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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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/**
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 * Frame type VLC coding.
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 */
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static VLC frame_type_vlc;
59

    
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/**
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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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};
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/**
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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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 */
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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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     * @{
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     */
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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
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    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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    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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    /**
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     * @}
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
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    float postfilter_agc;         ///< gain control memory, used in
270
                                  ///< #adaptive_gain_control()
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    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
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    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
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                                  ///< synthesis
284
    /**
285
     * @}
286
     */
287
} WMAVoiceContext;
288

    
289
/**
290
 * Set 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 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 zero_exc_pf destination for zero synthesis filter (16-byte aligned)
748
 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
749
 * @param pitch Pitch of the input signal
750
 */
751
static void postfilter(WMAVoiceContext *s, const float *synth,
752
                       float *samples,    int size,
753
                       const float *lpcs, float *zero_exc_pf,
754
                       int fcb_type,      int pitch)
755
{
756
    float synth_filter_in_buf[MAX_FRAMESIZE / 2],
757
          *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
758
          *synth_filter_in = zero_exc_pf;
759

    
760
    assert(size <= MAX_FRAMESIZE / 2);
761

    
762
    /* generate excitation from input signal */
763
    ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
764

    
765
    if (fcb_type >= FCB_TYPE_AW_PULSES &&
766
        !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
767
        synth_filter_in = synth_filter_in_buf;
768

    
769
    /* re-synthesize speech after smoothening, and keep history */
770
    ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
771
                                 synth_filter_in, size, s->lsps);
772
    memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
773
           sizeof(synth_pf[0]) * s->lsps);
774

    
775
    wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
776

    
777
    adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
778
                          &s->postfilter_agc);
779

    
780
    if (s->dc_level > 8) {
781
        /* remove ultra-low frequency DC noise / highpass filter;
782
         * coefficients are identical to those used in SIPR decoding,
783
         * and very closely resemble those used in AMR-NB decoding. */
784
        ff_acelp_apply_order_2_transfer_function(samples, samples,
785
            (const float[2]) { -1.99997,      1.0 },
786
            (const float[2]) { -1.9330735188, 0.93589198496 },
787
            0.93980580475, s->dcf_mem, size);
788
    }
789
}
790
/**
791
 * @}
792
 */
793

    
794
/**
795
 * Dequantize LSPs
796
 * @param lsps output pointer to the array that will hold the LSPs
797
 * @param num number of LSPs to be dequantized
798
 * @param values quantized values, contains n_stages values
799
 * @param sizes range (i.e. max value) of each quantized value
800
 * @param n_stages number of dequantization runs
801
 * @param table dequantization table to be used
802
 * @param mul_q LSF multiplier
803
 * @param base_q base (lowest) LSF values
804
 */
805
static void dequant_lsps(double *lsps, int num,
806
                         const uint16_t *values,
807
                         const uint16_t *sizes,
808
                         int n_stages, const uint8_t *table,
809
                         const double *mul_q,
810
                         const double *base_q)
811
{
812
    int n, m;
813

    
814
    memset(lsps, 0, num * sizeof(*lsps));
815
    for (n = 0; n < n_stages; n++) {
816
        const uint8_t *t_off = &table[values[n] * num];
817
        double base = base_q[n], mul = mul_q[n];
818

    
819
        for (m = 0; m < num; m++)
820
            lsps[m] += base + mul * t_off[m];
821

    
822
        table += sizes[n] * num;
823
    }
824
}
825

    
826
/**
827
 * @defgroup lsp_dequant LSP dequantization routines
828
 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
829
 * @note we assume enough bits are available, caller should check.
830
 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
831
 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
832
 * @{
833
 */
834
/**
835
 * Parse 10 independently-coded LSPs.
836
 */
837
static void dequant_lsp10i(GetBitContext *gb, double *lsps)
838
{
839
    static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
840
    static const double mul_lsf[4] = {
841
        5.2187144800e-3,    1.4626986422e-3,
842
        9.6179549166e-4,    1.1325736225e-3
843
    };
844
    static const double base_lsf[4] = {
845
        M_PI * -2.15522e-1, M_PI * -6.1646e-2,
846
        M_PI * -3.3486e-2,  M_PI * -5.7408e-2
847
    };
848
    uint16_t v[4];
849

    
850
    v[0] = get_bits(gb, 8);
851
    v[1] = get_bits(gb, 6);
852
    v[2] = get_bits(gb, 5);
853
    v[3] = get_bits(gb, 5);
854

    
855
    dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
856
                 mul_lsf, base_lsf);
857
}
858

    
859
/**
860
 * Parse 10 independently-coded LSPs, and then derive the tables to
861
 * generate LSPs for the other frames from them (residual coding).
862
 */
863
static void dequant_lsp10r(GetBitContext *gb,
864
                           double *i_lsps, const double *old,
865
                           double *a1, double *a2, int q_mode)
866
{
867
    static const uint16_t vec_sizes[3] = { 128, 64, 64 };
868
    static const double mul_lsf[3] = {
869
        2.5807601174e-3,    1.2354460219e-3,   1.1763821673e-3
870
    };
871
    static const double base_lsf[3] = {
872
        M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
873
    };
874
    const float (*ipol_tab)[2][10] = q_mode ?
875
        wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
876
    uint16_t interpol, v[3];
877
    int n;
878

    
879
    dequant_lsp10i(gb, i_lsps);
880

    
881
    interpol = get_bits(gb, 5);
882
    v[0]     = get_bits(gb, 7);
883
    v[1]     = get_bits(gb, 6);
884
    v[2]     = get_bits(gb, 6);
885

    
886
    for (n = 0; n < 10; n++) {
887
        double delta = old[n] - i_lsps[n];
888
        a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
889
        a1[10 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
890
    }
891

    
892
    dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
893
                 mul_lsf, base_lsf);
894
}
895

    
896
/**
897
 * Parse 16 independently-coded LSPs.
898
 */
899
static void dequant_lsp16i(GetBitContext *gb, double *lsps)
900
{
901
    static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
902
    static const double mul_lsf[5] = {
903
        3.3439586280e-3,    6.9908173703e-4,
904
        3.3216608306e-3,    1.0334960326e-3,
905
        3.1899104283e-3
906
    };
907
    static const double base_lsf[5] = {
908
        M_PI * -1.27576e-1, M_PI * -2.4292e-2,
909
        M_PI * -1.28094e-1, M_PI * -3.2128e-2,
910
        M_PI * -1.29816e-1
911
    };
912
    uint16_t v[5];
913

