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/*
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 * Windows Media Audio Voice decoder.
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 * Copyright (c) 2009 Ronald S. Bultje
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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/**
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 * @file
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 * @brief Windows Media Audio Voice compatible decoder
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 * @author Ronald S. Bultje <rsbultje@gmail.com>
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 */
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#include <math.h>
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#include "avcodec.h"
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#include "get_bits.h"
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#include "put_bits.h"
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#include "wmavoice_data.h"
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#include "celp_math.h"
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#include "celp_filters.h"
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#include "acelp_vectors.h"
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#include "acelp_filters.h"
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#include "lsp.h"
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#include "libavutil/lzo.h"
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#include "avfft.h"
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#include "fft.h"
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#define MAX_BLOCKS           8   ///< maximum number of blocks per frame
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#define MAX_LSPS             16  ///< maximum filter order
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#define MAX_LSPS_ALIGN16     16  ///< same as #MAX_LSPS; needs to be multiple
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                                 ///< of 16 for ASM input buffer alignment
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#define MAX_FRAMES           3   ///< maximum number of frames per superframe
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#define MAX_FRAMESIZE        160 ///< maximum number of samples per frame
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#define MAX_SIGNAL_HISTORY   416 ///< maximum excitation signal history
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#define MAX_SFRAMESIZE       (MAX_FRAMESIZE * MAX_FRAMES)
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                                 ///< maximum number of samples per superframe
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#define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
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                                 ///< was split over two packets
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#define VLC_NBITS            6   ///< number of bits to read per VLC iteration
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55
/**
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 * Frame type VLC coding.
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 */
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static VLC frame_type_vlc;
59

    
60
/**
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 * Adaptive codebook types.
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 */
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enum {
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    ACB_TYPE_NONE       = 0, ///< no adaptive codebook (only hardcoded fixed)
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    ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
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                             ///< we interpolate to get a per-sample pitch.
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                             ///< Signal is generated using an asymmetric sinc
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                             ///< window function
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                             ///< @note see #wmavoice_ipol1_coeffs
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    ACB_TYPE_HAMMING    = 2  ///< Per-block pitch with signal generation using
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                             ///< a Hamming sinc window function
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                             ///< @note see #wmavoice_ipol2_coeffs
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};
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/**
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 * Fixed codebook types.
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 */
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enum {
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    FCB_TYPE_SILENCE    = 0, ///< comfort noise during silence
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                             ///< generated from a hardcoded (fixed) codebook
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                             ///< with per-frame (low) gain values
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    FCB_TYPE_HARDCODED  = 1, ///< hardcoded (fixed) codebook with per-block
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                             ///< gain values
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    FCB_TYPE_AW_PULSES  = 2, ///< Pitch-adaptive window (AW) pulse signals,
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                             ///< used in particular for low-bitrate streams
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    FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
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                             ///< combinations of either single pulses or
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                             ///< pulse pairs
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};
90

    
91
/**
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 * Description of frame types.
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 */
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static const struct frame_type_desc {
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    uint8_t n_blocks;     ///< amount of blocks per frame (each block
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                          ///< (contains 160/#n_blocks samples)
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    uint8_t log_n_blocks; ///< log2(#n_blocks)
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    uint8_t acb_type;     ///< Adaptive codebook type (ACB_TYPE_*)
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    uint8_t fcb_type;     ///< Fixed codebook type (FCB_TYPE_*)
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    uint8_t dbl_pulses;   ///< how many pulse vectors have pulse pairs
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                          ///< (rather than just one single pulse)
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                          ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
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    uint16_t frame_size;  ///< the amount of bits that make up the block
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                          ///< data (per frame)
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} frame_descs[17] = {
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    { 1, 0, ACB_TYPE_NONE,       FCB_TYPE_SILENCE,    0,   0 },
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    { 2, 1, ACB_TYPE_NONE,       FCB_TYPE_HARDCODED,  0,  28 },
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    { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES,  0,  46 },
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    { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2,  80 },
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    { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
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    { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
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    { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
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    { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
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    { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0,  64 },
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    { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2,  80 },
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    { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 104 },
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    { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 108 },
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    { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 132 },
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    { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 168 },
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    { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 176 },
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    { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 208 },
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    { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 256 }
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};
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/**
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 * WMA Voice decoding context.
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 */
128
typedef struct {
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    /**
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     * @defgroup struct_global Global values
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     * Global values, specified in the stream header / extradata or used
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     * all over.
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     * @{
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     */
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    GetBitContext gb;             ///< packet bitreader. During decoder init,
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                                  ///< it contains the extradata from the
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                                  ///< demuxer. During decoding, it contains
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                                  ///< packet data.
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    int8_t vbm_tree[25];          ///< converts VLC codes to frame type
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    int spillover_bitsize;        ///< number of bits used to specify
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                                  ///< #spillover_nbits in the packet header
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                                  ///< = ceil(log2(ctx->block_align << 3))
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    int history_nsamples;         ///< number of samples in history for signal
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                                  ///< prediction (through ACB)
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    /* postfilter specific values */
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    int do_apf;                   ///< whether to apply the averaged
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                                  ///< projection filter (APF)
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    int denoise_strength;         ///< strength of denoising in Wiener filter
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                                  ///< [0-11]
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    int denoise_tilt_corr;        ///< Whether to apply tilt correction to the
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                                  ///< Wiener filter coefficients (postfilter)
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    int dc_level;                 ///< Predicted amount of DC noise, based
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                                  ///< on which a DC removal filter is used
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    int lsps;                     ///< number of LSPs per frame [10 or 16]
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    int lsp_q_mode;               ///< defines quantizer defaults [0, 1]
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    int lsp_def_mode;             ///< defines different sets of LSP defaults
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                                  ///< [0, 1]
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    int frame_lsp_bitsize;        ///< size (in bits) of LSPs, when encoded
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                                  ///< per-frame (independent coding)
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    int sframe_lsp_bitsize;       ///< size (in bits) of LSPs, when encoded
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                                  ///< per superframe (residual coding)
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    int min_pitch_val;            ///< base value for pitch parsing code
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    int max_pitch_val;            ///< max value + 1 for pitch parsing
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    int pitch_nbits;              ///< number of bits used to specify the
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                                  ///< pitch value in the frame header
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    int block_pitch_nbits;        ///< number of bits used to specify the
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                                  ///< first block's pitch value
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    int block_pitch_range;        ///< range of the block pitch
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    int block_delta_pitch_nbits;  ///< number of bits used to specify the
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                                  ///< delta pitch between this and the last
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                                  ///< block's pitch value, used in all but
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                                  ///< first block
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    int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
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                                  ///< from -this to +this-1)
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    uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
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                                  ///< conversion
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    /**
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     * @}
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     * @defgroup struct_packet Packet values
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     * Packet values, specified in the packet header or related to a packet.
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     * A packet is considered to be a single unit of data provided to this
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     * decoder by the demuxer.
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     * @{
189
     */
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    int spillover_nbits;          ///< number of bits of the previous packet's
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                                  ///< last superframe preceeding this
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                                  ///< packet's first full superframe (useful
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                                  ///< for re-synchronization also)
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    int has_residual_lsps;        ///< if set, superframes contain one set of
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                                  ///< LSPs that cover all frames, encoded as
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                                  ///< independent and residual LSPs; if not
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                                  ///< set, each frame contains its own, fully
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                                  ///< independent, LSPs
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    int skip_bits_next;           ///< number of bits to skip at the next call
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                                  ///< to #wmavoice_decode_packet() (since
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                                  ///< they're part of the previous superframe)
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    uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
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                                  ///< cache for superframe data split over
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                                  ///< multiple packets
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    int sframe_cache_size;        ///< set to >0 if we have data from an
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                                  ///< (incomplete) superframe from a previous
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                                  ///< packet that spilled over in the current
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                                  ///< packet; specifies the amount of bits in
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                                  ///< #sframe_cache
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    PutBitContext pb;             ///< bitstream writer for #sframe_cache
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213
    /**
214
     * @}
215
     * @defgroup struct_frame Frame and superframe values
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     * Superframe and frame data - these can change from frame to frame,
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     * although some of them do in that case serve as a cache / history for
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     * the next frame or superframe.
219
     * @{
220
     */
221
    double prev_lsps[MAX_LSPS];   ///< LSPs of the last frame of the previous
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                                  ///< superframe
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    int last_pitch_val;           ///< pitch value of the previous frame
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    int last_acb_type;            ///< frame type [0-2] of the previous frame
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    int pitch_diff_sh16;          ///< ((cur_pitch_val - #last_pitch_val)
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                                  ///< << 16) / #MAX_FRAMESIZE
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    float silence_gain;           ///< set for use in blocks if #ACB_TYPE_NONE
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    int aw_idx_is_ext;            ///< whether the AW index was encoded in
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                                  ///< 8 bits (instead of 6)
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    int aw_pulse_range;           ///< the range over which #aw_pulse_set1()
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                                  ///< can apply the pulse, relative to the
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                                  ///< value in aw_first_pulse_off. The exact
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                                  ///< position of the first AW-pulse is within
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                                  ///< [pulse_off, pulse_off + this], and
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                                  ///< depends on bitstream values; [16 or 24]
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    int aw_n_pulses[2];           ///< number of AW-pulses in each block; note
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                                  ///< that this number can be negative (in
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                                  ///< which case it basically means "zero")
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    int aw_first_pulse_off[2];    ///< index of first sample to which to
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                                  ///< apply AW-pulses, or -0xff if unset
242
    int aw_next_pulse_off_cache;  ///< the position (relative to start of the
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                                  ///< second block) at which pulses should
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                                  ///< start to be positioned, serves as a
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                                  ///< cache for pitch-adaptive window pulses
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                                  ///< between blocks
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248
    int frame_cntr;               ///< current frame index [0 - 0xFFFE]; is
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                                  ///< only used for comfort noise in #pRNG()
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    float gain_pred_err[6];       ///< cache for gain prediction
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    float excitation_history[MAX_SIGNAL_HISTORY];
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                                  ///< cache of the signal of previous
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                                  ///< superframes, used as a history for
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                                  ///< signal generation
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    float synth_history[MAX_LSPS]; ///< see #excitation_history
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    /**
257
     * @}
258
     * @defgroup post_filter Postfilter values
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     * Varibales used for postfilter implementation, mostly history for
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     * smoothing and so on, and context variables for FFT/iFFT.
261
     * @{
262
     */
263
    RDFTContext rdft, irdft;      ///< contexts for FFT-calculation in the
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                                  ///< postfilter (for denoise filter)
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    DCTContext dct, dst;          ///< contexts for phase shift (in Hilbert
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                                  ///< transform, part of postfilter)
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    float sin[511], cos[511];     ///< 8-bit cosine/sine windows over [-pi,pi]
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                                  ///< range
269
    float postfilter_agc;         ///< gain control memory, used in
270
                                  ///< #adaptive_gain_control()
271
    float dcf_mem[2];             ///< DC filter history
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    float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
273
                                  ///< zero filter output (i.e. excitation)
274
                                  ///< by postfilter
275
    float denoise_filter_cache[MAX_FRAMESIZE];
276
    int   denoise_filter_cache_size; ///< samples in #denoise_filter_cache
277
    DECLARE_ALIGNED(16, float, tilted_lpcs_pf)[0x80];
278
                                  ///< aligned buffer for LPC tilting
279
    DECLARE_ALIGNED(16, float, denoise_coeffs_pf)[0x80];
280
                                  ///< aligned buffer for denoise coefficients
281
    DECLARE_ALIGNED(16, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
282
                                  ///< aligned buffer for postfilter speech
283
                                  ///< synthesis
284
    /**
285
     * @}
286
     */
287
} WMAVoiceContext;
288

