477
    int i, j, n, ch, sum;
478

    
479
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
480

    
481
    for (ch = 0; ch < q->nb_channels; ch++)
482
        for (i = 0; i < n; i++) {
483
            sum = 0;
484

    
485
            for (j = 0; j < 8; j++)
486
                sum += q->quantized_coeffs[ch][i][j];
487

    
488
            sum /= 8;
489
            if (sum > 0)
490
                sum--;
491

    
492
            for (j=0; j < 8; j++)
493
                q->quantized_coeffs[ch][i][j] = sum;
494
        }
495
}
496

    
497

    
498
/**
499
 * Build subband samples with noise weighted by q->tone_level.
500
 * Called by synthfilt_build_sb_samples.
501
 *
502
 * @param q     context
503
 * @param sb    subband index
504
 */
505
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
506
{
507
    int ch, j;
508

    
509
    FIX_NOISE_IDX(q->noise_idx);
510

    
511
    if (!q->nb_channels)
512
        return;
513

    
514
    for (ch = 0; ch < q->nb_channels; ch++)
515
        for (j = 0; j < 64; j++) {
516
            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);
517
            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);
518
        }
519
}
520

    
521

    
522
/**
523
 * Called while processing data from subpackets 11 and 12.
524
 * Used after making changes to coding_method array.
525
 *
526
 * @param sb               subband index
527
 * @param channels         number of channels
528
 * @param coding_method    q->coding_method[0][0][0]
529
 */
530
static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
531
{
532
    int j,k;
533
    int ch;
534
    int run, case_val;
535
    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};
536

    
537
    for (ch = 0; ch < channels; ch++) {
538
        for (j = 0; j < 64; ) {
539
            if((coding_method[ch][sb][j] - 8) > 22) {
540
                run = 1;
541
                case_val = 8;
542
            } else {
543
                switch (switchtable[coding_method[ch][sb][j]-8]) {
544
                    case 0: run = 10; case_val = 10; break;
545
                    case 1: run = 1; case_val = 16; break;
546
                    case 2: run = 5; case_val = 24; break;
547
                    case 3: run = 3; case_val = 30; break;
548
                    case 4: run = 1; case_val = 30; break;
549
                    case 5: run = 1; case_val = 8; break;
550
                    default: run = 1; case_val = 8; break;
551
                }
552
            }
553
            for (k = 0; k < run; k++)
554
                if (j + k < 128)
555
                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
556
                        if (k > 0) {
557
                           SAMPLES_NEEDED
558
                            //not debugged, almost never used
559
                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
560
                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
561
                        }
562
            j += run;
563
        }
564
    }
565
}
566

    
567

    
568
/**
569
 * Related to synthesis filter
570
 * Called by process_subpacket_10
571
 *
572
 * @param q       context
573
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
574
 */
575
static void fill_tone_level_array (QDM2Context *q, int flag)
576
{
577
    int i, sb, ch, sb_used;
578
    int tmp, tab;
579

    
580
    // This should never happen
581
    if (q->nb_channels <= 0)
582
        return;
583

    
584
    for (ch = 0; ch < q->nb_channels; ch++)
585
        for (sb = 0; sb < 30; sb++)
586
            for (i = 0; i < 8; i++) {
587
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
588
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
589
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
590
                else
591
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
592
                if(tmp < 0)
593
                    tmp += 0xff;
594
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
595
            }
596

    
597
    sb_used = QDM2_SB_USED(q->sub_sampling);
598

    
599
    if ((q->superblocktype_2_3 != 0) && !flag) {
600
        for (sb = 0; sb < sb_used; sb++)
601
            for (ch = 0; ch < q->nb_channels; ch++)
602
                for (i = 0; i < 64; i++) {
603
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
604
                    if (q->tone_level_idx[ch][sb][i] < 0)
605
                        q->tone_level[ch][sb][i] = 0;
606
                    else
607
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
608
                }
609
    } else {
610
        tab = q->superblocktype_2_3 ? 0 : 1;
611
        for (sb = 0; sb < sb_used; sb++) {
612
            if ((sb >= 4) && (sb <= 23)) {
613
                for (ch = 0; ch < q->nb_channels; ch++)
614
                    for (i = 0; i < 64; i++) {
615
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
616
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
617
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
618
                              q->tone_level_idx_hi2[ch][sb - 4];
619
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
620
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
621
                            q->tone_level[ch][sb][i] = 0;
622
                        else
623
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
624
                }
625
            } else {
626
                if (sb > 4) {
627
                    for (ch = 0; ch < q->nb_channels; ch++)
628
                        for (i = 0; i < 64; i++) {
629
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
630
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
631
                                  q->tone_level_idx_hi2[ch][sb - 4];
632
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
633
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
634
                                q->tone_level[ch][sb][i] = 0;
635
                            else
636
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
637
                    }
638
                } else {
639
                    for (ch = 0; ch < q->nb_channels; ch++)
640
                        for (i = 0; i < 64; i++) {
641
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
642
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
643
                                q->tone_level[ch][sb][i] = 0;
644
                            else
645
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
646
                        }
647
                }
648
            }
649
        }
650
    }
651

    
652
    return;
653
}
654

    
655

    
656
/**
657
 * Related to synthesis filter
658
 * Called by process_subpacket_11
659
 * c is built with data from subpacket 11
660
 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
661
 *
662
 * @param tone_level_idx
663
 * @param tone_level_idx_temp
664
 * @param coding_method        q->coding_method[0][0][0]
665
 * @param nb_channels          number of channels
666
 * @param c                    coming from subpacket 11, passed as 8*c
667
 * @param superblocktype_2_3   flag based on superblock packet type
668
 * @param cm_table_select      q->cm_table_select
669
 */
670
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
671
                sb_int8_array coding_method, int nb_channels,
672
                int c, int superblocktype_2_3, int cm_table_select)
673
{
674
    int ch, sb, j;
675
    int tmp, acc, esp_40, comp;
676
    int add1, add2, add3, add4;
677
    int64_t multres;
678

    
679
    // This should never happen
680
    if (nb_channels <= 0)
681
        return;
682