    
914
    v[0] = get_bits(gb, 8);
915
    v[1] = get_bits(gb, 6);
916
    v[2] = get_bits(gb, 7);
917
    v[3] = get_bits(gb, 6);
918
    v[4] = get_bits(gb, 7);
919

    
920
    dequant_lsps( lsps,     5,  v,     vec_sizes,    2,
921
                 wmavoice_dq_lsp16i1,  mul_lsf,     base_lsf);
922
    dequant_lsps(&lsps[5],  5, &v[2], &vec_sizes[2], 2,
923
                 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
924
    dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
925
                 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
926
}
927

    
928
/**
929
 * Parse 16 independently-coded LSPs, and then derive the tables to
930
 * generate LSPs for the other frames from them (residual coding).
931
 */
932
static void dequant_lsp16r(GetBitContext *gb,
933
                           double *i_lsps, const double *old,
934
                           double *a1, double *a2, int q_mode)
935
{
936
    static const uint16_t vec_sizes[3] = { 128, 128, 128 };
937
    static const double mul_lsf[3] = {
938
        1.2232979501e-3,   1.4062241527e-3,   1.6114744851e-3
939
    };
940
    static const double base_lsf[3] = {
941
        M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
942
    };
943
    const float (*ipol_tab)[2][16] = q_mode ?
944
        wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
945
    uint16_t interpol, v[3];
946
    int n;
947

    
948
    dequant_lsp16i(gb, i_lsps);
949

    
950
    interpol = get_bits(gb, 5);
951
    v[0]     = get_bits(gb, 7);
952
    v[1]     = get_bits(gb, 7);
953
    v[2]     = get_bits(gb, 7);
954

    
955
    for (n = 0; n < 16; n++) {
956
        double delta = old[n] - i_lsps[n];
957
        a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
958
        a1[16 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
959
    }
960

    
961
    dequant_lsps( a2,     10,  v,     vec_sizes,    1,
962
                 wmavoice_dq_lsp16r1,  mul_lsf,     base_lsf);
963
    dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
964
                 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
965
    dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
966
                 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
967
}
968

    
969
/**
970
 * @}
971
 * @defgroup aw Pitch-adaptive window coding functions
972
 * The next few functions are for pitch-adaptive window coding.
973
 * @{
974
 */
975
/**
976
 * Parse the offset of the first pitch-adaptive window pulses, and
977
 * the distribution of pulses between the two blocks in this frame.
978
 * @param s WMA Voice decoding context private data
979
 * @param gb bit I/O context
980
 * @param pitch pitch for each block in this frame
981
 */
982
static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
983
                            const int *pitch)
984
{
985
    static const int16_t start_offset[94] = {
986
        -11,  -9,  -7,  -5,  -3,  -1,   1,   3,   5,   7,   9,  11,
987
         13,  15,  18,  17,  19,  20,  21,  22,  23,  24,  25,  26,
988
         27,  28,  29,  30,  31,  32,  33,  35,  37,  39,  41,  43,
989
         45,  47,  49,  51,  53,  55,  57,  59,  61,  63,  65,  67,
990
         69,  71,  73,  75,  77,  79,  81,  83,  85,  87,  89,  91,
991
         93,  95,  97,  99, 101, 103, 105, 107, 109, 111, 113, 115,
992
        117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
993
        141, 143, 145, 147, 149, 151, 153, 155, 157, 159
994
    };
995
    int bits, offset;
996

    
997
    /* position of pulse */
998
    s->aw_idx_is_ext = 0;
999
    if ((bits = get_bits(gb, 6)) >= 54) {
1000
        s->aw_idx_is_ext = 1;
1001
        bits += (bits - 54) * 3 + get_bits(gb, 2);
1002
    }
1003

    
1004
    /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1005
     * the distribution of the pulses in each block contained in this frame. */
1006
    s->aw_pulse_range        = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1007
    for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1008
    s->aw_n_pulses[0]        = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1009
    s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1010
    offset                  += s->aw_n_pulses[0] * pitch[0];
1011
    s->aw_n_pulses[1]        = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1012
    s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1013

    
1014
    /* if continuing from a position before the block, reset position to
1015
     * start of block (when corrected for the range over which it can be
1016
     * spread in aw_pulse_set1()). */
1017
    if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1018
        while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1019
            s->aw_first_pulse_off[1] -= pitch[1];
1020
        if (start_offset[bits] < 0)
1021
            while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1022
                s->aw_first_pulse_off[0] -= pitch[0];
1023
    }
1024
}
1025

    
1026
/**
1027
 * Apply second set of pitch-adaptive window pulses.
1028
 * @param s WMA Voice decoding context private data
1029
 * @param gb bit I/O context
1030
 * @param block_idx block index in frame [0, 1]
1031
 * @param fcb structure containing fixed codebook vector info
1032
 */
1033
static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1034
                          int block_idx, AMRFixed *fcb)
1035
{
1036
    uint16_t use_mask[7]; // only 5 are used, rest is padding
1037
    /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1038
     * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1039
     * of idx are the position of the bit within a particular item in the
1040
     * array (0 being the most significant bit, and 15 being the least
1041
     * significant bit), and the remainder (>> 4) is the index in the
1042
     * use_mask[]-array. This is faster and uses less memory than using a
1043
     * 80-byte/80-int array. */
1044
    int pulse_off = s->aw_first_pulse_off[block_idx],
1045
        pulse_start, n, idx, range, aidx, start_off = 0;
1046

    
1047
    /* set offset of first pulse to within this block */
1048
    if (s->aw_n_pulses[block_idx] > 0)
1049
        while (pulse_off + s->aw_pulse_range < 1)
1050
            pulse_off += fcb->pitch_lag;
1051

    
1052
    /* find range per pulse */
1053
    if (s->aw_n_pulses[0] > 0) {
1054
        if (block_idx == 0) {
1055
            range = 32;
1056
        } else /* block_idx = 1 */ {
1057
            range = 8;
1058
            if (s->aw_n_pulses[block_idx] > 0)
1059
                pulse_off = s->aw_next_pulse_off_cache;
1060
        }
1061
    } else
1062
        range = 16;
1063
    pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1064