    
289
/**
290
 * Sets up the variable bit mode (VBM) tree from container extradata.
291
 * @param gb bit I/O context.
292
 *           The bit context (s->gb) should be loaded with byte 23-46 of the
293
 *           container extradata (i.e. the ones containing the VBM tree).
294
 * @param vbm_tree pointer to array to which the decoded VBM tree will be
295
 *                 written.
296
 * @return 0 on success, <0 on error.
297
 */
298
static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
299
{
300
    static const uint8_t bits[] = {
301
         2,  2,  2,  4,  4,  4,
302
         6,  6,  6,  8,  8,  8,
303
        10, 10, 10, 12, 12, 12,
304
        14, 14, 14, 14
305
    };
306
    static const uint16_t codes[] = {
307
          0x0000, 0x0001, 0x0002,        //              00/01/10
308
          0x000c, 0x000d, 0x000e,        //           11+00/01/10
309
          0x003c, 0x003d, 0x003e,        //         1111+00/01/10
310
          0x00fc, 0x00fd, 0x00fe,        //       111111+00/01/10
311
          0x03fc, 0x03fd, 0x03fe,        //     11111111+00/01/10
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          0x0ffc, 0x0ffd, 0x0ffe,        //   1111111111+00/01/10
313
          0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
314
    };
315
    int cntr[8], n, res;
316

    
317
    memset(vbm_tree, 0xff, sizeof(vbm_tree));
318
    memset(cntr,     0,    sizeof(cntr));
319
    for (n = 0; n < 17; n++) {
320
        res = get_bits(gb, 3);
321
        if (cntr[res] > 3) // should be >= 3 + (res == 7))
322
            return -1;
323
        vbm_tree[res * 3 + cntr[res]++] = n;
324
    }
325
    INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
326
                    bits, 1, 1, codes, 2, 2, 132);
327
    return 0;
328
}
329

    
330
/**
331
 * Set up decoder with parameters from demuxer (extradata etc.).
332
 */
333
static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
334
{
335
    int n, flags, pitch_range, lsp16_flag;
336
    WMAVoiceContext *s = ctx->priv_data;
337

    
338
    /**
339
     * Extradata layout:
340
     * - byte  0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
341
     * - byte 19-22: flags field (annoyingly in LE; see below for known
342
     *               values),
343
     * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
344
     *               rest is 0).
345
     */
346
    if (ctx->extradata_size != 46) {
347
        av_log(ctx, AV_LOG_ERROR,
348
               "Invalid extradata size %d (should be 46)\n",
349
               ctx->extradata_size);
350
        return -1;
351
    }
352
    flags                = AV_RL32(ctx->extradata + 18);
353
    s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
354
    s->do_apf            =    flags & 0x1;
355
    if (s->do_apf) {
356
        ff_rdft_init(&s->rdft,  7, DFT_R2C);
357
        ff_rdft_init(&s->irdft, 7, IDFT_C2R);
358
        ff_dct_init(&s->dct,  6, DCT_I);
359
        ff_dct_init(&s->dst,  6, DST_I);
360

    
361
        ff_sine_window_init(s->cos, 256);
362
        memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
363
        for (n = 0; n < 255; n++) {
364
            s->sin[n]       = -s->sin[510 - n];
365
            s->cos[510 - n] =  s->cos[n];
366
        }
367
    }
368
    s->denoise_strength  =   (flags >> 2) & 0xF;
369
    if (s->denoise_strength >= 12) {
370
        av_log(ctx, AV_LOG_ERROR,
371
               "Invalid denoise filter strength %d (max=11)\n",
372
               s->denoise_strength);
373
        return -1;
374
    }
375
    s->denoise_tilt_corr = !!(flags & 0x40);
376
    s->dc_level          =   (flags >> 7) & 0xF;
377
    s->lsp_q_mode        = !!(flags & 0x2000);
378
    s->lsp_def_mode      = !!(flags & 0x4000);
379
    lsp16_flag           =    flags & 0x1000;
380
    if (lsp16_flag) {
381
        s->lsps               = 16;
382
        s->frame_lsp_bitsize  = 34;
383
        s->sframe_lsp_bitsize = 60;
384
    } else {
385
        s->lsps               = 10;
386
        s->frame_lsp_bitsize  = 24;
387
        s->sframe_lsp_bitsize = 48;
388
    }
389
    for (n = 0; n < s->lsps; n++)
390
        s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
391

    
392
    init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
393
    if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
394
        av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
395
        return -1;
396
    }
397

    
398
    s->min_pitch_val    = ((ctx->sample_rate << 8)      /  400 + 50) >> 8;
399
    s->max_pitch_val    = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
400
    pitch_range         = s->max_pitch_val - s->min_pitch_val;
401
    s->pitch_nbits      = av_ceil_log2(pitch_range);
402
    s->last_pitch_val   = 40;
403
    s->last_acb_type    = ACB_TYPE_NONE;
404
    s->history_nsamples = s->max_pitch_val + 8;
405

    
406
    if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
407
        int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
408
            max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
409

    
410
        av_log(ctx, AV_LOG_ERROR,
411
               "Unsupported samplerate %d (min=%d, max=%d)\n",
412
               ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
413

    
414
        return -1;
415
    }
416

    
417
    s->block_conv_table[0]      = s->min_pitch_val;
418
    s->block_conv_table[1]      = (pitch_range * 25) >> 6;
419
    s->block_conv_table[2]      = (pitch_range * 44) >> 6;
420
    s->block_conv_table[3]      = s->max_pitch_val - 1;
421
    s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
422
    s->block_delta_pitch_nbits  = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
423
    s->block_pitch_range        = s->block_conv_table[2] +
424
                                  s->block_conv_table[3] + 1 +
425
                                  2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
426
    s->block_pitch_nbits        = av_ceil_log2(s->block_pitch_range);
427

    
428
    ctx->sample_fmt             = SAMPLE_FMT_FLT;
429

    
430
    return 0;
431
}
432

    
433
/**
434
 * @defgroup postfilter Postfilter functions
435
 * Postfilter functions (gain control, wiener denoise filter, DC filter,
436
 * kalman smoothening, plus surrounding code to wrap it)
437
 * @{
438
 */
439
/**
440
 * Adaptive gain control (as used in postfilter).
441
 *
442
 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
443
 * that the energy here is calculated using sum(abs(...)), whereas the
444
 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
445
 *
446
 * @param out output buffer for filtered samples
447
 * @param in input buffer containing the samples as they are after the
448
 *           postfilter steps so far
449
 * @param speech_synth input buffer containing speech synth before postfilter
450
 * @param size input buffer size
451
 * @param alpha exponential filter factor
452
 * @param gain_mem pointer to filter memory (single float)
453
 */
454
static void adaptive_gain_control(float *out, const float *in,
455
                                  const float *speech_synth,
456
                                  int size, float alpha, float *gain_mem)
457
{
458
    int i;
459
    float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
460
    float mem = *gain_mem;
461

    
462
    for (i = 0; i < size; i++) {
463
        speech_energy     += fabsf(speech_synth[i]);
464
        postfilter_energy += fabsf(in[i]);
465
    }
466
    gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
467

    
468
    for (i = 0; i < size; i++) {
469
        mem = alpha * mem + gain_scale_factor;
470
        out[i] = in[i] * mem;
471
    }
472

    
473
    *gain_mem = mem;
474
}
475

    
476
/**
477
 * Kalman smoothing function.
478
 *
479
 * This function looks back pitch +/- 3 samples back into history to find
480
 * the best fitting curve (that one giving the optimal gain of the two
481
 * signals, i.e. the highest dot product between the two), and then
482
 * uses that signal history to smoothen the output of the speech synthesis
483
 * filter.
484
 *
485
 * @param s WMA Voice decoding context
486
 * @param pitch pitch of the speech signal
487
 * @param in input speech signal
488
 * @param out output pointer for smoothened signal
489
 * @param size input/output buffer size
490
 *
491
 * @returns -1 if no smoothening took place, e.g. because no optimal
492
 *          fit could be found, or 0 on success.
493
 */
494
static int kalman_smoothen(WMAVoiceContext *s, int pitch,
495
                           const float *in, float *out, int size)
496
{
497
    int n;
498
    float optimal_gain = 0, dot;
499
    const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
500
                *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
501
                *best_hist_ptr;
502