    
683
    if (!superblocktype_2_3) {
684
        /* This case is untested, no samples available */
685
        SAMPLES_NEEDED
686
        for (ch = 0; ch < nb_channels; ch++)
687
            for (sb = 0; sb < 30; sb++) {
688
                for (j = 1; j < 64; j++) {
689
                    add1 = tone_level_idx[ch][sb][j] - 10;
690
                    if (add1 < 0)
691
                        add1 = 0;
692
                    add2 = add3 = add4 = 0;
693
                    if (sb > 1) {
694
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
695
                        if (add2 < 0)
696
                            add2 = 0;
697
                    }
698
                    if (sb > 0) {
699
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
700
                        if (add3 < 0)
701
                            add3 = 0;
702
                    }
703
                    if (sb < 29) {
704
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
705
                        if (add4 < 0)
706
                            add4 = 0;
707
                    }
708
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
709
                    if (tmp < 0)
710
                        tmp = 0;
711
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
712
                }
713
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
714
            }
715
            acc = 0;
716
            for (ch = 0; ch < nb_channels; ch++)
717
                for (sb = 0; sb < 30; sb++)
718
                    for (j = 0; j < 64; j++)
719
                        acc += tone_level_idx_temp[ch][sb][j];
720
            if (acc)
721
                tmp = c * 256 / (acc & 0xffff);
722
            multres = 0x66666667 * (acc * 10);
723
            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
724
            for (ch = 0;  ch < nb_channels; ch++)
725
                for (sb = 0; sb < 30; sb++)
726
                    for (j = 0; j < 64; j++) {
727
                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
728
                        if (comp < 0)
729
                            comp += 0xff;
730
                        comp /= 256; // signed shift
731
                        switch(sb) {
732
                            case 0:
733
                                if (comp < 30)
734
                                    comp = 30;
735
                                comp += 15;
736
                                break;
737
                            case 1:
738
                                if (comp < 24)
739
                                    comp = 24;
740
                                comp += 10;
741
                                break;
742
                            case 2:
743
                            case 3:
744
                            case 4:
745
                                if (comp < 16)
746
                                    comp = 16;
747
                        }
748
                        if (comp <= 5)
749
                            tmp = 0;
750
                        else if (comp <= 10)
751
                            tmp = 10;
752
                        else if (comp <= 16)
753
                            tmp = 16;
754
                        else if (comp <= 24)
755
                            tmp = -1;
756
                        else
757
                            tmp = 0;
758
                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
759
                    }
760
            for (sb = 0; sb < 30; sb++)
761
                fix_coding_method_array(sb, nb_channels, coding_method);
762
            for (ch = 0; ch < nb_channels; ch++)
763
                for (sb = 0; sb < 30; sb++)
764
                    for (j = 0; j < 64; j++)
765
                        if (sb >= 10) {
766
                            if (coding_method[ch][sb][j] < 10)
767
                                coding_method[ch][sb][j] = 10;
768
                        } else {
769
                            if (sb >= 2) {
770
                                if (coding_method[ch][sb][j] < 16)
771
                                    coding_method[ch][sb][j] = 16;
772
                            } else {
773
                                if (coding_method[ch][sb][j] < 30)
774
                                    coding_method[ch][sb][j] = 30;
775
                            }
776
                        }
777
    } else { // superblocktype_2_3 != 0
778
        for (ch = 0; ch < nb_channels; ch++)
779
            for (sb = 0; sb < 30; sb++)
780
                for (j = 0; j < 64; j++)
781
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
782
    }
783

    
784
    return;
785
}
786

    
787

    
788
/**
789
 *
790
 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
791
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
792
 *
793
 * @param q         context
794
 * @param gb        bitreader context
795
 * @param length    packet length in bits
796
 * @param sb_min    lower subband processed (sb_min included)
797
 * @param sb_max    higher subband processed (sb_max excluded)
798
 */
799
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
800
{
801
    int sb, j, k, n, ch, run, channels;
802
    int joined_stereo, zero_encoding, chs;
803
    int type34_first;
804
    float type34_div = 0;
805
    float type34_predictor;
806
    float samples[10], sign_bits[16];
807

    
808
    if (length == 0) {
809
        // If no data use noise
810
        for (sb=sb_min; sb < sb_max; sb++)
811
            build_sb_samples_from_noise (q, sb);
812

    
813
        return;
814
    }
815

    
816
    for (sb = sb_min; sb < sb_max; sb++) {
817
        FIX_NOISE_IDX(q->noise_idx);
818

    
819
        channels = q->nb_channels;
820

    
821
        if (q->nb_channels <= 1 || sb < 12)
822
            joined_stereo = 0;
823
        else if (sb >= 24)
824
            joined_stereo = 1;
825
        else
826
            joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
827

    
828
        if (joined_stereo) {
829
            if (BITS_LEFT(length,gb) >= 16)
830
                for (j = 0; j < 16; j++)
831
                    sign_bits[j] = get_bits1 (gb);
832

    
833
            for (j = 0; j < 64; j++)
834
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
835
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
836

    
837
            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
838
            channels = 1;
839
        }
840

    
841
        for (ch = 0; ch < channels; ch++) {
842
            zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
843
            type34_predictor = 0.0;
844
            type34_first = 1;
845

    
846
            for (j = 0; j < 128; ) {
847
                switch (q->coding_method[ch][sb][j / 2]) {
848
                    case 8:
849
                        if (BITS_LEFT(length,gb) >= 10) {
850
                            if (zero_encoding) {
851
                                for (k = 0; k < 5; k++) {
852
                                    if ((j + 2 * k) >= 128)
853
                                        break;
854
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
855
                                }
856
                            } else {
857
                                n = get_bits(gb, 8);
858
                                for (k = 0; k < 5; k++)
859
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
860
                            }
861
                            for (k = 0; k < 5; k++)
862
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
863
                        } else {
864
                            for (k = 0; k < 10; k++)
865
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
866
                        }
867
                        run = 10;
868
                        break;
869

    
870
                    case 10:
871
                        if (BITS_LEFT(length,gb) >= 1) {
872
                            float f = 0.81;
873