    
1065
    /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1066
     * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1067
     * we exclude that range from being pulsed again in this function. */
1068
    memset( use_mask,   -1, 5 * sizeof(use_mask[0]));
1069
    memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1070
    if (s->aw_n_pulses[block_idx] > 0)
1071
        for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1072
            int excl_range         = s->aw_pulse_range; // always 16 or 24
1073
            uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1074
            int first_sh           = 16 - (idx & 15);
1075
            *use_mask_ptr++       &= 0xFFFF << first_sh;
1076
            excl_range            -= first_sh;
1077
            if (excl_range >= 16) {
1078
                *use_mask_ptr++    = 0;
1079
                *use_mask_ptr     &= 0xFFFF >> (excl_range - 16);
1080
            } else
1081
                *use_mask_ptr     &= 0xFFFF >> excl_range;
1082
        }
1083

    
1084
    /* find the 'aidx'th offset that is not excluded */
1085
    aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1086
    for (n = 0; n <= aidx; pulse_start++) {
1087
        for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1088
        if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1089
            if (use_mask[0])      idx = 0x0F;
1090
            else if (use_mask[1]) idx = 0x1F;
1091
            else if (use_mask[2]) idx = 0x2F;
1092
            else if (use_mask[3]) idx = 0x3F;
1093
            else if (use_mask[4]) idx = 0x4F;
1094
            else                  return;
1095
            idx -= av_log2_16bit(use_mask[idx >> 4]);
1096
        }
1097
        if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1098
            use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1099
            n++;
1100
            start_off = idx;
1101
        }
1102
    }
1103

    
1104
    fcb->x[fcb->n] = start_off;
1105
    fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1106
    fcb->n++;
1107

    
1108
    /* set offset for next block, relative to start of that block */
1109
    n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1110
    s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1111
}
1112

    
1113
/**
1114
 * Apply first set of pitch-adaptive window pulses.
1115
 * @param s WMA Voice decoding context private data
1116
 * @param gb bit I/O context
1117
 * @param block_idx block index in frame [0, 1]
1118
 * @param fcb storage location for fixed codebook pulse info
1119
 */
1120
static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1121
                          int block_idx, AMRFixed *fcb)
1122
{
1123
    int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1124
    float v;
1125

    
1126
    if (s->aw_n_pulses[block_idx] > 0) {
1127
        int n, v_mask, i_mask, sh, n_pulses;
1128

    
1129
        if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1130
            n_pulses = 3;
1131
            v_mask   = 8;
1132
            i_mask   = 7;
1133
            sh       = 4;
1134
        } else { // 4 pulses, 1:sign + 2:index each
1135
            n_pulses = 4;
1136
            v_mask   = 4;
1137
            i_mask   = 3;
1138
            sh       = 3;
1139
        }
1140

    
1141
        for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1142
            fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1143
            fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1144
                                 s->aw_first_pulse_off[block_idx];
1145
            while (fcb->x[fcb->n] < 0)
1146
                fcb->x[fcb->n] += fcb->pitch_lag;
1147
            if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1148
                fcb->n++;
1149
        }
1150
    } else {
1151
        int num2 = (val & 0x1FF) >> 1, delta, idx;
1152

    
1153
        if (num2 < 1 * 79)      { delta = 1; idx = num2 + 1; }
1154
        else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1155
        else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1156
        else                    { delta = 7; idx = num2 + 1 - 3 * 75; }
1157
        v = (val & 0x200) ? -1.0 : 1.0;
1158

    
1159
        fcb->no_repeat_mask |= 3 << fcb->n;
1160
        fcb->x[fcb->n]       = idx - delta;
1161
        fcb->y[fcb->n]       = v;
1162
        fcb->x[fcb->n + 1]   = idx;
1163
        fcb->y[fcb->n + 1]   = (val & 1) ? -v : v;
1164
        fcb->n              += 2;
1165
    }
1166
}
1167

    
1168
/**
1169
 * @}
1170
 *
1171
 * Generate a random number from frame_cntr and block_idx, which will lief
1172
 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1173
 * table of size 1000 of which you want to read block_size entries).
1174
 *
1175
 * @param frame_cntr current frame number
1176
 * @param block_num current block index
1177
 * @param block_size amount of entries we want to read from a table
1178
 *                   that has 1000 entries
1179
 * @return a (non-)random number in the [0, 1000 - block_size] range.
1180
 */
1181
static int pRNG(int frame_cntr, int block_num, int block_size)
1182
{
1183
    /* array to simplify the calculation of z:
1184
     * y = (x % 9) * 5 + 6;
1185
     * z = (49995 * x) / y;
1186
     * Since y only has 9 values, we can remove the division by using a
1187
     * LUT and using FASTDIV-style divisions. For each of the 9 values
1188
     * of y, we can rewrite z as:
1189
     * z = x * (49995 / y) + x * ((49995 % y) / y)
1190
     * In this table, each col represents one possible value of y, the
1191
     * first number is 49995 / y, and the second is the FASTDIV variant
1192
     * of 49995 % y / y. */
1193
    static const unsigned int div_tbl[9][2] = {
1194
        { 8332,  3 * 715827883U }, // y =  6
1195
        { 4545,  0 * 390451573U }, // y = 11
1196
        { 3124, 11 * 268435456U }, // y = 16
1197
        { 2380, 15 * 204522253U }, // y = 21
1198
        { 1922, 23 * 165191050U }, // y = 26
1199
        { 1612, 23 * 138547333U }, // y = 31
1200
        { 1388, 27 * 119304648U }, // y = 36
1201
        { 1219, 16 * 104755300U }, // y = 41
1202
        { 1086, 39 *  93368855U }  // y = 46
1203
    };
1204
    unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1205
    if (x >= 0xFFFF) x -= 0xFFFF;   // max value of x is 8*1877+0xFFFE=0x13AA6,
1206
                                    // so this is effectively a modulo (%)
1207
    y = x - 9 * MULH(477218589, x); // x % 9
1208
    z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1209
                                    // z = x * 49995 / (y * 5 + 6)
1210
    return z % (1000 - block_size);
1211
}
1212

    
1213
/**
1214
 * Parse hardcoded signal for a single block.
1215
 * @note see #synth_block().
1216
 */
1217
static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1218
                                 int block_idx, int size,
1219
                                 const struct frame_type_desc *frame_desc,
1220
                                 float *excitation)
1221
{
1222
    float gain;
1223
    int n, r_idx;
1224