    
503
    /* find best fitting point in history */
504
    do {
505
        dot = ff_dot_productf(in, ptr, size);
506
        if (dot > optimal_gain) {
507
            optimal_gain  = dot;
508
            best_hist_ptr = ptr;
509
        }
510
    } while (--ptr >= end);
511

    
512
    if (optimal_gain <= 0)
513
        return -1;
514
    dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size);
515
    if (dot <= 0) // would be 1.0
516
        return -1;
517

    
518
    if (optimal_gain <= dot) {
519
        dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
520
    } else
521
        dot = 0.625;
522

    
523
    /* actual smoothing */
524
    for (n = 0; n < size; n++)
525
        out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
526

    
527
    return 0;
528
}
529

    
530
/**
531
 * Get the tilt factor of a formant filter from its transfer function
532
 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
533
 *      but somehow (??) it does a speech synthesis filter in the
534
 *      middle, which is missing here
535
 *
536
 * @param lpcs LPC coefficients
537
 * @param n_lpcs Size of LPC buffer
538
 * @returns the tilt factor
539
 */
540
static float tilt_factor(const float *lpcs, int n_lpcs)
541
{
542
    float rh0, rh1;
543

    
544
    rh0 = 1.0     + ff_dot_productf(lpcs,  lpcs,    n_lpcs);
545
    rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1);
546

    
547
    return rh1 / rh0;
548
}
549

    
550
/**
551
 * Derive denoise filter coefficients (in real domain) from the LPCs.
552
 */
553
static void calc_input_response(WMAVoiceContext *s, float *lpcs,
554
                                int fcb_type, float *coeffs, int remainder)
555
{
556
    float last_coeff, min = 15.0, max = -15.0;
557
    float irange, angle_mul, gain_mul, range, sq;
558
    int n, idx;
559

    
560
    /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
561
    ff_rdft_calc(&s->rdft, lpcs);
562
#define log_range(var, assign) do { \
563
        float tmp = log10f(assign);  var = tmp; \
564
        max       = FFMAX(max, tmp); min = FFMIN(min, tmp); \
565
    } while (0)
566
    log_range(last_coeff,  lpcs[1]         * lpcs[1]);
567
    for (n = 1; n < 64; n++)
568
        log_range(lpcs[n], lpcs[n * 2]     * lpcs[n * 2] +
569
                           lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
570
    log_range(lpcs[0],     lpcs[0]         * lpcs[0]);
571
#undef log_range
572
    range    = max - min;
573
    lpcs[64] = last_coeff;
574

    
575
    /* Now, use this spectrum to pick out these frequencies with higher
576
     * (relative) power/energy (which we then take to be "not noise"),
577
     * and set up a table (still in lpc[]) of (relative) gains per frequency.
578
     * These frequencies will be maintained, while others ("noise") will be
579
     * decreased in the filter output. */
580
    irange    = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
581
    gain_mul  = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
582
                                                          (5.0 / 14.7));
583
    angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
584
    for (n = 0; n <= 64; n++) {
585
        float pow;
586

    
587
        idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
588
        pow = wmavoice_denoise_power_table[s->denoise_strength][idx];
589
        lpcs[n] = angle_mul * pow;
590

    
591
        /* 70.57 =~ 1/log10(1.0331663) */
592
        idx = (pow * gain_mul - 0.0295) * 70.570526123;
593
        if (idx > 127) { // fallback if index falls outside table range
594
            coeffs[n] = wmavoice_energy_table[127] *
595
                        powf(1.0331663, idx - 127);
596
        } else
597
            coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
598
    }
599

    
600
    /* calculate the Hilbert transform of the gains, which we do (since this
601
     * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
602
     * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
603
     * "moment" of the LPCs in this filter. */
604
    ff_dct_calc(&s->dct, lpcs);
605
    ff_dct_calc(&s->dst, lpcs);
606

    
607
    /* Split out the coefficient indexes into phase/magnitude pairs */
608
    idx = 255 + av_clip(lpcs[64],               -255, 255);
609
    coeffs[0]  = coeffs[0]  * s->cos[idx];
610
    idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
611
    last_coeff = coeffs[64] * s->cos[idx];
612
    for (n = 63;; n--) {
613
        idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
614
        coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
615
        coeffs[n * 2]     = coeffs[n] * s->cos[idx];
616

    
617
        if (!--n) break;
618

    
619
        idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
620
        coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
621
        coeffs[n * 2]     = coeffs[n] * s->cos[idx];
622
    }
623
    coeffs[1] = last_coeff;
624

    
625
    /* move into real domain */
626
    ff_rdft_calc(&s->irdft, coeffs);
627

    
628
    /* tilt correction and normalize scale */
629
    memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
630
    if (s->denoise_tilt_corr) {
631
        float tilt_mem = 0;
632

    
633
        coeffs[remainder - 1] = 0;
634
        ff_tilt_compensation(&tilt_mem,
635
                             -1.8 * tilt_factor(coeffs, remainder - 1),
636
                             coeffs, remainder);
637
    }
638
    sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder));
639
    for (n = 0; n < remainder; n++)
640
        coeffs[n] *= sq;
641
}
642

    
643
/**
644
 * This function applies a Wiener filter on the (noisy) speech signal as
645
 * a means to denoise it.
646
 *
647
 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
648
 * - using this power spectrum, calculate (for each frequency) the Wiener
649
 *    filter gain, which depends on the frequency power and desired level
650
 *    of noise subtraction (when set too high, this leads to artifacts)
651
 *    We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
652
 *    of 4-8kHz);
653
 * - by doing a phase shift, calculate the Hilbert transform of this array
654
 *    of per-frequency filter-gains to get the filtering coefficients;
655
 * - smoothen/normalize/de-tilt these filter coefficients as desired;
656
 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
657
 *    to get the denoised speech signal;
658
 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
659
 *    the frame boundary) are saved and applied to subsequent frames by an
660
 *    overlap-add method (otherwise you get clicking-artifacts).
661
 *
662
 * @param s WMA Voice decoding context
663
 * @param s fcb_type Frame (codebook) type
664
 * @param synth_pf input: the noisy speech signal, output: denoised speech
665
 *                 data; should be 16-byte aligned (for ASM purposes)
666
 * @param size size of the speech data
667
 * @param lpcs LPCs used to synthesize this frame's speech data
668
 */
669
static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
670
                           float *synth_pf, int size,
671
                           const float *lpcs)
672
{
673
    int remainder, lim, n;
674

    
675
    if (fcb_type != FCB_TYPE_SILENCE) {
676
        float *tilted_lpcs = s->tilted_lpcs_pf,
677
              *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
678

    
679
        tilted_lpcs[0]           = 1.0;
680
        memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
681
        memset(&tilted_lpcs[s->lsps + 1], 0,
682
               sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
683
        ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
684
                             tilted_lpcs, s->lsps + 2);
685

    
686
        /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
687
         * size is applied to the next frame. All input beyond this is zero,
688
         * and thus all output beyond this will go towards zero, hence we can
689
         * limit to min(size-1, 127-size) as a performance consideration. */
690
        remainder = FFMIN(127 - size, size - 1);
691
        calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
692

    
693
        /* apply coefficients (in frequency spectrum domain), i.e. complex
694
         * number multiplication */
695
        memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
696
        ff_rdft_calc(&s->rdft, synth_pf);
697
        ff_rdft_calc(&s->rdft, coeffs);
698
        synth_pf[0] *= coeffs[0];
699
        synth_pf[1] *= coeffs[1];
700
        for (n = 1; n < 64; n++) {
701
            float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
702
            synth_pf[n * 2]     = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
703
            synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
704
        }
705
        ff_rdft_calc(&s->irdft, synth_pf);
706
    }
707

    
708
    /* merge filter output with the history of previous runs */
709
    if (s->denoise_filter_cache_size) {
710
        lim = FFMIN(s->denoise_filter_cache_size, size);
711
        for (n = 0; n < lim; n++)
712
            synth_pf[n] += s->denoise_filter_cache[n];
713
        s->denoise_filter_cache_size -= lim;
714
        memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
715
                sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
716
    }
717

    
718
    /* move remainder of filter output into a cache for future runs */
719
    if (fcb_type != FCB_TYPE_SILENCE) {
720
        lim = FFMIN(remainder, s->denoise_filter_cache_size);
721
        for (n = 0; n < lim; n++)
722
            s->denoise_filter_cache[n] += synth_pf[size + n];
723
        if (lim < remainder) {
724
            memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
725
                   sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
726
            s->denoise_filter_cache_size = remainder;
727
        }
728
    }
729
}
730

    
731
/**
732
 * Averaging projection filter, the postfilter used in WMAVoice.
733
 *
734
 * This uses the following steps:
735
 * - A zero-synthesis filter (generate excitation from synth signal)
736
 * - Kalman smoothing on excitation, based on pitch
737
 * - Re-synthesized smoothened output
738
 * - Iterative Wiener denoise filter
739
 * - Adaptive gain filter
740
 * - DC filter
741
 *
742
 * @param s WMAVoice decoding context
743
 * @param synth Speech synthesis output (before postfilter)
744
 * @param samples Output buffer for filtered samples
745
 * @param size Buffer size of synth & samples
746
 * @param lpcs Generated LPCs used for speech synthesis
747
 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
748
 * @param pitch Pitch of the input signal
749
 */
750
static void postfilter(WMAVoiceContext *s, const float *synth,
751
                       float *samples,    int size,
752
                       const float *lpcs, float *zero_exc_pf,
753
                       int fcb_type,      int pitch)
754
{
755
    float synth_filter_in_buf[MAX_FRAMESIZE / 2],
756
          *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
757
          *synth_filter_in = zero_exc_pf;
758

    
759
    assert(size <= MAX_FRAMESIZE / 2);
760

    
761
    /* generate excitation from input signal */
762
    ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
763