    
874
                            if (get_bits1(gb))
875
                                f = -f;
876
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
877
                            samples[0] = f;
878
                        } else {
879
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
880
                        }
881
                        run = 1;
882
                        break;
883

    
884
                    case 16:
885
                        if (BITS_LEFT(length,gb) >= 10) {
886
                            if (zero_encoding) {
887
                                for (k = 0; k < 5; k++) {
888
                                    if ((j + k) >= 128)
889
                                        break;
890
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
891
                                }
892
                            } else {
893
                                n = get_bits (gb, 8);
894
                                for (k = 0; k < 5; k++)
895
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
896
                            }
897
                        } else {
898
                            for (k = 0; k < 5; k++)
899
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
900
                        }
901
                        run = 5;
902
                        break;
903

    
904
                    case 24:
905
                        if (BITS_LEFT(length,gb) >= 7) {
906
                            n = get_bits(gb, 7);
907
                            for (k = 0; k < 3; k++)
908
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
909
                        } else {
910
                            for (k = 0; k < 3; k++)
911
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
912
                        }
913
                        run = 3;
914
                        break;
915

    
916
                    case 30:
917
                        if (BITS_LEFT(length,gb) >= 4)
918
                            samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
919
                        else
920
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
921

    
922
                        run = 1;
923
                        break;
924

    
925
                    case 34:
926
                        if (BITS_LEFT(length,gb) >= 7) {
927
                            if (type34_first) {
928
                                type34_div = (float)(1 << get_bits(gb, 2));
929
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
930
                                type34_predictor = samples[0];
931
                                type34_first = 0;
932
                            } else {
933
                                samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
934
                                type34_predictor = samples[0];
935
                            }
936
                        } else {
937
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
938
                        }
939
                        run = 1;
940
                        break;
941

    
942
                    default:
943
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
944
                        run = 1;
945
                        break;
946
                }
947

    
948
                if (joined_stereo) {
949
                    float tmp[10][MPA_MAX_CHANNELS];
950

    
951
                    for (k = 0; k < run; k++) {
952
                        tmp[k][0] = samples[k];
953
                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
954
                    }
955
                    for (chs = 0; chs < q->nb_channels; chs++)
956
                        for (k = 0; k < run; k++)
957
                            if ((j + k) < 128)
958
                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
959
                } else {
960
                    for (k = 0; k < run; k++)
961
                        if ((j + k) < 128)
962
                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
963
                }
964

    
965
                j += run;
966
            } // j loop
967
        } // channel loop
968
    } // subband loop
969
}
970

    
971

    
972
/**
973
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
974
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
975
 * same VLC tables as process_subpacket_9 are used.
976
 *
977
 * @param q         context
978
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
979
 * @param gb        bitreader context
980
 * @param length    packet length in bits
981
 */
982
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
983
{
984
    int i, k, run, level, diff;
985

    
986
    if (BITS_LEFT(length,gb) < 16)
987
        return;
988
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
989

    
990
    quantized_coeffs[0] = level;
991

    
992
    for (i = 0; i < 7; ) {
993
        if (BITS_LEFT(length,gb) < 16)
994
            break;
995
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
996

    
997
        if (BITS_LEFT(length,gb) < 16)
998
            break;
999
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1000

    
1001
        for (k = 1; k <= run; k++)
1002
            quantized_coeffs[i + k] = (level + ((k * diff) / run));
1003

    
1004
        level += diff;
1005
        i += run;
1006
    }
1007
}
1008

    
1009

    
1010
/**
1011
 * Related to synthesis filter, process data from packet 10
1012
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1013
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1014
 *
1015
 * @param q         context
1016
 * @param gb        bitreader context
1017
 * @param length    packet length in bits
1018
 */
1019
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1020
{
1021
    int sb, j, k, n, ch;
1022

    
1023
    for (ch = 0; ch < q->nb_channels; ch++) {
1024
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1025

    
1026
        if (BITS_LEFT(length,gb) < 16) {
1027
            memset(q->quantized_coeffs[ch][0], 0, 8);
1028
            break;
1029
        }
1030
    }
1031

    
1032
    n = q->sub_sampling + 1;
1033

    
1034
    for (sb = 0; sb < n; sb++)
1035
        for (ch = 0; ch < q->nb_channels; ch++)
1036
            for (j = 0; j < 8; j++) {
1037
                if (BITS_LEFT(length,gb) < 1)
1038
                    break;
1039
                if (get_bits1(gb)) {
1040
                    for (k=0; k < 8; k++) {
1041
                        if (BITS_LEFT(length,gb) < 16)
1042
                            break;
1043
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1044
                    }
1045
                } else {
1046
                    for (k=0; k < 8; k++)
1047
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1048
                }
1049
            }
1050

    
1051
    n = QDM2_SB_USED(q->sub_sampling) - 4;
1052

    
1053
    for (sb = 0; sb < n; sb++)
1054
        for (ch = 0; ch < q->nb_channels; ch++) {
1055
            if (BITS_LEFT(length,gb) < 16)
1056
                break;
1057
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1058
            if (sb > 19)
1059
                q->tone_level_idx_hi2[ch][sb] -= 16;
1060
            else
1061
                for (j = 0; j < 8; j++)
1062
                    q->tone_level_idx_mid[ch][sb][j] = -16;
1063
        }
1064

    
1065
    n = QDM2_SB_USED(q->sub_sampling) - 5;
1066

    
1067
    for (sb = 0; sb < n; sb++)
1068
        for (ch = 0; ch < q->nb_channels; ch++)
1069
            for (j = 0; j < 8; j++) {
1070
                if (BITS_LEFT(length,gb) < 16)
1071
                    break;
1072
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1073
            }
1074
}
1075

    
1076
/**
1077
 * Process subpacket 9, init quantized_coeffs with data from it
1078
 *
1079
 * @param q       context
1080
 * @param node    pointer to node with packet
1081
 */
1082
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1083
{
1084
    GetBitContext gb;
1085
    int i, j, k, n, ch, run, level, diff;
1086