    
1225
    assert(size <= MAX_FRAMESIZE);
1226

    
1227
    /* Set the offset from which we start reading wmavoice_std_codebook */
1228
    if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1229
        r_idx = pRNG(s->frame_cntr, block_idx, size);
1230
        gain  = s->silence_gain;
1231
    } else /* FCB_TYPE_HARDCODED */ {
1232
        r_idx = get_bits(gb, 8);
1233
        gain  = wmavoice_gain_universal[get_bits(gb, 6)];
1234
    }
1235

    
1236
    /* Clear gain prediction parameters */
1237
    memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1238

    
1239
    /* Apply gain to hardcoded codebook and use that as excitation signal */
1240
    for (n = 0; n < size; n++)
1241
        excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1242
}
1243

    
1244
/**
1245
 * Parse FCB/ACB signal for a single block.
1246
 * @note see #synth_block().
1247
 */
1248
static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1249
                                int block_idx, int size,
1250
                                int block_pitch_sh2,
1251
                                const struct frame_type_desc *frame_desc,
1252
                                float *excitation)
1253
{
1254
    static const float gain_coeff[6] = {
1255
        0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1256
    };
1257
    float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1258
    int n, idx, gain_weight;
1259
    AMRFixed fcb;
1260

    
1261
    assert(size <= MAX_FRAMESIZE / 2);
1262
    memset(pulses, 0, sizeof(*pulses) * size);
1263

    
1264
    fcb.pitch_lag      = block_pitch_sh2 >> 2;
1265
    fcb.pitch_fac      = 1.0;
1266
    fcb.no_repeat_mask = 0;
1267
    fcb.n              = 0;
1268

    
1269
    /* For the other frame types, this is where we apply the innovation
1270
     * (fixed) codebook pulses of the speech signal. */
1271
    if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1272
        aw_pulse_set1(s, gb, block_idx, &fcb);
1273
        aw_pulse_set2(s, gb, block_idx, &fcb);
1274
    } else /* FCB_TYPE_EXC_PULSES */ {
1275
        int offset_nbits = 5 - frame_desc->log_n_blocks;
1276

    
1277
        fcb.no_repeat_mask = -1;
1278
        /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1279
         * (instead of double) for a subset of pulses */
1280
        for (n = 0; n < 5; n++) {
1281
            float sign;
1282
            int pos1, pos2;
1283

    
1284
            sign           = get_bits1(gb) ? 1.0 : -1.0;
1285
            pos1           = get_bits(gb, offset_nbits);
1286
            fcb.x[fcb.n]   = n + 5 * pos1;
1287
            fcb.y[fcb.n++] = sign;
1288
            if (n < frame_desc->dbl_pulses) {
1289
                pos2           = get_bits(gb, offset_nbits);
1290
                fcb.x[fcb.n]   = n + 5 * pos2;
1291
                fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1292
            }
1293
        }
1294
    }
1295
    ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1296

    
1297
    /* Calculate gain for adaptive & fixed codebook signal.
1298
     * see ff_amr_set_fixed_gain(). */
1299
    idx = get_bits(gb, 7);
1300
    fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) -
1301
                    5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1302
    acb_gain = wmavoice_gain_codebook_acb[idx];
1303
    pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1304
                        -2.9957322736 /* log(0.05) */,
1305
                         1.6094379124 /* log(5.0)  */);
1306

    
1307
    gain_weight = 8 >> frame_desc->log_n_blocks;
1308
    memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1309
            sizeof(*s->gain_pred_err) * (6 - gain_weight));
1310
    for (n = 0; n < gain_weight; n++)
1311
        s->gain_pred_err[n] = pred_err;
1312

    
1313
    /* Calculation of adaptive codebook */
1314
    if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1315
        int len;
1316
        for (n = 0; n < size; n += len) {
1317
            int next_idx_sh16;
1318
            int abs_idx    = block_idx * size + n;
1319
            int pitch_sh16 = (s->last_pitch_val << 16) +
1320
                             s->pitch_diff_sh16 * abs_idx;
1321
            int pitch      = (pitch_sh16 + 0x6FFF) >> 16;
1322
            int idx_sh16   = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1323
            idx            = idx_sh16 >> 16;
1324
            if (s->pitch_diff_sh16) {
1325
                if (s->pitch_diff_sh16 > 0) {
1326
                    next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1327
                } else
1328
                    next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1329
                len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1330
                              1, size - n);
1331
            } else
1332
                len = size;
1333

    
1334
            ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1335
                                  wmavoice_ipol1_coeffs, 17,
1336
                                  idx, 9, len);
1337
        }
1338
    } else /* ACB_TYPE_HAMMING */ {
1339
        int block_pitch = block_pitch_sh2 >> 2;
1340
        idx             = block_pitch_sh2 & 3;
1341
        if (idx) {
1342
            ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1343
                                  wmavoice_ipol2_coeffs, 4,
1344
                                  idx, 8, size);
1345
        } else
1346
            av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1347
                              sizeof(float) * size);
1348
    }
1349

    
1350
    /* Interpolate ACB/FCB and use as excitation signal */
1351
    ff_weighted_vector_sumf(excitation, excitation, pulses,
1352
                            acb_gain, fcb_gain, size);
1353
}
1354

    
1355
/**
1356
 * Parse data in a single block.
1357
 * @note we assume enough bits are available, caller should check.
1358
 *
1359
 * @param s WMA Voice decoding context private data
1360
 * @param gb bit I/O context
1361
 * @param block_idx index of the to-be-read block
1362
 * @param size amount of samples to be read in this block
1363
 * @param block_pitch_sh2 pitch for this block << 2
1364
 * @param lsps LSPs for (the end of) this frame
1365
 * @param prev_lsps LSPs for the last frame
1366
 * @param frame_desc frame type descriptor
1367
 * @param excitation target memory for the ACB+FCB interpolated signal
1368
 * @param synth target memory for the speech synthesis filter output
1369
 * @return 0 on success, <0 on error.
1370
 */
1371
static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1372
                        int block_idx, int size,
1373
                        int block_pitch_sh2,
1374
                        const double *lsps, const double *prev_lsps,
1375
                        const struct frame_type_desc *frame_desc,
1376
                        float *excitation, float *synth)
1377
{
1378
    double i_lsps[MAX_LSPS];
1379
    float lpcs[MAX_LSPS];
1380
    float fac;
1381
    int n;
1382

    
1383
    if (frame_desc->acb_type == ACB_TYPE_NONE)
1384
        synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1385
    else
1386
        synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1387
                            frame_desc, excitation);
1388