    
764
    if (fcb_type >= FCB_TYPE_AW_PULSES &&
765
        !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
766
        synth_filter_in = synth_filter_in_buf;
767

    
768
    /* re-synthesize speech after smoothening, and keep history */
769
    ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
770
                                 synth_filter_in, size, s->lsps);
771
    memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
772
           sizeof(synth_pf[0]) * s->lsps);
773

    
774
    wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
775

    
776
    adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
777
                          &s->postfilter_agc);
778

    
779
    if (s->dc_level > 8) {
780
        /* remove ultra-low frequency DC noise / highpass filter;
781
         * coefficients are identical to those used in SIPR decoding,
782
         * and very closely resemble those used in AMR-NB decoding. */
783
        ff_acelp_apply_order_2_transfer_function(samples, samples,
784
            (const float[2]) { -1.99997,      1.0 },
785
            (const float[2]) { -1.9330735188, 0.93589198496 },
786
            0.93980580475, s->dcf_mem, size);
787
    }
788
}
789
/**
790
 * @}
791
 */
792

    
793
/**
794
 * Dequantize LSPs
795
 * @param lsps output pointer to the array that will hold the LSPs
796
 * @param num number of LSPs to be dequantized
797
 * @param values quantized values, contains n_stages values
798
 * @param sizes range (i.e. max value) of each quantized value
799
 * @param n_stages number of dequantization runs
800
 * @param table dequantization table to be used
801
 * @param mul_q LSF multiplier
802
 * @param base_q base (lowest) LSF values
803
 */
804
static void dequant_lsps(double *lsps, int num,
805
                         const uint16_t *values,
806
                         const uint16_t *sizes,
807
                         int n_stages, const uint8_t *table,
808
                         const double *mul_q,
809
                         const double *base_q)
810
{
811
    int n, m;
812

    
813
    memset(lsps, 0, num * sizeof(*lsps));
814
    for (n = 0; n < n_stages; n++) {
815
        const uint8_t *t_off = &table[values[n] * num];
816
        double base = base_q[n], mul = mul_q[n];
817

    
818
        for (m = 0; m < num; m++)
819
            lsps[m] += base + mul * t_off[m];
820

    
821
        table += sizes[n] * num;
822
    }
823
}
824

    
825
/**
826
 * @defgroup lsp_dequant LSP dequantization routines
827
 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
828
 * @note we assume enough bits are available, caller should check.
829
 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
830
 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
831
 * @{
832
 */
833
/**
834
 * Parse 10 independently-coded LSPs.
835
 */
836
static void dequant_lsp10i(GetBitContext *gb, double *lsps)
837
{
838
    static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
839
    static const double mul_lsf[4] = {
840
        5.2187144800e-3,    1.4626986422e-3,
841
        9.6179549166e-4,    1.1325736225e-3
842
    };
843
    static const double base_lsf[4] = {
844
        M_PI * -2.15522e-1, M_PI * -6.1646e-2,
845
        M_PI * -3.3486e-2,  M_PI * -5.7408e-2
846
    };
847
    uint16_t v[4];
848

    
849
    v[0] = get_bits(gb, 8);
850
    v[1] = get_bits(gb, 6);
851
    v[2] = get_bits(gb, 5);
852
    v[3] = get_bits(gb, 5);
853

    
854
    dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
855
                 mul_lsf, base_lsf);
856
}
857

    
858
/**
859
 * Parse 10 independently-coded LSPs, and then derive the tables to
860
 * generate LSPs for the other frames from them (residual coding).
861
 */
862
static void dequant_lsp10r(GetBitContext *gb,
863
                           double *i_lsps, const double *old,
864
                           double *a1, double *a2, int q_mode)
865
{
866
    static const uint16_t vec_sizes[3] = { 128, 64, 64 };
867
    static const double mul_lsf[3] = {
868
        2.5807601174e-3,    1.2354460219e-3,   1.1763821673e-3
869
    };
870
    static const double base_lsf[3] = {
871
        M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
872
    };
873
    const float (*ipol_tab)[2][10] = q_mode ?
874
        wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
875
    uint16_t interpol, v[3];
876
    int n;
877

    
878
    dequant_lsp10i(gb, i_lsps);
879

    
880
    interpol = get_bits(gb, 5);
881
    v[0]     = get_bits(gb, 7);
882
    v[1]     = get_bits(gb, 6);
883
    v[2]     = get_bits(gb, 6);
884

    
885
    for (n = 0; n < 10; n++) {
886
        double delta = old[n] - i_lsps[n];
887
        a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
888
        a1[10 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
889
    }
890

    
891
    dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
892
                 mul_lsf, base_lsf);
893
}
894

    
895
/**
896
 * Parse 16 independently-coded LSPs.
897
 */
898
static void dequant_lsp16i(GetBitContext *gb, double *lsps)
899
{
900
    static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
901
    static const double mul_lsf[5] = {
902
        3.3439586280e-3,    6.9908173703e-4,
903
        3.3216608306e-3,    1.0334960326e-3,
904
        3.1899104283e-3
905
    };
906
    static const double base_lsf[5] = {
907
        M_PI * -1.27576e-1, M_PI * -2.4292e-2,
908
        M_PI * -1.28094e-1, M_PI * -3.2128e-2,
909
        M_PI * -1.29816e-1
910
    };
911
    uint16_t v[5];
912

    
913
    v[0] = get_bits(gb, 8);
914
    v[1] = get_bits(gb, 6);
915
    v[2] = get_bits(gb, 7);
916
    v[3] = get_bits(gb, 6);
917
    v[4] = get_bits(gb, 7);
918

    
919
    dequant_lsps( lsps,     5,  v,     vec_sizes,    2,
920
                 wmavoice_dq_lsp16i1,  mul_lsf,     base_lsf);
921
    dequant_lsps(&lsps[5],  5, &v[2], &vec_sizes[2], 2,
922
                 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
923
    dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
924
                 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
925
}
926

    
927
/**
928
 * Parse 16 independently-coded LSPs, and then derive the tables to
929
 * generate LSPs for the other frames from them (residual coding).
930
 */
931
static void dequant_lsp16r(GetBitContext *gb,
932
                           double *i_lsps, const double *old,
933
                           double *a1, double *a2, int q_mode)
934
{
935
    static const uint16_t vec_sizes[3] = { 128, 128, 128 };
936
    static const double mul_lsf[3] = {
937
        1.2232979501e-3,   1.4062241527e-3,   1.6114744851e-3
938
    };
939
    static const double base_lsf[3] = {
940
        M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
941
    };
942
    const float (*ipol_tab)[2][16] = q_mode ?
943
        wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
944
    uint16_t interpol, v[3];
945
    int n;
946

    
947
    dequant_lsp16i(gb, i_lsps);
948

    
949
    interpol = get_bits(gb, 5);
950
    v[0]     = get_bits(gb, 7);
951
    v[1]     = get_bits(gb, 7);
952
    v[2]     = get_bits(gb, 7);
953

    
954
    for (n = 0; n < 16; n++) {
955
        double delta = old[n] - i_lsps[n];
956
        a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
957
        a1[16 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
958
    }
959

    
960
    dequant_lsps( a2,     10,  v,     vec_sizes,    1,
961
                 wmavoice_dq_lsp16r1,  mul_lsf,     base_lsf);
962
    dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
963
                 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
964
    dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
965
                 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
966
}
967

    
968
/**
969
 * @}
970
 * @defgroup aw Pitch-adaptive window coding functions
971
 * The next few functions are for pitch-adaptive window coding.
972
 * @{
973
 */
974
/**
975
 * Parse the offset of the first pitch-adaptive window pulses, and
976
 * the distribution of pulses between the two blocks in this frame.
977
 * @param s WMA Voice decoding context private data
978
 * @param gb bit I/O context
979
 * @param pitch pitch for each block in this frame
980
 */
981
static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
982
                            const int *pitch)
983
{
984
    static const int16_t start_offset[94] = {
985
        -11,  -9,  -7,  -5,  -3,  -1,   1,   3,   5,   7,   9,  11,
986
         13,  15,  18,  17,  19,  20,  21,  22,  23,  24,  25,  26,
987
         27,  28,  29,  30,  31,  32,  33,  35,  37,  39,  41,  43,
988
         45,  47,  49,  51,  53,  55,  57,  59,  61,  63,  65,  67,
989
         69,  71,  73,  75,  77,  79,  81,  83,  85,  87,  89,  91,
990
         93,  95,  97,  99, 101, 103, 105, 107, 109, 111, 113, 115,
991
        117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
992
        141, 143, 145, 147, 149, 151, 153, 155, 157, 159
993
    };
994
    int bits, offset;
995

    
996
    /* position of pulse */
997
    s->aw_idx_is_ext = 0;
998
    if ((bits = get_bits(gb, 6)) >= 54) {
999
        s->aw_idx_is_ext = 1;
1000
        bits += (bits - 54) * 3 + get_bits(gb, 2);
1001
    }
1002

    
1003
    /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1004
     * the distribution of the pulses in each block contained in this frame. */
1005
    s->aw_pulse_range        = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1006
    for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1007
    s->aw_n_pulses[0]        = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1008
    s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1009
    offset                  += s->aw_n_pulses[0] * pitch[0];
1010
    s->aw_n_pulses[1]        = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1011
    s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1012

    
1013
    /* if continuing from a position before the block, reset position to
1014
     * start of block (when corrected for the range over which it can be
1015
     * spread in aw_pulse_set1()). */
1016
    if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1017
        while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1018
            s->aw_first_pulse_off[1] -= pitch[1];
1019
        if (start_offset[bits] < 0)
1020
            while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1021
                s->aw_first_pulse_off[0] -= pitch[0];
1022
    }
1023
}
1024