    
1087
    init_get_bits(&gb, node->packet->data, node->packet->size*8);
1088

    
1089
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1090

    
1091
    for (i = 1; i < n; i++)
1092
        for (ch=0; ch < q->nb_channels; ch++) {
1093
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1094
            q->quantized_coeffs[ch][i][0] = level;
1095

    
1096
            for (j = 0; j < (8 - 1); ) {
1097
                run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1098
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1099

    
1100
                for (k = 1; k <= run; k++)
1101
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1102

    
1103
                level += diff;
1104
                j += run;
1105
            }
1106
        }
1107

    
1108
    for (ch = 0; ch < q->nb_channels; ch++)
1109
        for (i = 0; i < 8; i++)
1110
            q->quantized_coeffs[ch][0][i] = 0;
1111
}
1112

    
1113

    
1114
/**
1115
 * Process subpacket 10 if not null, else
1116
 *
1117
 * @param q         context
1118
 * @param node      pointer to node with packet
1119
 * @param length    packet length in bits
1120
 */
1121
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1122
{
1123
    GetBitContext gb;
1124

    
1125
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1126

    
1127
    if (length != 0) {
1128
        init_tone_level_dequantization(q, &gb, length);
1129
        fill_tone_level_array(q, 1);
1130
    } else {
1131
        fill_tone_level_array(q, 0);
1132
    }
1133
}
1134

    
1135

    
1136
/**
1137
 * Process subpacket 11
1138
 *
1139
 * @param q         context
1140
 * @param node      pointer to node with packet
1141
 * @param length    packet length in bit
1142
 */
1143
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1144
{
1145
    GetBitContext gb;
1146

    
1147
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1148
    if (length >= 32) {
1149
        int c = get_bits (&gb, 13);
1150

    
1151
        if (c > 3)
1152
            fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1153
                                      q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1154
    }
1155

    
1156
    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1157
}
1158

    
1159

    
1160
/**
1161
 * Process subpacket 12
1162
 *
1163
 * @param q         context
1164
 * @param node      pointer to node with packet
1165
 * @param length    packet length in bits
1166
 */
1167
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1168
{
1169
    GetBitContext gb;
1170

    
1171
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1172
    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1173
}
1174

    
1175
/*
1176
 * Process new subpackets for synthesis filter
1177
 *
1178
 * @param q       context
1179
 * @param list    list with synthesis filter packets (list D)
1180
 */
1181
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1182
{
1183
    QDM2SubPNode *nodes[4];
1184

    
1185
    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1186
    if (nodes[0] != NULL)
1187
        process_subpacket_9(q, nodes[0]);
1188

    
1189
    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1190
    if (nodes[1] != NULL)
1191
        process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1192
    else
1193
        process_subpacket_10(q, NULL, 0);
1194

    
1195
    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1196
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1197
        process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1198
    else
1199
        process_subpacket_11(q, NULL, 0);
1200

    
1201
    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1202
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1203
        process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1204
    else
1205
        process_subpacket_12(q, NULL, 0);
1206
}
1207

    
1208

    
1209
/*
1210
 * Decode superblock, fill packet lists.
1211
 *
1212
 * @param q    context
1213
 */
1214
static void qdm2_decode_super_block (QDM2Context *q)
1215
{
1216
    GetBitContext gb;
1217
    QDM2SubPacket header, *packet;
1218
    int i, packet_bytes, sub_packet_size, sub_packets_D;
1219
    unsigned int next_index = 0;
1220

    
1221
    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1222
    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1223
    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1224

    
1225
    q->sub_packets_B = 0;
1226
    sub_packets_D = 0;
1227

    
1228
    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1229

    
1230
    init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1231
    qdm2_decode_sub_packet_header(&gb, &header);
1232

    
1233
    if (header.type < 2 || header.type >= 8) {
1234
        q->has_errors = 1;
1235
        av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1236
        return;
1237
    }
1238

    
1239
    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1240
    packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1241

    
1242
    init_get_bits(&gb, header.data, header.size*8);
1243

    
1244
    if (header.type == 2 || header.type == 4 || header.type == 5) {
1245
        int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1246

    
1247
        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1248

    
1249
        if (csum != 0) {
1250
            q->has_errors = 1;
1251
            av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1252
            return;
1253
        }
1254
    }
1255

    
1256
    q->sub_packet_list_B[0].packet = NULL;
1257
    q->sub_packet_list_D[0].packet = NULL;
1258

    
1259
    for (i = 0; i < 6; i++)
1260
        if (--q->fft_level_exp[i] < 0)
1261
            q->fft_level_exp[i] = 0;
1262

    
1263
    for (i = 0; packet_bytes > 0; i++) {
1264
        int j;
1265

    
1266
        q->sub_packet_list_A[i].next = NULL;
1267

    
1268
        if (i > 0) {
1269
            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1270

    
1271
            /* seek to next block */
1272
            init_get_bits(&gb, header.data, header.size*8);
1273
            skip_bits(&gb, next_index*8);
1274

    
1275
            if (next_index >= header.size)
1276
                break;
1277
        }
1278

    
1279
        /* decode subpacket */
1280
        packet = &q->sub_packets[i];
1281
        qdm2_decode_sub_packet_header(&gb, packet);
1282
        next_index = packet->size + get_bits_count(&gb) / 8;
1283
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1284

    
1285
        if (packet->type == 0)
1286
            break;
1287

    
1288
        if (sub_packet_size > packet_bytes) {
1289
            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1290
                break;
1291
            packet->size += packet_bytes - sub_packet_size;
1292
        }
1293

    
1294
        packet_bytes -= sub_packet_size;
1295

    
1296
        /* add subpacket to 'all subpackets' list */
1297
        q->sub_packet_list_A[i].packet = packet;
1298