    
1389
    /* convert interpolated LSPs to LPCs */
1390
    fac = (block_idx + 0.5) / frame_desc->n_blocks;
1391
    for (n = 0; n < s->lsps; n++) // LSF -> LSP
1392
        i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1393
    ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1394

    
1395
    /* Speech synthesis */
1396
    ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1397
}
1398

    
1399
/**
1400
 * Synthesize output samples for a single frame.
1401
 * @note we assume enough bits are available, caller should check.
1402
 *
1403
 * @param ctx WMA Voice decoder context
1404
 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1405
 * @param frame_idx Frame number within superframe [0-2]
1406
 * @param samples pointer to output sample buffer, has space for at least 160
1407
 *                samples
1408
 * @param lsps LSP array
1409
 * @param prev_lsps array of previous frame's LSPs
1410
 * @param excitation target buffer for excitation signal
1411
 * @param synth target buffer for synthesized speech data
1412
 * @return 0 on success, <0 on error.
1413
 */
1414
static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1415
                       float *samples,
1416
                       const double *lsps, const double *prev_lsps,
1417
                       float *excitation, float *synth)
1418
{
1419
    WMAVoiceContext *s = ctx->priv_data;
1420
    int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
1421
    int pitch[MAX_BLOCKS], last_block_pitch;
1422

    
1423
    /* Parse frame type ("frame header"), see frame_descs */
1424
    int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)],
1425
        block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1426

    
1427
    if (bd_idx < 0) {
1428
        av_log(ctx, AV_LOG_ERROR,
1429
               "Invalid frame type VLC code, skipping\n");
1430
        return -1;
1431
    }
1432

    
1433
    /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1434
    if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1435
        /* Pitch is provided per frame, which is interpreted as the pitch of
1436
         * the last sample of the last block of this frame. We can interpolate
1437
         * the pitch of other blocks (and even pitch-per-sample) by gradually
1438
         * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1439
        n_blocks_x2      = frame_descs[bd_idx].n_blocks << 1;
1440
        log_n_blocks_x2  = frame_descs[bd_idx].log_n_blocks + 1;
1441
        cur_pitch_val    = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1442
        cur_pitch_val    = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1443
        if (s->last_acb_type == ACB_TYPE_NONE ||
1444
            20 * abs(cur_pitch_val - s->last_pitch_val) >
1445
                (cur_pitch_val + s->last_pitch_val))
1446
            s->last_pitch_val = cur_pitch_val;
1447

    
1448
        /* pitch per block */
1449
        for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1450
            int fac = n * 2 + 1;
1451

    
1452
            pitch[n] = (MUL16(fac,                 cur_pitch_val) +
1453
                        MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1454
                        frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1455
        }
1456

    
1457
        /* "pitch-diff-per-sample" for calculation of pitch per sample */
1458
        s->pitch_diff_sh16 =
1459
            ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1460
    }
1461

    
1462
    /* Global gain (if silence) and pitch-adaptive window coordinates */
1463
    switch (frame_descs[bd_idx].fcb_type) {
1464
    case FCB_TYPE_SILENCE:
1465
        s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1466
        break;
1467
    case FCB_TYPE_AW_PULSES:
1468
        aw_parse_coords(s, gb, pitch);
1469
        break;
1470
    }
1471

    
1472
    for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1473
        int bl_pitch_sh2;
1474

    
1475
        /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1476
        switch (frame_descs[bd_idx].acb_type) {
1477
        case ACB_TYPE_HAMMING: {
1478
            /* Pitch is given per block. Per-block pitches are encoded as an
1479
             * absolute value for the first block, and then delta values
1480
             * relative to this value) for all subsequent blocks. The scale of
1481
             * this pitch value is semi-logaritmic compared to its use in the
1482
             * decoder, so we convert it to normal scale also. */
1483
            int block_pitch,
1484
                t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1485
                t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1486
                t3 =  s->block_conv_table[3] - s->block_conv_table[2] + 1;
1487

    
1488
            if (n == 0) {
1489
                block_pitch = get_bits(gb, s->block_pitch_nbits);
1490
            } else
1491
                block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1492
                                 get_bits(gb, s->block_delta_pitch_nbits);
1493
            /* Convert last_ so that any next delta is within _range */
1494
            last_block_pitch = av_clip(block_pitch,
1495
                                       s->block_delta_pitch_hrange,
1496
                                       s->block_pitch_range -
1497
                                           s->block_delta_pitch_hrange);
1498

    
1499
            /* Convert semi-log-style scale back to normal scale */
1500
            if (block_pitch < t1) {
1501
                bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1502
            } else {
1503
                block_pitch -= t1;
1504
                if (block_pitch < t2) {
1505
                    bl_pitch_sh2 =
1506
                        (s->block_conv_table[1] << 2) + (block_pitch << 1);
1507
                } else {
1508
                    block_pitch -= t2;
1509
                    if (block_pitch < t3) {
1510
                        bl_pitch_sh2 =
1511
                            (s->block_conv_table[2] + block_pitch) << 2;
1512
                    } else
1513
                        bl_pitch_sh2 = s->block_conv_table[3] << 2;
1514
                }
1515
            }
1516
            pitch[n] = bl_pitch_sh2 >> 2;
1517
            break;
1518
        }
1519

    
1520
        case ACB_TYPE_ASYMMETRIC: {
1521
            bl_pitch_sh2 = pitch[n] << 2;
1522
            break;
1523
        }
1524

    
1525
        default: // ACB_TYPE_NONE has no pitch
1526
            bl_pitch_sh2 = 0;
1527
            break;
1528
        }
1529

    
1530
        synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1531
                    lsps, prev_lsps, &frame_descs[bd_idx],
1532
                    &excitation[n * block_nsamples],
1533
                    &synth[n * block_nsamples]);
1534
    }
1535

    
1536
    /* Averaging projection filter, if applicable. Else, just copy samples
1537
     * from synthesis buffer */
1538
    if (s->do_apf) {
1539
        double i_lsps[MAX_LSPS];
1540
        float lpcs[MAX_LSPS];
1541

    
1542
        for (n = 0; n < s->lsps; n++) // LSF -> LSP
1543
            i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1544
        ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1545
        postfilter(s, synth, samples, 80, lpcs,
1546
                   &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1547
                   frame_descs[bd_idx].fcb_type, pitch[0]);
1548