    
1025
/**
1026
 * Apply second set of pitch-adaptive window pulses.
1027
 * @param s WMA Voice decoding context private data
1028
 * @param gb bit I/O context
1029
 * @param block_idx block index in frame [0, 1]
1030
 * @param fcb structure containing fixed codebook vector info
1031
 */
1032
static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1033
                          int block_idx, AMRFixed *fcb)
1034
{
1035
    uint16_t use_mask[7]; // only 5 are used, rest is padding
1036
    /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1037
     * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1038
     * of idx are the position of the bit within a particular item in the
1039
     * array (0 being the most significant bit, and 15 being the least
1040
     * significant bit), and the remainder (>> 4) is the index in the
1041
     * use_mask[]-array. This is faster and uses less memory than using a
1042
     * 80-byte/80-int array. */
1043
    int pulse_off = s->aw_first_pulse_off[block_idx],
1044
        pulse_start, n, idx, range, aidx, start_off = 0;
1045

    
1046
    /* set offset of first pulse to within this block */
1047
    if (s->aw_n_pulses[block_idx] > 0)
1048
        while (pulse_off + s->aw_pulse_range < 1)
1049
            pulse_off += fcb->pitch_lag;
1050

    
1051
    /* find range per pulse */
1052
    if (s->aw_n_pulses[0] > 0) {
1053
        if (block_idx == 0) {
1054
            range = 32;
1055
        } else /* block_idx = 1 */ {
1056
            range = 8;
1057
            if (s->aw_n_pulses[block_idx] > 0)
1058
                pulse_off = s->aw_next_pulse_off_cache;
1059
        }
1060
    } else
1061
        range = 16;
1062
    pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1063

    
1064
    /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1065
     * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1066
     * we exclude that range from being pulsed again in this function. */
1067
    memset( use_mask,   -1, 5 * sizeof(use_mask[0]));
1068
    memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1069
    if (s->aw_n_pulses[block_idx] > 0)
1070
        for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1071
            int excl_range         = s->aw_pulse_range; // always 16 or 24
1072
            uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1073
            int first_sh           = 16 - (idx & 15);
1074
            *use_mask_ptr++       &= 0xFFFF << first_sh;
1075
            excl_range            -= first_sh;
1076
            if (excl_range >= 16) {
1077
                *use_mask_ptr++    = 0;
1078
                *use_mask_ptr     &= 0xFFFF >> (excl_range - 16);
1079
            } else
1080
                *use_mask_ptr     &= 0xFFFF >> excl_range;
1081
        }
1082

    
1083
    /* find the 'aidx'th offset that is not excluded */
1084
    aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1085
    for (n = 0; n <= aidx; pulse_start++) {
1086
        for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1087
        if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1088
            if (use_mask[0])      idx = 0x0F;
1089
            else if (use_mask[1]) idx = 0x1F;
1090
            else if (use_mask[2]) idx = 0x2F;
1091
            else if (use_mask[3]) idx = 0x3F;
1092
            else if (use_mask[4]) idx = 0x4F;
1093
            else                  return;
1094
            idx -= av_log2_16bit(use_mask[idx >> 4]);
1095
        }
1096
        if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1097
            use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1098
            n++;
1099
            start_off = idx;
1100
        }
1101
    }
1102

    
1103
    fcb->x[fcb->n] = start_off;
1104
    fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1105
    fcb->n++;
1106

    
1107
    /* set offset for next block, relative to start of that block */
1108
    n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1109
    s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1110
}
1111

    
1112
/**
1113
 * Apply first set of pitch-adaptive window pulses.
1114
 * @param s WMA Voice decoding context private data
1115
 * @param gb bit I/O context
1116
 * @param block_idx block index in frame [0, 1]
1117
 * @param fcb storage location for fixed codebook pulse info
1118
 */
1119
static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1120
                          int block_idx, AMRFixed *fcb)
1121
{
1122
    int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1123
    float v;
1124

    
1125
    if (s->aw_n_pulses[block_idx] > 0) {
1126
        int n, v_mask, i_mask, sh, n_pulses;
1127

    
1128
        if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1129
            n_pulses = 3;
1130
            v_mask   = 8;
1131
            i_mask   = 7;
1132
            sh       = 4;
1133
        } else { // 4 pulses, 1:sign + 2:index each
1134
            n_pulses = 4;
1135
            v_mask   = 4;
1136
            i_mask   = 3;
1137
            sh       = 3;
1138
        }
1139

    
1140
        for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1141
            fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1142
            fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1143
                                 s->aw_first_pulse_off[block_idx];
1144
            while (fcb->x[fcb->n] < 0)
1145
                fcb->x[fcb->n] += fcb->pitch_lag;
1146
            if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1147
                fcb->n++;
1148
        }
1149
    } else {
1150
        int num2 = (val & 0x1FF) >> 1, delta, idx;
1151

    
1152
        if (num2 < 1 * 79)      { delta = 1; idx = num2 + 1; }
1153
        else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1154
        else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1155
        else                    { delta = 7; idx = num2 + 1 - 3 * 75; }
1156
        v = (val & 0x200) ? -1.0 : 1.0;
1157

    
1158
        fcb->no_repeat_mask |= 3 << fcb->n;
1159
        fcb->x[fcb->n]       = idx - delta;
1160
        fcb->y[fcb->n]       = v;
1161
        fcb->x[fcb->n + 1]   = idx;
1162
        fcb->y[fcb->n + 1]   = (val & 1) ? -v : v;
1163
        fcb->n              += 2;
1164
    }
1165
}
1166

    
1167
/**
1168
 * @}
1169
 *
1170
 * Generate a random number from frame_cntr and block_idx, which will lief
1171
 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1172
 * table of size 1000 of which you want to read block_size entries).
1173
 *
1174
 * @param frame_cntr current frame number
1175
 * @param block_num current block index
1176
 * @param block_size amount of entries we want to read from a table
1177
 *                   that has 1000 entries
1178
 * @return a (non-)random number in the [0, 1000 - block_size] range.
1179
 */
1180
static int pRNG(int frame_cntr, int block_num, int block_size)
1181
{
1182
    /* array to simplify the calculation of z:
1183
     * y = (x % 9) * 5 + 6;
1184
     * z = (49995 * x) / y;
1185
     * Since y only has 9 values, we can remove the division by using a
1186
     * LUT and using FASTDIV-style divisions. For each of the 9 values
1187
     * of y, we can rewrite z as:
1188
     * z = x * (49995 / y) + x * ((49995 % y) / y)
1189
     * In this table, each col represents one possible value of y, the
1190
     * first number is 49995 / y, and the second is the FASTDIV variant
1191
     * of 49995 % y / y. */
1192
    static const unsigned int div_tbl[9][2] = {
1193
        { 8332,  3 * 715827883U }, // y =  6
1194
        { 4545,  0 * 390451573U }, // y = 11
1195
        { 3124, 11 * 268435456U }, // y = 16
1196
        { 2380, 15 * 204522253U }, // y = 21
1197
        { 1922, 23 * 165191050U }, // y = 26
1198
        { 1612, 23 * 138547333U }, // y = 31
1199
        { 1388, 27 * 119304648U }, // y = 36
1200
        { 1219, 16 * 104755300U }, // y = 41
1201
        { 1086, 39 *  93368855U }  // y = 46
1202
    };
1203
    unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1204
    if (x >= 0xFFFF) x -= 0xFFFF;   // max value of x is 8*1877+0xFFFE=0x13AA6,
1205
                                    // so this is effectively a modulo (%)
1206
    y = x - 9 * MULH(477218589, x); // x % 9
1207
    z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1208
                                    // z = x * 49995 / (y * 5 + 6)
1209
    return z % (1000 - block_size);
1210
}
1211

    
1212
/**
1213
 * Parse hardcoded signal for a single block.
1214
 * @note see #synth_block().
1215
 */
1216
static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1217
                                 int block_idx, int size,
1218
                                 const struct frame_type_desc *frame_desc,
1219
                                 float *excitation)
1220
{
1221
    float gain;
1222
    int n, r_idx;
1223

    
1224
    assert(size <= MAX_FRAMESIZE);
1225

    
1226
    /* Set the offset from which we start reading wmavoice_std_codebook */
1227
    if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1228
        r_idx = pRNG(s->frame_cntr, block_idx, size);
1229
        gain  = s->silence_gain;
1230
    } else /* FCB_TYPE_HARDCODED */ {
1231
        r_idx = get_bits(gb, 8);
1232
        gain  = wmavoice_gain_universal[get_bits(gb, 6)];
1233
    }
1234

    
1235
    /* Clear gain prediction parameters */
1236
    memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1237

    
1238
    /* Apply gain to hardcoded codebook and use that as excitation signal */
1239
    for (n = 0; n < size; n++)
1240
        excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1241
}
1242

    
1243
/**
1244
 * Parse FCB/ACB signal for a single block.
1245
 * @note see #synth_block().
1246
 */
1247
static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1248
                                int block_idx, int size,
1249
                                int block_pitch_sh2,
1250
                                const struct frame_type_desc *frame_desc,
1251
                                float *excitation)
1252
{
1253
    static const float gain_coeff[6] = {
1254
        0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1255
    };
1256
    float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1257
    int n, idx, gain_weight;
1258
    AMRFixed fcb;
1259

    
1260
    assert(size <= MAX_FRAMESIZE / 2);
1261
    memset(pulses, 0, sizeof(*pulses) * size);
1262

    
1263
    fcb.pitch_lag      = block_pitch_sh2 >> 2;
1264
    fcb.pitch_fac      = 1.0;
1265
    fcb.no_repeat_mask = 0;
1266
    fcb.n              = 0;
1267

    
1268
    /* For the other frame types, this is where we apply the innovation
1269
     * (fixed) codebook pulses of the speech signal. */
1270
    if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1271
        aw_pulse_set1(s, gb, block_idx, &fcb);
1272
        aw_pulse_set2(s, gb, block_idx, &fcb);
1273
    } else /* FCB_TYPE_EXC_PULSES */ {
1274
        int offset_nbits = 5 - frame_desc->log_n_blocks;
1275