    
1299
        /* add subpacket to related list */
1300
        if (packet->type == 8) {
1301
            SAMPLES_NEEDED_2("packet type 8");
1302
            return;
1303
        } else if (packet->type >= 9 && packet->type <= 12) {
1304
            /* packets for MPEG Audio like Synthesis Filter */
1305
            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1306
        } else if (packet->type == 13) {
1307
            for (j = 0; j < 6; j++)
1308
                q->fft_level_exp[j] = get_bits(&gb, 6);
1309
        } else if (packet->type == 14) {
1310
            for (j = 0; j < 6; j++)
1311
                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1312
        } else if (packet->type == 15) {
1313
            SAMPLES_NEEDED_2("packet type 15")
1314
            return;
1315
        } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1316
            /* packets for FFT */
1317
            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1318
        }
1319
    } // Packet bytes loop
1320

    
1321
/* **************************************************************** */
1322
    if (q->sub_packet_list_D[0].packet != NULL) {
1323
        process_synthesis_subpackets(q, q->sub_packet_list_D);
1324
        q->do_synth_filter = 1;
1325
    } else if (q->do_synth_filter) {
1326
        process_subpacket_10(q, NULL, 0);
1327
        process_subpacket_11(q, NULL, 0);
1328
        process_subpacket_12(q, NULL, 0);
1329
    }
1330
/* **************************************************************** */
1331
}
1332

    
1333

    
1334
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1335
                       int offset, int duration, int channel,
1336
                       int exp, int phase)
1337
{
1338
    if (q->fft_coefs_min_index[duration] < 0)
1339
        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1340

    
1341
    q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1342
    q->fft_coefs[q->fft_coefs_index].channel = channel;
1343
    q->fft_coefs[q->fft_coefs_index].offset = offset;
1344
    q->fft_coefs[q->fft_coefs_index].exp = exp;
1345
    q->fft_coefs[q->fft_coefs_index].phase = phase;
1346
    q->fft_coefs_index++;
1347
}
1348

    
1349

    
1350
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1351
{
1352
    int channel, stereo, phase, exp;
1353
    int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
1354
    int local_int_14, stereo_exp, local_int_20, local_int_28;
1355
    int n, offset;
1356

    
1357
    local_int_4 = 0;
1358
    local_int_28 = 0;
1359
    local_int_20 = 2;
1360
    local_int_8 = (4 - duration);
1361
    local_int_10 = 1 << (q->group_order - duration - 1);
1362
    offset = 1;
1363

    
1364
    while (1) {
1365
        if (q->superblocktype_2_3) {
1366
            while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1367
                offset = 1;
1368
                if (n == 0) {
1369
                    local_int_4 += local_int_10;
1370
                    local_int_28 += (1 << local_int_8);
1371
                } else {
1372
                    local_int_4 += 8*local_int_10;
1373
                    local_int_28 += (8 << local_int_8);
1374
                }
1375
            }
1376
            offset += (n - 2);
1377
        } else {
1378
            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1379
            while (offset >= (local_int_10 - 1)) {
1380
                offset += (1 - (local_int_10 - 1));
1381
                local_int_4  += local_int_10;
1382
                local_int_28 += (1 << local_int_8);
1383
            }
1384
        }
1385

    
1386
        if (local_int_4 >= q->group_size)
1387
            return;
1388

    
1389
        local_int_14 = (offset >> local_int_8);
1390

    
1391
        if (q->nb_channels > 1) {
1392
            channel = get_bits1(gb);
1393
            stereo = get_bits1(gb);
1394
        } else {
1395
            channel = 0;
1396
            stereo = 0;
1397
        }
1398

    
1399
        exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1400
        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1401
        exp = (exp < 0) ? 0 : exp;
1402

    
1403
        phase = get_bits(gb, 3);
1404
        stereo_exp = 0;
1405
        stereo_phase = 0;
1406

    
1407
        if (stereo) {
1408
            stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1409
            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1410
            if (stereo_phase < 0)
1411
                stereo_phase += 8;
1412
        }
1413

    
1414
        if (q->frequency_range > (local_int_14 + 1)) {
1415
            int sub_packet = (local_int_20 + local_int_28);
1416

    
1417
            qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1418
            if (stereo)
1419
                qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1420
        }
1421

    
1422
        offset++;
1423
    }
1424
}
1425

    
1426

    
1427
static void qdm2_decode_fft_packets (QDM2Context *q)
1428
{
1429
    int i, j, min, max, value, type, unknown_flag;
1430
    GetBitContext gb;
1431

    
1432
    if (q->sub_packet_list_B[0].packet == NULL)
1433
        return;
1434

    
1435
    /* reset minimum indices for FFT coefficients */
1436
    q->fft_coefs_index = 0;
1437
    for (i=0; i < 5; i++)
1438
        q->fft_coefs_min_index[i] = -1;
1439

    
1440
    /* process subpackets ordered by type, largest type first */
1441
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1442
        QDM2SubPacket *packet;
1443

    
1444
        /* find subpacket with largest type less than max */
1445
        for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1446
            value = q->sub_packet_list_B[j].packet->type;
1447
            if (value > min && value < max) {
1448
                min = value;
1449
                packet = q->sub_packet_list_B[j].packet;
1450
            }
1451
        }
1452

    
1453
        max = min;
1454

    
1455
        /* check for errors (?) */
1456
        if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1457
            return;
1458

    
1459
        /* decode FFT tones */
1460
        init_get_bits (&gb, packet->data, packet->size*8);
1461

    
1462
        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1463
            unknown_flag = 1;
1464
        else
1465
            unknown_flag = 0;
1466

    
1467
        type = packet->type;
1468

    
1469
        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1470
            int duration = q->sub_sampling + 5 - (type & 15);
1471

    
1472
            if (duration >= 0 && duration < 4)
1473
                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1474
        } else if (type == 31) {
1475
            for (j=0; j < 4; j++)
1476
                qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1477
        } else if (type == 46) {
1478
            for (j=0; j < 6; j++)
1479
                q->fft_level_exp[j] = get_bits(&gb, 6);
1480
            for (j=0; j < 4; j++)
1481
            qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1482
        }
1483
    } // Loop on B packets
1484

    
1485
    /* calculate maximum indices for FFT coefficients */
1486
    for (i = 0, j = -1; i < 5; i++)
1487
        if (q->fft_coefs_min_index[i] >= 0) {
1488
            if (j >= 0)
1489
                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1490
            j = i;
1491
        }
1492
    if (j >= 0)
1493
        q->fft_coefs_max_index[j] = q->fft_coefs_index;
1494
}
1495