    
1549
        for (n = 0; n < s->lsps; n++) // LSF -> LSP
1550
            i_lsps[n] = cos(lsps[n]);
1551
        ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1552
        postfilter(s, &synth[80], &samples[80], 80, lpcs,
1553
                   &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1554
                   frame_descs[bd_idx].fcb_type, pitch[0]);
1555
    } else
1556
        memcpy(samples, synth, 160 * sizeof(synth[0]));
1557

    
1558
    /* Cache values for next frame */
1559
    s->frame_cntr++;
1560
    if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1561
    s->last_acb_type = frame_descs[bd_idx].acb_type;
1562
    switch (frame_descs[bd_idx].acb_type) {
1563
    case ACB_TYPE_NONE:
1564
        s->last_pitch_val = 0;
1565
        break;
1566
    case ACB_TYPE_ASYMMETRIC:
1567
        s->last_pitch_val = cur_pitch_val;
1568
        break;
1569
    case ACB_TYPE_HAMMING:
1570
        s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1571
        break;
1572
    }
1573

    
1574
    return 0;
1575
}
1576

    
1577
/**
1578
 * Ensure minimum value for first item, maximum value for last value,
1579
 * proper spacing between each value and proper ordering.
1580
 *
1581
 * @param lsps array of LSPs
1582
 * @param num size of LSP array
1583
 *
1584
 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1585
 *       useful to put in a generic location later on. Parts are also
1586
 *       present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1587
 *       which is in float.
1588
 */
1589
static void stabilize_lsps(double *lsps, int num)
1590
{
1591
    int n, m, l;
1592

    
1593
    /* set minimum value for first, maximum value for last and minimum
1594
     * spacing between LSF values.
1595
     * Very similar to ff_set_min_dist_lsf(), but in double. */
1596
    lsps[0]       = FFMAX(lsps[0],       0.0015 * M_PI);
1597
    for (n = 1; n < num; n++)
1598
        lsps[n]   = FFMAX(lsps[n],       lsps[n - 1] + 0.0125 * M_PI);
1599
    lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1600

    
1601
    /* reorder (looks like one-time / non-recursed bubblesort).
1602
     * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1603
    for (n = 1; n < num; n++) {
1604
        if (lsps[n] < lsps[n - 1]) {
1605
            for (m = 1; m < num; m++) {
1606
                double tmp = lsps[m];
1607
                for (l = m - 1; l >= 0; l--) {
1608
                    if (lsps[l] <= tmp) break;
1609
                    lsps[l + 1] = lsps[l];
1610
                }
1611
                lsps[l + 1] = tmp;
1612
            }
1613
            break;
1614
        }
1615
    }
1616
}
1617

    
1618
/**
1619
 * Test if there's enough bits to read 1 superframe.
1620
 *
1621
 * @param orig_gb bit I/O context used for reading. This function
1622
 *                does not modify the state of the bitreader; it
1623
 *                only uses it to copy the current stream position
1624
 * @param s WMA Voice decoding context private data
1625
 * @return -1 if unsupported, 1 on not enough bits or 0 if OK.
1626
 */
1627
static int check_bits_for_superframe(GetBitContext *orig_gb,
1628
                                     WMAVoiceContext *s)
1629
{
1630
    GetBitContext s_gb, *gb = &s_gb;
1631
    int n, need_bits, bd_idx;
1632
    const struct frame_type_desc *frame_desc;
1633

    
1634
    /* initialize a copy */
1635
    init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
1636
    skip_bits_long(gb, get_bits_count(orig_gb));
1637
    assert(get_bits_left(gb) == get_bits_left(orig_gb));
1638

    
1639
    /* superframe header */
1640
    if (get_bits_left(gb) < 14)
1641
        return 1;
1642
    if (!get_bits1(gb))
1643
        return -1;                        // WMAPro-in-WMAVoice superframe
1644
    if (get_bits1(gb)) skip_bits(gb, 12); // number of  samples in superframe
1645
    if (s->has_residual_lsps) {           // residual LSPs (for all frames)
1646
        if (get_bits_left(gb) < s->sframe_lsp_bitsize)
1647
            return 1;
1648
        skip_bits_long(gb, s->sframe_lsp_bitsize);
1649
    }
1650

    
1651
    /* frames */
1652
    for (n = 0; n < MAX_FRAMES; n++) {
1653
        int aw_idx_is_ext = 0;
1654

    
1655
        if (!s->has_residual_lsps) {     // independent LSPs (per-frame)
1656
           if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
1657
           skip_bits_long(gb, s->frame_lsp_bitsize);
1658
        }
1659
        bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
1660
        if (bd_idx < 0)
1661
            return -1;                   // invalid frame type VLC code
1662
        frame_desc = &frame_descs[bd_idx];
1663
        if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1664
            if (get_bits_left(gb) < s->pitch_nbits)
1665
                return 1;
1666
            skip_bits_long(gb, s->pitch_nbits);
1667
        }
1668
        if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1669
            skip_bits(gb, 8);
1670
        } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1671
            int tmp = get_bits(gb, 6);
1672
            if (tmp >= 0x36) {
1673
                skip_bits(gb, 2);
1674
                aw_idx_is_ext = 1;
1675
            }
1676
        }
1677

    
1678
        /* blocks */
1679
        if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
1680
            need_bits = s->block_pitch_nbits +
1681
                (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
1682
        } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1683
            need_bits = 2 * !aw_idx_is_ext;
1684
        } else
1685
            need_bits = 0;
1686
        need_bits += frame_desc->frame_size;
1687
        if (get_bits_left(gb) < need_bits)
1688
            return 1;
1689
        skip_bits_long(gb, need_bits);
1690
    }
1691

    
1692
    return 0;
1693
}
1694

    
1695
/**
1696
 * Synthesize output samples for a single superframe. If we have any data
1697
 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1698
 * in s->gb.
1699
 *
1700
 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1701
 * to give a total of 480 samples per frame. See #synth_frame() for frame
1702
 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1703
 * (if these are globally specified for all frames (residually); they can
1704
 * also be specified individually per-frame. See the s->has_residual_lsps
1705
 * option), and can specify the number of samples encoded in this superframe
1706
 * (if less than 480), usually used to prevent blanks at track boundaries.
1707
 *
1708
 * @param ctx WMA Voice decoder context
1709
 * @param samples pointer to output buffer for voice samples
1710
 * @param data_size pointer containing the size of #samples on input, and the
1711
 *                  amount of #samples filled on output
1712
 * @return 0 on success, <0 on error or 1 if there was not enough data to
1713
 *         fully parse the superframe
1714
 */
1715
static int synth_superframe(AVCodecContext *ctx,
1716
                            float *samples, int *data_size)
1717
{
1718
    WMAVoiceContext *s = ctx->priv_data;
1719
    GetBitContext *gb = &s->gb, s_gb;
1720
    int n, res, n_samples = 480;
1721
    double lsps[MAX_FRAMES][MAX_LSPS];
1722
    const double *mean_lsf = s->lsps == 16 ?
1723
        wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1724
    float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1725
    float synth[MAX_LSPS + MAX_SFRAMESIZE];
1726