    
1276
        fcb.no_repeat_mask = -1;
1277
        /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1278
         * (instead of double) for a subset of pulses */
1279
        for (n = 0; n < 5; n++) {
1280
            float sign;
1281
            int pos1, pos2;
1282

    
1283
            sign           = get_bits1(gb) ? 1.0 : -1.0;
1284
            pos1           = get_bits(gb, offset_nbits);
1285
            fcb.x[fcb.n]   = n + 5 * pos1;
1286
            fcb.y[fcb.n++] = sign;
1287
            if (n < frame_desc->dbl_pulses) {
1288
                pos2           = get_bits(gb, offset_nbits);
1289
                fcb.x[fcb.n]   = n + 5 * pos2;
1290
                fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1291
            }
1292
        }
1293
    }
1294
    ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1295

    
1296
    /* Calculate gain for adaptive & fixed codebook signal.
1297
     * see ff_amr_set_fixed_gain(). */
1298
    idx = get_bits(gb, 7);
1299
    fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) -
1300
                    5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1301
    acb_gain = wmavoice_gain_codebook_acb[idx];
1302
    pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1303
                        -2.9957322736 /* log(0.05) */,
1304
                         1.6094379124 /* log(5.0)  */);
1305

    
1306
    gain_weight = 8 >> frame_desc->log_n_blocks;
1307
    memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1308
            sizeof(*s->gain_pred_err) * (6 - gain_weight));
1309
    for (n = 0; n < gain_weight; n++)
1310
        s->gain_pred_err[n] = pred_err;
1311

    
1312
    /* Calculation of adaptive codebook */
1313
    if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1314
        int len;
1315
        for (n = 0; n < size; n += len) {
1316
            int next_idx_sh16;
1317
            int abs_idx    = block_idx * size + n;
1318
            int pitch_sh16 = (s->last_pitch_val << 16) +
1319
                             s->pitch_diff_sh16 * abs_idx;
1320
            int pitch      = (pitch_sh16 + 0x6FFF) >> 16;
1321
            int idx_sh16   = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1322
            idx            = idx_sh16 >> 16;
1323
            if (s->pitch_diff_sh16) {
1324
                if (s->pitch_diff_sh16 > 0) {
1325
                    next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1326
                } else
1327
                    next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1328
                len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1329
                              1, size - n);
1330
            } else
1331
                len = size;
1332

    
1333
            ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1334
                                  wmavoice_ipol1_coeffs, 17,
1335
                                  idx, 9, len);
1336
        }
1337
    } else /* ACB_TYPE_HAMMING */ {
1338
        int block_pitch = block_pitch_sh2 >> 2;
1339
        idx             = block_pitch_sh2 & 3;
1340
        if (idx) {
1341
            ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1342
                                  wmavoice_ipol2_coeffs, 4,
1343
                                  idx, 8, size);
1344
        } else
1345
            av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1346
                              sizeof(float) * size);
1347
    }
1348

    
1349
    /* Interpolate ACB/FCB and use as excitation signal */
1350
    ff_weighted_vector_sumf(excitation, excitation, pulses,
1351
                            acb_gain, fcb_gain, size);
1352
}
1353

    
1354
/**
1355
 * Parse data in a single block.
1356
 * @note we assume enough bits are available, caller should check.
1357
 *
1358
 * @param s WMA Voice decoding context private data
1359
 * @param gb bit I/O context
1360
 * @param block_idx index of the to-be-read block
1361
 * @param size amount of samples to be read in this block
1362
 * @param block_pitch_sh2 pitch for this block << 2
1363
 * @param lsps LSPs for (the end of) this frame
1364
 * @param prev_lsps LSPs for the last frame
1365
 * @param frame_desc frame type descriptor
1366
 * @param excitation target memory for the ACB+FCB interpolated signal
1367
 * @param synth target memory for the speech synthesis filter output
1368
 * @return 0 on success, <0 on error.
1369
 */
1370
static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1371
                        int block_idx, int size,
1372
                        int block_pitch_sh2,
1373
                        const double *lsps, const double *prev_lsps,
1374
                        const struct frame_type_desc *frame_desc,
1375
                        float *excitation, float *synth)
1376
{
1377
    double i_lsps[MAX_LSPS];
1378
    float lpcs[MAX_LSPS];
1379
    float fac;
1380
    int n;
1381

    
1382
    if (frame_desc->acb_type == ACB_TYPE_NONE)
1383
        synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1384
    else
1385
        synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1386
                            frame_desc, excitation);
1387

    
1388
    /* convert interpolated LSPs to LPCs */
1389
    fac = (block_idx + 0.5) / frame_desc->n_blocks;
1390
    for (n = 0; n < s->lsps; n++) // LSF -> LSP
1391
        i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1392
    ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1393

    
1394
    /* Speech synthesis */
1395
    ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1396
}
1397

    
1398
/**
1399
 * Synthesize output samples for a single frame.
1400
 * @note we assume enough bits are available, caller should check.
1401
 *
1402
 * @param ctx WMA Voice decoder context
1403
 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1404
 * @param frame_idx Frame number within superframe [0-2]
1405
 * @param samples pointer to output sample buffer, has space for at least 160
1406
 *                samples
1407
 * @param lsps LSP array
1408
 * @param prev_lsps array of previous frame's LSPs
1409
 * @param excitation target buffer for excitation signal
1410
 * @param synth target buffer for synthesized speech data
1411
 * @return 0 on success, <0 on error.
1412
 */
1413
static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1414
                       float *samples,
1415
                       const double *lsps, const double *prev_lsps,
1416
                       float *excitation, float *synth)
1417
{
1418
    WMAVoiceContext *s = ctx->priv_data;
1419
    int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
1420
    int pitch[MAX_BLOCKS], last_block_pitch;
1421

    
1422
    /* Parse frame type ("frame header"), see frame_descs */
1423
    int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)],
1424
        block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1425

    
1426
    if (bd_idx < 0) {
1427
        av_log(ctx, AV_LOG_ERROR,
1428
               "Invalid frame type VLC code, skipping\n");
1429
        return -1;
1430
    }
1431

    
1432
    /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1433
    if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1434
        /* Pitch is provided per frame, which is interpreted as the pitch of
1435
         * the last sample of the last block of this frame. We can interpolate
1436
         * the pitch of other blocks (and even pitch-per-sample) by gradually
1437
         * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1438
        n_blocks_x2      = frame_descs[bd_idx].n_blocks << 1;
1439
        log_n_blocks_x2  = frame_descs[bd_idx].log_n_blocks + 1;
1440
        cur_pitch_val    = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1441
        cur_pitch_val    = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1442
        if (s->last_acb_type == ACB_TYPE_NONE ||
1443
            20 * abs(cur_pitch_val - s->last_pitch_val) >
1444
                (cur_pitch_val + s->last_pitch_val))
1445
            s->last_pitch_val = cur_pitch_val;
1446

    
1447
        /* pitch per block */
1448
        for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1449
            int fac = n * 2 + 1;
1450

    
1451
            pitch[n] = (MUL16(fac,                 cur_pitch_val) +
1452
                        MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1453
                        frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1454
        }
1455

    
1456
        /* "pitch-diff-per-sample" for calculation of pitch per sample */
1457
        s->pitch_diff_sh16 =
1458
            ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1459
    }
1460

    
1461
    /* Global gain (if silence) and pitch-adaptive window coordinates */
1462
    switch (frame_descs[bd_idx].fcb_type) {
1463
    case FCB_TYPE_SILENCE:
1464
        s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1465
        break;
1466
    case FCB_TYPE_AW_PULSES:
1467
        aw_parse_coords(s, gb, pitch);
1468
        break;
1469
    }
1470

    
1471
    for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1472
        int bl_pitch_sh2;
1473

    
1474
        /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1475
        switch (frame_descs[bd_idx].acb_type) {
1476
        case ACB_TYPE_HAMMING: {
1477
            /* Pitch is given per block. Per-block pitches are encoded as an
1478
             * absolute value for the first block, and then delta values
1479
             * relative to this value) for all subsequent blocks. The scale of
1480
             * this pitch value is semi-logaritmic compared to its use in the
1481
             * decoder, so we convert it to normal scale also. */
1482
            int block_pitch,
1483
                t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1484
                t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1485
                t3 =  s->block_conv_table[3] - s->block_conv_table[2] + 1;
1486

    
1487
            if (n == 0) {
1488
                block_pitch = get_bits(gb, s->block_pitch_nbits);
1489
            } else
1490
                block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1491
                                 get_bits(gb, s->block_delta_pitch_nbits);
1492
            /* Convert last_ so that any next delta is within _range */
1493
            last_block_pitch = av_clip(block_pitch,
1494
                                       s->block_delta_pitch_hrange,
1495
                                       s->block_pitch_range -
1496
                                           s->block_delta_pitch_hrange);
1497

    
1498
            /* Convert semi-log-style scale back to normal scale */
1499
            if (block_pitch < t1) {
1500
                bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1501
            } else {
1502
                block_pitch -= t1;
1503
                if (block_pitch < t2) {
1504
                    bl_pitch_sh2 =
1505
                        (s->block_conv_table[1] << 2) + (block_pitch << 1);
1506
                } else {
1507
                    block_pitch -= t2;
1508
                    if (block_pitch < t3) {
1509
                        bl_pitch_sh2 =
1510
                            (s->block_conv_table[2] + block_pitch) << 2;
1511
                    } else
1512
                        bl_pitch_sh2 = s->block_conv_table[3] << 2;
1513
                }
1514
            }
1515
            pitch[n] = bl_pitch_sh2 >> 2;
1516
            break;
1517
        }
1518

    
1519
        case ACB_TYPE_ASYMMETRIC: {
1520
            bl_pitch_sh2 = pitch[n] << 2;
1521
            break;
1522
        }
1523

    
1524
        default: // ACB_TYPE_NONE has no pitch
1525
            bl_pitch_sh2 = 0;
1526
            break;
1527
        }
1528