    
1496

    
1497
static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1498
{
1499
   float level, f[6];
1500
   int i;
1501
   QDM2Complex c;
1502
   const double iscale = 2.0*M_PI / 512.0;
1503

    
1504
    tone->phase += tone->phase_shift;
1505

    
1506
    /* calculate current level (maximum amplitude) of tone */
1507
    level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1508
    c.im = level * sin(tone->phase*iscale);
1509
    c.re = level * cos(tone->phase*iscale);
1510

    
1511
    /* generate FFT coefficients for tone */
1512
    if (tone->duration >= 3 || tone->cutoff >= 3) {
1513
        tone->samples_im[0] += c.im;
1514
        tone->samples_re[0] += c.re;
1515
        tone->samples_im[1] -= c.im;
1516
        tone->samples_re[1] -= c.re;
1517
    } else {
1518
        f[1] = -tone->table[4];
1519
        f[0] =  tone->table[3] - tone->table[0];
1520
        f[2] =  1.0 - tone->table[2] - tone->table[3];
1521
        f[3] =  tone->table[1] + tone->table[4] - 1.0;
1522
        f[4] =  tone->table[0] - tone->table[1];
1523
        f[5] =  tone->table[2];
1524
        for (i = 0; i < 2; i++) {
1525
            tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1526
            tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1527
        }
1528
        for (i = 0; i < 4; i++) {
1529
            tone->samples_re[i] += c.re * f[i+2];
1530
            tone->samples_im[i] += c.im * f[i+2];
1531
        }
1532
    }
1533

    
1534
    /* copy the tone if it has not yet died out */
1535
    if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1536
      memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1537
      q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1538
    }
1539
}
1540

    
1541

    
1542
static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1543
{
1544
    int i, j, ch;
1545
    const double iscale = 0.25 * M_PI;
1546

    
1547
    for (ch = 0; ch < q->channels; ch++) {
1548
        memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1549
        memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1550
    }
1551

    
1552

    
1553
    /* apply FFT tones with duration 4 (1 FFT period) */
1554
    if (q->fft_coefs_min_index[4] >= 0)
1555
        for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1556
            float level;
1557
            QDM2Complex c;
1558

    
1559
            if (q->fft_coefs[i].sub_packet != sub_packet)
1560
                break;
1561

    
1562
            ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1563
            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];
1564

    
1565
            c.re = level * cos(q->fft_coefs[i].phase * iscale);
1566
            c.im = level * sin(q->fft_coefs[i].phase * iscale);
1567
            q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1568
            q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1569
            q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1570
            q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1571
        }
1572

    
1573
    /* generate existing FFT tones */
1574
    for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1575
        qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1576
        q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1577
    }
1578

    
1579
    /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1580
    for (i = 0; i < 4; i++)
1581
        if (q->fft_coefs_min_index[i] >= 0) {
1582
            for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1583
                int offset, four_i;
1584
                FFTTone tone;
1585

    
1586
                if (q->fft_coefs[j].sub_packet != sub_packet)
1587
                    break;
1588

    
1589
                four_i = (4 - i);
1590
                offset = q->fft_coefs[j].offset >> four_i;
1591
                ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1592

    
1593
                if (offset < q->frequency_range) {
1594
                    if (offset < 2)
1595
                        tone.cutoff = offset;
1596
                    else
1597
                        tone.cutoff = (offset >= 60) ? 3 : 2;
1598

    
1599
                    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];
1600
                    tone.samples_im = &q->fft.samples_im[ch][offset];
1601
                    tone.samples_re = &q->fft.samples_re[ch][offset];
1602
                    tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1603
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1604
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1605
                    tone.duration = i;
1606
                    tone.time_index = 0;
1607

    
1608
                    qdm2_fft_generate_tone(q, &tone);
1609
                }
1610
            }
1611
            q->fft_coefs_min_index[i] = j;
1612
        }
1613
}
1614

    
1615

    
1616
static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1617
{
1618
    const int n = 1 << (q->fft_order - 1);
1619
    const int n2 = n >> 1;
1620
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1621
    float c, s, f0, f1, f2, f3;
1622
    int i, j;
1623

    
1624
    /* prerotation (or something like that) */
1625
    for (i=1; i < n2; i++) {
1626
        j  = (n - i);
1627
        c = q->exptab[i].re;
1628
        s = -q->exptab[i].im;
1629
        f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1630
        f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1631
        f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1632
        f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1633
        q->fft.complex[i].re =  s * f0 - c * f1 + f2;
1634
        q->fft.complex[i].im =  c * f0 + s * f1 + f3;
1635
        q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1636
        q->fft.complex[j].im =  c * f0 + s * f1 - f3;
1637
    }
1638

    
1639
    q->fft.complex[ 0].re =  q->fft.samples_re[channel][ 0] * gain * 2.0;
1640
    q->fft.complex[ 0].im =  q->fft.samples_re[channel][ 0] * gain * 2.0;
1641
    q->fft.complex[n2].re =  q->fft.samples_re[channel][n2] * gain * 2.0;
1642
    q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1643

    
1644
    ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1645
    ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1646
    /* add samples to output buffer */
1647
    for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1648
        q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1649
}
1650

    
1651

    
1652
/**
1653
 * @param q        context
1654
 * @param index    subpacket number
1655
 */
1656
static void qdm2_synthesis_filter (QDM2Context *q, int index)
1657
{
1658
    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1659
    int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1660

    
1661
    /* copy sb_samples */
1662
    sb_used = QDM2_SB_USED(q->sub_sampling);
1663

    
1664
    for (ch = 0; ch < q->channels; ch++)
1665
        for (i = 0; i < 8; i++)
1666
            for (k=sb_used; k < SBLIMIT; k++)
1667
                q->sb_samples[ch][(8 * index) + i][k] = 0;
1668

    
1669
    for (ch = 0; ch < q->nb_channels; ch++) {
1670
        OUT_INT *samples_ptr = samples + ch;
1671