    
1727
    memcpy(synth,      s->synth_history,
1728
           s->lsps             * sizeof(*synth));
1729
    memcpy(excitation, s->excitation_history,
1730
           s->history_nsamples * sizeof(*excitation));
1731

    
1732
    if (s->sframe_cache_size > 0) {
1733
        gb = &s_gb;
1734
        init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1735
        s->sframe_cache_size = 0;
1736
    }
1737

    
1738
    if ((res = check_bits_for_superframe(gb, s)) == 1) return 1;
1739

    
1740
    /* First bit is speech/music bit, it differentiates between WMAVoice
1741
     * speech samples (the actual codec) and WMAVoice music samples, which
1742
     * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1743
     * the wild yet. */
1744
    if (!get_bits1(gb)) {
1745
        av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1);
1746
        return -1;
1747
    }
1748

    
1749
    /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1750
    if (get_bits1(gb)) {
1751
        if ((n_samples = get_bits(gb, 12)) > 480) {
1752
            av_log(ctx, AV_LOG_ERROR,
1753
                   "Superframe encodes >480 samples (%d), not allowed\n",
1754
                   n_samples);
1755
            return -1;
1756
        }
1757
    }
1758
    /* Parse LSPs, if global for the superframe (can also be per-frame). */
1759
    if (s->has_residual_lsps) {
1760
        double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1761

    
1762
        for (n = 0; n < s->lsps; n++)
1763
            prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1764

    
1765
        if (s->lsps == 10) {
1766
            dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1767
        } else /* s->lsps == 16 */
1768
            dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1769

    
1770
        for (n = 0; n < s->lsps; n++) {
1771
            lsps[0][n]  = mean_lsf[n] + (a1[n]           - a2[n * 2]);
1772
            lsps[1][n]  = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1773
            lsps[2][n] += mean_lsf[n];
1774
        }
1775
        for (n = 0; n < 3; n++)
1776
            stabilize_lsps(lsps[n], s->lsps);
1777
    }
1778

    
1779
    /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
1780
    for (n = 0; n < 3; n++) {
1781
        if (!s->has_residual_lsps) {
1782
            int m;
1783

    
1784
            if (s->lsps == 10) {
1785
                dequant_lsp10i(gb, lsps[n]);
1786
            } else /* s->lsps == 16 */
1787
                dequant_lsp16i(gb, lsps[n]);
1788

    
1789
            for (m = 0; m < s->lsps; m++)
1790
                lsps[n][m] += mean_lsf[m];
1791
            stabilize_lsps(lsps[n], s->lsps);
1792
        }
1793

    
1794
        if ((res = synth_frame(ctx, gb, n,
1795
                               &samples[n * MAX_FRAMESIZE],
1796
                               lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1797
                               &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1798
                               &synth[s->lsps + n * MAX_FRAMESIZE])))
1799
            return res;
1800
    }
1801

    
1802
    /* Statistics? FIXME - we don't check for length, a slight overrun
1803
     * will be caught by internal buffer padding, and anything else
1804
     * will be skipped, not read. */
1805
    if (get_bits1(gb)) {
1806
        res = get_bits(gb, 4);
1807
        skip_bits(gb, 10 * (res + 1));
1808
    }
1809

    
1810
    /* Specify nr. of output samples */
1811
    *data_size = n_samples * sizeof(float);
1812

    
1813
    /* Update history */
1814
    memcpy(s->prev_lsps,           lsps[2],
1815
           s->lsps             * sizeof(*s->prev_lsps));
1816
    memcpy(s->synth_history,      &synth[MAX_SFRAMESIZE],
1817
           s->lsps             * sizeof(*synth));
1818
    memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1819
           s->history_nsamples * sizeof(*excitation));
1820
    if (s->do_apf)
1821
        memmove(s->zero_exc_pf,       &s->zero_exc_pf[MAX_SFRAMESIZE],
1822
                s->history_nsamples * sizeof(*s->zero_exc_pf));
1823

    
1824
    return 0;
1825
}
1826

    
1827
/**
1828
 * Parse the packet header at the start of each packet (input data to this
1829
 * decoder).
1830
 *
1831
 * @param s WMA Voice decoding context private data
1832
 * @return 1 if not enough bits were available, or 0 on success.
1833
 */
1834
static int parse_packet_header(WMAVoiceContext *s)
1835
{
1836
    GetBitContext *gb = &s->gb;
1837
    unsigned int res;
1838

    
1839
    if (get_bits_left(gb) < 11)
1840
        return 1;
1841
    skip_bits(gb, 4);          // packet sequence number
1842
    s->has_residual_lsps = get_bits1(gb);
1843
    do {
1844
        res = get_bits(gb, 6); // number of superframes per packet
1845
                               // (minus first one if there is spillover)
1846
        if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
1847
            return 1;
1848
    } while (res == 0x3F);
1849
    s->spillover_nbits   = get_bits(gb, s->spillover_bitsize);
1850

    
1851
    return 0;
1852
}
1853

    
1854
/**
1855
 * Copy (unaligned) bits from gb/data/size to pb.
1856
 *
1857
 * @param pb target buffer to copy bits into
1858
 * @param data source buffer to copy bits from
1859
 * @param size size of the source data, in bytes
1860
 * @param gb bit I/O context specifying the current position in the source.
1861
 *           data. This function might use this to align the bit position to
1862
 *           a whole-byte boundary before calling #ff_copy_bits() on aligned
1863
 *           source data
1864
 * @param nbits the amount of bits to copy from source to target
1865
 *
1866
 * @note after calling this function, the current position in the input bit
1867
 *       I/O context is undefined.
1868
 */
1869
static void copy_bits(PutBitContext *pb,
1870
                      const uint8_t *data, int size,
1871
                      GetBitContext *gb, int nbits)
1872
{
1873
    int rmn_bytes, rmn_bits;
1874