    
1529
        synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1530
                    lsps, prev_lsps, &frame_descs[bd_idx],
1531
                    &excitation[n * block_nsamples],
1532
                    &synth[n * block_nsamples]);
1533
    }
1534

    
1535
    /* Averaging projection filter, if applicable. Else, just copy samples
1536
     * from synthesis buffer */
1537
    if (s->do_apf) {
1538
        double i_lsps[MAX_LSPS];
1539
        float lpcs[MAX_LSPS];
1540

    
1541
        for (n = 0; n < s->lsps; n++) // LSF -> LSP
1542
            i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1543
        ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1544
        postfilter(s, synth, samples, 80, lpcs,
1545
                   &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1546
                   frame_descs[bd_idx].fcb_type, pitch[0]);
1547

    
1548
        for (n = 0; n < s->lsps; n++) // LSF -> LSP
1549
            i_lsps[n] = cos(lsps[n]);
1550
        ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1551
        postfilter(s, &synth[80], &samples[80], 80, lpcs,
1552
                   &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1553
                   frame_descs[bd_idx].fcb_type, pitch[0]);
1554
    } else
1555
        memcpy(samples, synth, 160 * sizeof(synth[0]));
1556

    
1557
    /* Cache values for next frame */
1558
    s->frame_cntr++;
1559
    if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1560
    s->last_acb_type = frame_descs[bd_idx].acb_type;
1561
    switch (frame_descs[bd_idx].acb_type) {
1562
    case ACB_TYPE_NONE:
1563
        s->last_pitch_val = 0;
1564
        break;
1565
    case ACB_TYPE_ASYMMETRIC:
1566
        s->last_pitch_val = cur_pitch_val;
1567
        break;
1568
    case ACB_TYPE_HAMMING:
1569
        s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1570
        break;
1571
    }
1572

    
1573
    return 0;
1574
}
1575

    
1576
/**
1577
 * Ensure minimum value for first item, maximum value for last value,
1578
 * proper spacing between each value and proper ordering.
1579
 *
1580
 * @param lsps array of LSPs
1581
 * @param num size of LSP array
1582
 *
1583
 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1584
 *       useful to put in a generic location later on. Parts are also
1585
 *       present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1586
 *       which is in float.
1587
 */
1588
static void stabilize_lsps(double *lsps, int num)
1589
{
1590
    int n, m, l;
1591

    
1592
    /* set minimum value for first, maximum value for last and minimum
1593
     * spacing between LSF values.
1594
     * Very similar to ff_set_min_dist_lsf(), but in double. */
1595
    lsps[0]       = FFMAX(lsps[0],       0.0015 * M_PI);
1596
    for (n = 1; n < num; n++)
1597
        lsps[n]   = FFMAX(lsps[n],       lsps[n - 1] + 0.0125 * M_PI);
1598
    lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1599

    
1600
    /* reorder (looks like one-time / non-recursed bubblesort).
1601
     * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1602
    for (n = 1; n < num; n++) {
1603
        if (lsps[n] < lsps[n - 1]) {
1604
            for (m = 1; m < num; m++) {
1605
                double tmp = lsps[m];
1606
                for (l = m - 1; l >= 0; l--) {
1607
                    if (lsps[l] <= tmp) break;
1608
                    lsps[l + 1] = lsps[l];
1609
                }
1610
                lsps[l + 1] = tmp;
1611
            }
1612
            break;
1613
        }
1614
    }
1615
}
1616

    
1617
/**
1618
 * Test if there's enough bits to read 1 superframe.
1619
 *
1620
 * @param orig_gb bit I/O context used for reading. This function
1621
 *                does not modify the state of the bitreader; it
1622
 *                only uses it to copy the current stream position
1623
 * @param s WMA Voice decoding context private data
1624
 * @return -1 if unsupported, 1 on not enough bits or 0 if OK.
1625
 */
1626
static int check_bits_for_superframe(GetBitContext *orig_gb,
1627
                                     WMAVoiceContext *s)
1628
{
1629
    GetBitContext s_gb, *gb = &s_gb;
1630
    int n, need_bits, bd_idx;
1631
    const struct frame_type_desc *frame_desc;
1632

    
1633
    /* initialize a copy */
1634
    init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
1635
    skip_bits_long(gb, get_bits_count(orig_gb));
1636
    assert(get_bits_left(gb) == get_bits_left(orig_gb));
1637

    
1638
    /* superframe header */
1639
    if (get_bits_left(gb) < 14)
1640
        return 1;
1641
    if (!get_bits1(gb))
1642
        return -1;                        // WMAPro-in-WMAVoice superframe
1643
    if (get_bits1(gb)) skip_bits(gb, 12); // number of  samples in superframe
1644
    if (s->has_residual_lsps) {           // residual LSPs (for all frames)
1645
        if (get_bits_left(gb) < s->sframe_lsp_bitsize)
1646
            return 1;
1647
        skip_bits_long(gb, s->sframe_lsp_bitsize);
1648
    }
1649

    
1650
    /* frames */
1651
    for (n = 0; n < MAX_FRAMES; n++) {
1652
        int aw_idx_is_ext = 0;
1653

    
1654
        if (!s->has_residual_lsps) {     // independent LSPs (per-frame)
1655
           if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
1656
           skip_bits_long(gb, s->frame_lsp_bitsize);
1657
        }
1658
        bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
1659
        if (bd_idx < 0)
1660
            return -1;                   // invalid frame type VLC code
1661
        frame_desc = &frame_descs[bd_idx];
1662
        if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1663
            if (get_bits_left(gb) < s->pitch_nbits)
1664
                return 1;
1665
            skip_bits_long(gb, s->pitch_nbits);
1666
        }
1667
        if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1668
            skip_bits(gb, 8);
1669
        } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1670
            int tmp = get_bits(gb, 6);
1671
            if (tmp >= 0x36) {
1672
                skip_bits(gb, 2);
1673
                aw_idx_is_ext = 1;
1674
            }
1675
        }
1676

    
1677
        /* blocks */
1678
        if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
1679
            need_bits = s->block_pitch_nbits +
1680
                (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
1681
        } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1682
            need_bits = 2 * !aw_idx_is_ext;
1683
        } else
1684
            need_bits = 0;
1685
        need_bits += frame_desc->frame_size;
1686
        if (get_bits_left(gb) < need_bits)
1687
            return 1;
1688
        skip_bits_long(gb, need_bits);
1689
    }
1690

    
1691
    return 0;
1692
}
1693

    
1694
/**
1695
 * Synthesize output samples for a single superframe. If we have any data
1696
 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1697
 * in s->gb.
1698
 *
1699
 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1700
 * to give a total of 480 samples per frame. See #synth_frame() for frame
1701
 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1702
 * (if these are globally specified for all frames (residually); they can
1703
 * also be specified individually per-frame. See the s->has_residual_lsps
1704
 * option), and can specify the number of samples encoded in this superframe
1705
 * (if less than 480), usually used to prevent blanks at track boundaries.
1706
 *
1707
 * @param ctx WMA Voice decoder context
1708
 * @param samples pointer to output buffer for voice samples
1709
 * @param data_size pointer containing the size of #samples on input, and the
1710
 *                  amount of #samples filled on output
1711
 * @return 0 on success, <0 on error or 1 if there was not enough data to
1712
 *         fully parse the superframe
1713
 */
1714
static int synth_superframe(AVCodecContext *ctx,
1715
                            float *samples, int *data_size)
1716
{
1717
    WMAVoiceContext *s = ctx->priv_data;
1718
    GetBitContext *gb = &s->gb, s_gb;
1719
    int n, res, n_samples = 480;
1720
    double lsps[MAX_FRAMES][MAX_LSPS];
1721
    const double *mean_lsf = s->lsps == 16 ?
1722
        wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1723
    float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1724
    float synth[MAX_LSPS + MAX_SFRAMESIZE];
1725

    
1726
    memcpy(synth,      s->synth_history,
1727
           s->lsps             * sizeof(*synth));
1728
    memcpy(excitation, s->excitation_history,
1729
           s->history_nsamples * sizeof(*excitation));
1730

    
1731
    if (s->sframe_cache_size > 0) {
1732
        gb = &s_gb;
1733
        init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1734
        s->sframe_cache_size = 0;
1735
    }
1736

    
1737
    if ((res = check_bits_for_superframe(gb, s)) == 1) return 1;
1738

    
1739
    /* First bit is speech/music bit, it differentiates between WMAVoice
1740
     * speech samples (the actual codec) and WMAVoice music samples, which
1741
     * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1742
     * the wild yet. */
1743
    if (!get_bits1(gb)) {
1744
        av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1);
1745
        return -1;
1746
    }
1747

    
1748
    /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1749
    if (get_bits1(gb)) {
1750
        if ((n_samples = get_bits(gb, 12)) > 480) {
1751
            av_log(ctx, AV_LOG_ERROR,
1752
                   "Superframe encodes >480 samples (%d), not allowed\n",
1753
                   n_samples);
1754
            return -1;
1755
        }
1756
    }
1757
    /* Parse LSPs, if global for the superframe (can also be per-frame). */
1758
    if (s->has_residual_lsps) {
1759
        double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1760

    
1761
        for (n = 0; n < s->lsps; n++)
1762
            prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1763

    
1764
        if (s->lsps == 10) {
1765
            dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1766
        } else /* s->lsps == 16 */
1767
            dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1768

    
1769
        for (n = 0; n < s->lsps; n++) {
1770
            lsps[0][n]  = mean_lsf[n] + (a1[n]           - a2[n * 2]);
1771
            lsps[1][n]  = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1772
            lsps[2][n] += mean_lsf[n];
1773
        }
1774
        for (n = 0; n < 3; n++)
1775
            stabilize_lsps(lsps[n], s->lsps);
1776
    }
1777

    
1778
    /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
1779
    for (n = 0; n < 3; n++) {
1780
        if (!s->has_residual_lsps) {
1781
            int m;
1782

    
1783
            if (s->lsps == 10) {
1784
                dequant_lsp10i(gb, lsps[n]);
1785
            } else /* s->lsps == 16 */
1786
                dequant_lsp16i(gb, lsps[n]);
1787