    
1672
        for (i = 0; i < 8; i++) {
1673
            ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1674
                mpa_window, &dither_state,
1675
                samples_ptr, q->nb_channels,
1676
                q->sb_samples[ch][(8 * index) + i]);
1677
            samples_ptr += 32 * q->nb_channels;
1678
        }
1679
    }
1680

    
1681
    /* add samples to output buffer */
1682
    sub_sampling = (4 >> q->sub_sampling);
1683

    
1684
    for (ch = 0; ch < q->channels; ch++)
1685
        for (i = 0; i < q->frame_size; i++)
1686
            q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1687
}
1688

    
1689

    
1690
/**
1691
 * Init static data (does not depend on specific file)
1692
 *
1693
 * @param q    context
1694
 */
1695
static void qdm2_init(QDM2Context *q) {
1696
    static int inited = 0;
1697

    
1698
    if (inited != 0)
1699
        return;
1700
    inited = 1;
1701

    
1702
    qdm2_init_vlc();
1703
    ff_mpa_synth_init(mpa_window);
1704
    softclip_table_init();
1705
    rnd_table_init();
1706
    init_noise_samples();
1707

    
1708
    av_log(NULL, AV_LOG_DEBUG, "init done\n");
1709
}
1710

    
1711

    
1712
#if 0
1713
static void dump_context(QDM2Context *q)
1714
{
1715
    int i;
1716
#define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1717
    PRINT("compressed_data",q->compressed_data);
1718
    PRINT("compressed_size",q->compressed_size);
1719
    PRINT("frame_size",q->frame_size);
1720
    PRINT("checksum_size",q->checksum_size);
1721
    PRINT("channels",q->channels);
1722
    PRINT("nb_channels",q->nb_channels);
1723
    PRINT("fft_frame_size",q->fft_frame_size);
1724
    PRINT("fft_size",q->fft_size);
1725
    PRINT("sub_sampling",q->sub_sampling);
1726
    PRINT("fft_order",q->fft_order);
1727
    PRINT("group_order",q->group_order);
1728
    PRINT("group_size",q->group_size);
1729
    PRINT("sub_packet",q->sub_packet);
1730
    PRINT("frequency_range",q->frequency_range);
1731
    PRINT("has_errors",q->has_errors);
1732
    PRINT("fft_tone_end",q->fft_tone_end);
1733
    PRINT("fft_tone_start",q->fft_tone_start);
1734
    PRINT("fft_coefs_index",q->fft_coefs_index);
1735
    PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1736
    PRINT("cm_table_select",q->cm_table_select);
1737
    PRINT("noise_idx",q->noise_idx);
1738

1739
    for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1740
    {
1741
    FFTTone *t = &q->fft_tones[i];
1742

1743
    av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1744
    av_log(NULL,AV_LOG_DEBUG,"  level = %f\n", t->level);
1745
//  PRINT(" level", t->level);
1746
    PRINT(" phase", t->phase);
1747
    PRINT(" phase_shift", t->phase_shift);
1748
    PRINT(" duration", t->duration);
1749
    PRINT(" samples_im", t->samples_im);
1750
    PRINT(" samples_re", t->samples_re);
1751
    PRINT(" table", t->table);
1752
    }
1753

1754
}
1755
#endif
1756

    
1757

    
1758
/**
1759
 * Init parameters from codec extradata
1760
 */
1761
static int qdm2_decode_init(AVCodecContext *avctx)
1762
{
1763
    QDM2Context *s = avctx->priv_data;
1764
    uint8_t *extradata;
1765
    int extradata_size;
1766
    int tmp_val, tmp, size;
1767
    int i;
1768
    float alpha;
1769

    
1770
    /* extradata parsing
1771

1772
    Structure:
1773
    wave {
1774
        frma (QDM2)
1775
        QDCA
1776
        QDCP
1777
    }
1778

1779
    32  size (including this field)
1780
    32  tag (=frma)
1781
    32  type (=QDM2 or QDMC)
1782

1783
    32  size (including this field, in bytes)
1784
    32  tag (=QDCA) // maybe mandatory parameters
1785
    32  unknown (=1)
1786
    32  channels (=2)
1787
    32  samplerate (=44100)
1788
    32  bitrate (=96000)
1789
    32  block size (=4096)
1790
    32  frame size (=256) (for one channel)
1791
    32  packet size (=1300)
1792

1793
    32  size (including this field, in bytes)
1794
    32  tag (=QDCP) // maybe some tuneable parameters
1795
    32  float1 (=1.0)
1796
    32  zero ?
1797
    32  float2 (=1.0)
1798
    32  float3 (=1.0)
1799
    32  unknown (27)
1800
    32  unknown (8)
1801
    32  zero ?
1802
    */
1803

    
1804
    if (!avctx->extradata || (avctx->extradata_size < 48)) {
1805
        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1806
        return -1;
1807
    }
1808

    
1809
    extradata = avctx->extradata;
1810
    extradata_size = avctx->extradata_size;
1811

    
1812
    while (extradata_size > 7) {
1813
        if (!memcmp(extradata, "frmaQDM", 7))
1814
            break;
1815
        extradata++;
1816
        extradata_size--;
1817
    }
1818

    
1819
    if (extradata_size < 12) {
1820
        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1821
               extradata_size);
1822
        return -1;
1823
    }
1824

    
1825
    if (memcmp(extradata, "frmaQDM", 7)) {
1826
        av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1827
        return -1;
1828
    }
1829

    
1830
    if (extradata[7] == 'C') {
1831
//        s->is_qdmc = 1;
1832
        av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1833
        return -1;
1834
    }
1835

    
1836
    extradata += 8;
1837
    extradata_size -= 8;
1838

    
1839
    size = BE_32(extradata);
1840

    
1841
    if(size > extradata_size){
1842
        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1843
               extradata_size, size);
1844
        return -1;
1845
    }
1846

    
1847
    extradata += 4;
1848
    av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1849
    if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
1850
        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1851
        return -1;
1852
    }
1853