    
1875
    rmn_bits = rmn_bytes = get_bits_left(gb);
1876
    if (rmn_bits < nbits)
1877
        return;
1878
    rmn_bits &= 7; rmn_bytes >>= 3;
1879
    if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1880
        put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1881
    ff_copy_bits(pb, data + size - rmn_bytes,
1882
                 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1883
}
1884

    
1885
/**
1886
 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1887
 * and we expect that the demuxer / application provides it to us as such
1888
 * (else you'll probably get garbage as output). Every packet has a size of
1889
 * ctx->block_align bytes, starts with a packet header (see
1890
 * #parse_packet_header()), and then a series of superframes. Superframe
1891
 * boundaries may exceed packets, i.e. superframes can split data over
1892
 * multiple (two) packets.
1893
 *
1894
 * For more information about frames, see #synth_superframe().
1895
 */
1896
static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1897
                                  int *data_size, AVPacket *avpkt)
1898
{
1899
    WMAVoiceContext *s = ctx->priv_data;
1900
    GetBitContext *gb = &s->gb;
1901
    int size, res, pos;
1902

    
1903
    if (*data_size < 480 * sizeof(float)) {
1904
        av_log(ctx, AV_LOG_ERROR,
1905
               "Output buffer too small (%d given - %lu needed)\n",
1906
               *data_size, 480 * sizeof(float));
1907
        return -1;
1908
    }
1909
    *data_size = 0;
1910

    
1911
    /* Packets are sometimes a multiple of ctx->block_align, with a packet
1912
     * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1913
     * feeds us ASF packets, which may concatenate multiple "codec" packets
1914
     * in a single "muxer" packet, so we artificially emulate that by
1915
     * capping the packet size at ctx->block_align. */
1916
    for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1917
    if (!size)
1918
        return 0;
1919
    init_get_bits(&s->gb, avpkt->data, size << 3);
1920

    
1921
    /* size == ctx->block_align is used to indicate whether we are dealing with
1922
     * a new packet or a packet of which we already read the packet header
1923
     * previously. */
1924
    if (size == ctx->block_align) { // new packet header
1925
        if ((res = parse_packet_header(s)) < 0)
1926
            return res;
1927

    
1928
        /* If the packet header specifies a s->spillover_nbits, then we want
1929
         * to push out all data of the previous packet (+ spillover) before
1930
         * continuing to parse new superframes in the current packet. */
1931
        if (s->spillover_nbits > 0) {
1932
            if (s->sframe_cache_size > 0) {
1933
                int cnt = get_bits_count(gb);
1934
                copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1935
                flush_put_bits(&s->pb);
1936
                s->sframe_cache_size += s->spillover_nbits;
1937
                if ((res = synth_superframe(ctx, data, data_size)) == 0 &&
1938
                    *data_size > 0) {
1939
                    cnt += s->spillover_nbits;
1940
                    s->skip_bits_next = cnt & 7;
1941
                    return cnt >> 3;
1942
                } else
1943
                    skip_bits_long (gb, s->spillover_nbits - cnt +
1944
                                    get_bits_count(gb)); // resync
1945
            } else
1946
                skip_bits_long(gb, s->spillover_nbits);  // resync
1947
        }
1948
    } else if (s->skip_bits_next)
1949
        skip_bits(gb, s->skip_bits_next);
1950

    
1951
    /* Try parsing superframes in current packet */
1952
    s->sframe_cache_size = 0;
1953
    s->skip_bits_next = 0;
1954
    pos = get_bits_left(gb);
1955
    if ((res = synth_superframe(ctx, data, data_size)) < 0) {
1956
        return res;
1957
    } else if (*data_size > 0) {
1958
        int cnt = get_bits_count(gb);
1959
        s->skip_bits_next = cnt & 7;
1960
        return cnt >> 3;
1961
    } else if ((s->sframe_cache_size = pos) > 0) {
1962
        /* rewind bit reader to start of last (incomplete) superframe... */
1963
        init_get_bits(gb, avpkt->data, size << 3);
1964
        skip_bits_long(gb, (size << 3) - pos);
1965
        assert(get_bits_left(gb) == pos);
1966

    
1967
        /* ...and cache it for spillover in next packet */
1968
        init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
1969
        copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
1970
        // FIXME bad - just copy bytes as whole and add use the
1971
        // skip_bits_next field
1972
    }
1973

    
1974
    return size;
1975
}
1976

    
1977
static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
1978
{
1979
    WMAVoiceContext *s = ctx->priv_data;
1980

    
1981
    if (s->do_apf) {
1982
        ff_rdft_end(&s->rdft);
1983
        ff_rdft_end(&s->irdft);
1984
        ff_dct_end(&s->dct);
1985
        ff_dct_end(&s->dst);
1986
    }
1987

    
1988
    return 0;
1989
}
1990

    
1991
static av_cold void wmavoice_flush(AVCodecContext *ctx)
1992
{
1993
    WMAVoiceContext *s = ctx->priv_data;
1994
    int n;
1995

    
1996
    s->postfilter_agc    = 0;
1997
    s->sframe_cache_size = 0;
1998
    s->skip_bits_next    = 0;
1999
    for (n = 0; n < s->lsps; n++)
2000
        s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2001
    memset(s->excitation_history, 0,
2002
           sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2003
    memset(s->synth_history,      0,
2004
           sizeof(*s->synth_history)      * MAX_LSPS);
2005
    memset(s->gain_pred_err,      0,
2006
           sizeof(s->gain_pred_err));
2007

    
2008
    if (s->do_apf) {
2009
        memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2010
               sizeof(*s->synth_filter_out_buf) * s->lsps);
2011
        memset(s->dcf_mem,              0,
2012
               sizeof(*s->dcf_mem)              * 2);
2013
        memset(s->zero_exc_pf,          0,
2014
               sizeof(*s->zero_exc_pf)          * s->history_nsamples);
2015
        memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2016
    }
2017
}
2018

    
2019
AVCodec wmavoice_decoder = {
2020
    "wmavoice",
2021
    AVMEDIA_TYPE_AUDIO,
2022
    CODEC_ID_WMAVOICE,
2023
    sizeof(WMAVoiceContext),
2024
    wmavoice_decode_init,
2025
    NULL,
2026
    wmavoice_decode_end,
2027
    wmavoice_decode_packet,
2028
    CODEC_CAP_SUBFRAMES,
2029
    .flush     = wmavoice_flush,
2030
    .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
2031
};