    
1788
            for (m = 0; m < s->lsps; m++)
1789
                lsps[n][m] += mean_lsf[m];
1790
            stabilize_lsps(lsps[n], s->lsps);
1791
        }
1792

    
1793
        if ((res = synth_frame(ctx, gb, n,
1794
                               &samples[n * MAX_FRAMESIZE],
1795
                               lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1796
                               &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1797
                               &synth[s->lsps + n * MAX_FRAMESIZE])))
1798
            return res;
1799
    }
1800

    
1801
    /* Statistics? FIXME - we don't check for length, a slight overrun
1802
     * will be caught by internal buffer padding, and anything else
1803
     * will be skipped, not read. */
1804
    if (get_bits1(gb)) {
1805
        res = get_bits(gb, 4);
1806
        skip_bits(gb, 10 * (res + 1));
1807
    }
1808

    
1809
    /* Specify nr. of output samples */
1810
    *data_size = n_samples * sizeof(float);
1811

    
1812
    /* Update history */
1813
    memcpy(s->prev_lsps,           lsps[2],
1814
           s->lsps             * sizeof(*s->prev_lsps));
1815
    memcpy(s->synth_history,      &synth[MAX_SFRAMESIZE],
1816
           s->lsps             * sizeof(*synth));
1817
    memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1818
           s->history_nsamples * sizeof(*excitation));
1819
    if (s->do_apf)
1820
        memmove(s->zero_exc_pf,       &s->zero_exc_pf[MAX_SFRAMESIZE],
1821
                s->history_nsamples * sizeof(*s->zero_exc_pf));
1822

    
1823
    return 0;
1824
}
1825

    
1826
/**
1827
 * Parse the packet header at the start of each packet (input data to this
1828
 * decoder).
1829
 *
1830
 * @param s WMA Voice decoding context private data
1831
 * @return 1 if not enough bits were available, or 0 on success.
1832
 */
1833
static int parse_packet_header(WMAVoiceContext *s)
1834
{
1835
    GetBitContext *gb = &s->gb;
1836
    unsigned int res;
1837

    
1838
    if (get_bits_left(gb) < 11)
1839
        return 1;
1840
    skip_bits(gb, 4);          // packet sequence number
1841
    s->has_residual_lsps = get_bits1(gb);
1842
    do {
1843
        res = get_bits(gb, 6); // number of superframes per packet
1844
                               // (minus first one if there is spillover)
1845
        if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
1846
            return 1;
1847
    } while (res == 0x3F);
1848
    s->spillover_nbits   = get_bits(gb, s->spillover_bitsize);
1849

    
1850
    return 0;
1851
}
1852

    
1853
/**
1854
 * Copy (unaligned) bits from gb/data/size to pb.
1855
 *
1856
 * @param pb target buffer to copy bits into
1857
 * @param data source buffer to copy bits from
1858
 * @param size size of the source data, in bytes
1859
 * @param gb bit I/O context specifying the current position in the source.
1860
 *           data. This function might use this to align the bit position to
1861
 *           a whole-byte boundary before calling #ff_copy_bits() on aligned
1862
 *           source data
1863
 * @param nbits the amount of bits to copy from source to target
1864
 *
1865
 * @note after calling this function, the current position in the input bit
1866
 *       I/O context is undefined.
1867
 */
1868
static void copy_bits(PutBitContext *pb,
1869
                      const uint8_t *data, int size,
1870
                      GetBitContext *gb, int nbits)
1871
{
1872
    int rmn_bytes, rmn_bits;
1873

    
1874
    rmn_bits = rmn_bytes = get_bits_left(gb);
1875
    if (rmn_bits < nbits)
1876
        return;
1877
    rmn_bits &= 7; rmn_bytes >>= 3;
1878
    if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1879
        put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1880
    ff_copy_bits(pb, data + size - rmn_bytes,
1881
                 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1882
}
1883

    
1884
/**
1885
 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1886
 * and we expect that the demuxer / application provides it to us as such
1887
 * (else you'll probably get garbage as output). Every packet has a size of
1888
 * ctx->block_align bytes, starts with a packet header (see
1889
 * #parse_packet_header()), and then a series of superframes. Superframe
1890
 * boundaries may exceed packets, i.e. superframes can split data over
1891
 * multiple (two) packets.
1892
 *
1893
 * For more information about frames, see #synth_superframe().
1894
 */
1895
static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1896
                                  int *data_size, AVPacket *avpkt)
1897
{
1898
    WMAVoiceContext *s = ctx->priv_data;
1899
    GetBitContext *gb = &s->gb;
1900
    int size, res, pos;
1901

    
1902
    if (*data_size < 480 * sizeof(float)) {
1903
        av_log(ctx, AV_LOG_ERROR,
1904
               "Output buffer too small (%d given - %lu needed)\n",
1905
               *data_size, 480 * sizeof(float));
1906
        return -1;
1907
    }
1908
    *data_size = 0;
1909

    
1910
    /* Packets are sometimes a multiple of ctx->block_align, with a packet
1911
     * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1912
     * feeds us ASF packets, which may concatenate multiple "codec" packets
1913
     * in a single "muxer" packet, so we artificially emulate that by
1914
     * capping the packet size at ctx->block_align. */
1915
    for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1916
    if (!size)
1917
        return 0;
1918
    init_get_bits(&s->gb, avpkt->data, size << 3);
1919

    
1920
    /* size == ctx->block_align is used to indicate whether we are dealing with
1921
     * a new packet or a packet of which we already read the packet header
1922
     * previously. */
1923
    if (size == ctx->block_align) { // new packet header
1924
        if ((res = parse_packet_header(s)) < 0)
1925
            return res;
1926

    
1927
        /* If the packet header specifies a s->spillover_nbits, then we want
1928
         * to push out all data of the previous packet (+ spillover) before
1929
         * continuing to parse new superframes in the current packet. */
1930
        if (s->spillover_nbits > 0) {
1931
            if (s->sframe_cache_size > 0) {
1932
                int cnt = get_bits_count(gb);
1933
                copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1934
                flush_put_bits(&s->pb);
1935
                s->sframe_cache_size += s->spillover_nbits;
1936
                if ((res = synth_superframe(ctx, data, data_size)) == 0 &&
1937
                    *data_size > 0) {
1938
                    cnt += s->spillover_nbits;
1939
                    s->skip_bits_next = cnt & 7;
1940
                    return cnt >> 3;
1941
                } else
1942
                    skip_bits_long (gb, s->spillover_nbits - cnt +
1943
                                    get_bits_count(gb)); // resync
1944
            } else
1945
                skip_bits_long(gb, s->spillover_nbits);  // resync
1946
        }
1947
    } else if (s->skip_bits_next)
1948
        skip_bits(gb, s->skip_bits_next);
1949

    
1950
    /* Try parsing superframes in current packet */
1951
    s->sframe_cache_size = 0;
1952
    s->skip_bits_next = 0;
1953
    pos = get_bits_left(gb);
1954
    if ((res = synth_superframe(ctx, data, data_size)) < 0) {
1955
        return res;
1956
    } else if (*data_size > 0) {
1957
        int cnt = get_bits_count(gb);
1958
        s->skip_bits_next = cnt & 7;
1959
        return cnt >> 3;
1960
    } else if ((s->sframe_cache_size = pos) > 0) {
1961
        /* rewind bit reader to start of last (incomplete) superframe... */
1962
        init_get_bits(gb, avpkt->data, size << 3);
1963
        skip_bits_long(gb, (size << 3) - pos);
1964
        assert(get_bits_left(gb) == pos);
1965

    
1966
        /* ...and cache it for spillover in next packet */
1967
        init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
1968
        copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
1969
        // FIXME bad - just copy bytes as whole and add use the
1970
        // skip_bits_next field
1971
    }
1972

    
1973
    return size;
1974
}
1975

    
1976
static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
1977
{
1978
    WMAVoiceContext *s = ctx->priv_data;
1979

    
1980
    if (s->do_apf) {
1981
        ff_rdft_end(&s->rdft);
1982
        ff_rdft_end(&s->irdft);
1983
        ff_dct_end(&s->dct);
1984
        ff_dct_end(&s->dst);
1985
    }
1986

    
1987
    return 0;
1988
}
1989

    
1990
static av_cold void wmavoice_flush(AVCodecContext *ctx)
1991
{
1992
    WMAVoiceContext *s = ctx->priv_data;
1993
    int n;
1994

    
1995
    s->postfilter_agc    = 0;
1996
    s->sframe_cache_size = 0;
1997
    s->skip_bits_next    = 0;
1998
    for (n = 0; n < s->lsps; n++)
1999
        s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2000
    memset(s->excitation_history, 0,
2001
           sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2002
    memset(s->synth_history,      0,
2003
           sizeof(*s->synth_history)      * MAX_LSPS);
2004
    memset(s->gain_pred_err,      0,
2005
           sizeof(s->gain_pred_err));
2006

    
2007
    if (s->do_apf) {
2008
        memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2009
               sizeof(*s->synth_filter_out_buf) * s->lsps);
2010
        memset(s->dcf_mem,              0,
2011
               sizeof(*s->dcf_mem)              * 2);
2012
        memset(s->zero_exc_pf,          0,
2013
               sizeof(*s->zero_exc_pf)          * s->history_nsamples);
2014
        memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2015
    }
2016
}
2017

    
2018
AVCodec wmavoice_decoder = {
2019
    "wmavoice",
2020
    AVMEDIA_TYPE_AUDIO,
2021
    CODEC_ID_WMAVOICE,
2022
    sizeof(WMAVoiceContext),
2023
    wmavoice_decode_init,
2024
    NULL,
2025
    wmavoice_decode_end,
2026
    wmavoice_decode_packet,
2027
    CODEC_CAP_SUBFRAMES,
2028
    .flush     = wmavoice_flush,
2029
    .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
2030
};