    
1854
    extradata += 8;
1855

    
1856
    avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
1857
    extradata += 4;
1858

    
1859
    avctx->sample_rate = BE_32(extradata);
1860
    extradata += 4;
1861

    
1862
    avctx->bit_rate = BE_32(extradata);
1863
    extradata += 4;
1864

    
1865
    s->group_size = BE_32(extradata);
1866
    extradata += 4;
1867

    
1868
    s->fft_size = BE_32(extradata);
1869
    extradata += 4;
1870

    
1871
    s->checksum_size = BE_32(extradata);
1872
    extradata += 4;
1873

    
1874
    s->fft_order = av_log2(s->fft_size) + 1;
1875
    s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1876

    
1877
    // something like max decodable tones
1878
    s->group_order = av_log2(s->group_size) + 1;
1879
    s->frame_size = s->group_size / 16; // 16 iterations per super block
1880

    
1881
    s->sub_sampling = s->fft_order - 7;
1882
    s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1883

    
1884
    switch ((s->sub_sampling * 2 + s->channels - 1)) {
1885
        case 0: tmp = 40; break;
1886
        case 1: tmp = 48; break;
1887
        case 2: tmp = 56; break;
1888
        case 3: tmp = 72; break;
1889
        case 4: tmp = 80; break;
1890
        case 5: tmp = 100;break;
1891
        default: tmp=s->sub_sampling; break;
1892
    }
1893
    tmp_val = 0;
1894
    if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
1895
    if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
1896
    if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
1897
    if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
1898
    s->cm_table_select = tmp_val;
1899

    
1900
    if (s->sub_sampling == 0)
1901
        tmp = 7999;
1902
    else
1903
        tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1904
    /*
1905
    0: 7999 -> 0
1906
    1: 20000 -> 2
1907
    2: 28000 -> 2
1908
    */
1909
    if (tmp < 8000)
1910
        s->coeff_per_sb_select = 0;
1911
    else if (tmp <= 16000)
1912
        s->coeff_per_sb_select = 1;
1913
    else
1914
        s->coeff_per_sb_select = 2;
1915

    
1916
    // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
1917
    if ((s->fft_order < 7) || (s->fft_order > 9)) {
1918
        av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1919
        return -1;
1920
    }
1921

    
1922
    ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1923

    
1924
    for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1925
        alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1926
        s->exptab[i].re = cos(alpha);
1927
        s->exptab[i].im = sin(alpha);
1928
    }
1929

    
1930
    qdm2_init(s);
1931

    
1932
//    dump_context(s);
1933
    return 0;
1934
}
1935

    
1936

    
1937
static int qdm2_decode_close(AVCodecContext *avctx)
1938
{
1939
    QDM2Context *s = avctx->priv_data;
1940

    
1941
    ff_fft_end(&s->fft_ctx);
1942

    
1943
    return 0;
1944
}
1945

    
1946

    
1947
static void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
1948
{
1949
    int ch, i;
1950
    const int frame_size = (q->frame_size * q->channels);
1951

    
1952
    /* select input buffer */
1953
    q->compressed_data = in;
1954
    q->compressed_size = q->checksum_size;
1955

    
1956
//  dump_context(q);
1957

    
1958
    /* copy old block, clear new block of output samples */
1959
    memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1960
    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1961

    
1962
    /* decode block of QDM2 compressed data */
1963
    if (q->sub_packet == 0) {
1964
        q->has_errors = 0; // zero it for a new super block
1965
        av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1966
        qdm2_decode_super_block(q);
1967
    }
1968

    
1969
    /* parse subpackets */
1970
    if (!q->has_errors) {
1971
        if (q->sub_packet == 2)
1972
            qdm2_decode_fft_packets(q);
1973

    
1974
        qdm2_fft_tone_synthesizer(q, q->sub_packet);
1975
    }
1976

    
1977
    /* sound synthesis stage 1 (FFT) */
1978
    for (ch = 0; ch < q->channels; ch++) {
1979
        qdm2_calculate_fft(q, ch, q->sub_packet);
1980

    
1981
        if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1982
            SAMPLES_NEEDED_2("has errors, and C list is not empty")
1983
            return;
1984
        }
1985
    }
1986

    
1987
    /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1988
    if (!q->has_errors && q->do_synth_filter)
1989
        qdm2_synthesis_filter(q, q->sub_packet);
1990

    
1991
    q->sub_packet = (q->sub_packet + 1) % 16;
1992

    
1993
    /* clip and convert output float[] to 16bit signed samples */
1994
    for (i = 0; i < frame_size; i++) {
1995
        int value = (int)q->output_buffer[i];
1996

    
1997
        if (value > SOFTCLIP_THRESHOLD)
1998
            value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
1999
        else if (value < -SOFTCLIP_THRESHOLD)
2000
            value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2001

    
2002
        out[i] = value;
2003
    }
2004
}
2005

    
2006

    
2007
static int qdm2_decode_frame(AVCodecContext *avctx,
2008
            void *data, int *data_size,
2009
            uint8_t *buf, int buf_size)
2010
{
2011
    QDM2Context *s = avctx->priv_data;
2012

    
2013
    if(!buf)
2014
        return 0;
2015
    if(buf_size < s->checksum_size)
2016
        return -1;
2017

    
2018
    *data_size = s->channels * s->frame_size * sizeof(int16_t);
2019

    
2020
    av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2021
       buf_size, buf, s->checksum_size, data, *data_size);
2022

    
2023
    qdm2_decode(s, buf, data);
2024

    
2025
    // reading only when next superblock found
2026
    if (s->sub_packet == 0) {
2027
        return s->checksum_size;
2028
    }
2029

    
2030
    return 0;
2031
}
2032

    
2033
AVCodec qdm2_decoder =
2034
{
2035
    .name = "qdm2",
2036
    .type = CODEC_TYPE_AUDIO,
2037
    .id = CODEC_ID_QDM2,
2038
    .priv_data_size = sizeof(QDM2Context),
2039
    .init = qdm2_decode_init,
2040
    .close = qdm2_decode_close,
2041
    .decode = qdm2_decode_frame,
2042
};