ffmpeg / libavcodec / wmavoice.c @ 869303be
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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 Libav.

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*

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* Libav 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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* Libav 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 Libav; if not, write to the Free Software

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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 021101301 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 "dct.h" 
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#include "rdft.h" 
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#include "sinewin.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 
55  
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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;

60  
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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 perframe pitch, which 
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///< we interpolate to get a persample 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 ///< Perblock 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 perframe (low) gain values

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FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with perblock 
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///< gain values

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FCB_TYPE_AW_PULSES = 2, ///< Pitchadaptive window (AW) pulse signals, 
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///< used in particular for lowbitrate 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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///< [011]

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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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///< perframe (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 +this1)

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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 resynchronization 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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/**

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* @}

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* @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.

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* @{

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*/

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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 [02] 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 AWpulse 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 AWpulses 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 AWpulses, 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 pitchadaptive 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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* @}

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* @defgroup post_filter Postfilter values

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* Variables used for postfilter implementation, mostly history for

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* smoothing and so on, and context variables for FFT/iFFT.

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* @{

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*/

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RDFTContext rdft, irdft; ///< contexts for FFTcalculation 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]; ///< 8bit cosine/sine windows over [pi,pi] 
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///< range

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float postfilter_agc; ///< gain control memory, used in 
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///< #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];

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///< zero filter output (i.e. excitation)

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///< by postfilter

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float denoise_filter_cache[MAX_FRAMESIZE];

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int denoise_filter_cache_size; ///< samples in #denoise_filter_cache 
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DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80]; 
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///< aligned buffer for LPC tilting

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DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; 
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///< aligned buffer for denoise coefficients

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DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; 
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///< aligned buffer for postfilter speech

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///< synthesis

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/**

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* @}

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*/

288 
} WMAVoiceContext; 
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/**

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* Set up the variable bit mode (VBM) tree from container extradata.

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* @param gb bit I/O context.

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* The bit context (s>gb) should be loaded with byte 2346 of the

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* container extradata (i.e. the ones containing the VBM tree).

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* @param vbm_tree pointer to array to which the decoded VBM tree will be

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* written.

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* @return 0 on success, <0 on error.

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*/

299 
static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) 
300 
{ 
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static const uint8_t bits[] = { 
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2, 2, 2, 4, 4, 4, 
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6, 6, 6, 8, 8, 8, 
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10, 10, 10, 12, 12, 12, 
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14, 14, 14, 14 
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}; 
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static const uint16_t codes[] = { 
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0x0000, 0x0001, 0x0002, // 00/01/10 
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0x000c, 0x000d, 0x000e, // 11+00/01/10 
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0x003c, 0x003d, 0x003e, // 1111+00/01/10 
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0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 
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0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 
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0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 
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0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx 
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}; 
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int cntr[8], n, res; 
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318 
memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); 
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memset(cntr, 0, sizeof(cntr)); 
320 
for (n = 0; n < 17; n++) { 
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res = get_bits(gb, 3);

322 
if (cntr[res] > 3) // should be >= 3 + (res == 7)) 
323 
return 1; 
324 
vbm_tree[res * 3 + cntr[res]++] = n;

325 
} 
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INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),

327 
bits, 1, 1, codes, 2, 2, 132); 
328 
return 0; 
329 
} 
330  
331 
/**

332 
* Set up decoder with parameters from demuxer (extradata etc.).

333 
*/

334 
static av_cold int wmavoice_decode_init(AVCodecContext *ctx) 
335 
{ 
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int n, flags, pitch_range, lsp16_flag;

337 
WMAVoiceContext *s = ctx>priv_data; 
338  
339 
/**

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* Extradata layout:

341 
*  byte 018: WMAProinWMAVoice extradata (see wmaprodec.c),

342 
*  byte 1922: flags field (annoyingly in LE; see below for known

343 
* values),

344 
*  byte 2346: variable bitmode tree (really just 17 * 3 bits,

345 
* rest is 0).

346 
*/

347 
if (ctx>extradata_size != 46) { 
348 
av_log(ctx, AV_LOG_ERROR, 
349 
"Invalid extradata size %d (should be 46)\n",

350 
ctx>extradata_size); 
351 
return 1; 
352 
} 
353 
flags = AV_RL32(ctx>extradata + 18);

354 
s>spillover_bitsize = 3 + av_ceil_log2(ctx>block_align);

355 
s>do_apf = flags & 0x1;

356 
if (s>do_apf) {

357 
ff_rdft_init(&s>rdft, 7, DFT_R2C);

358 
ff_rdft_init(&s>irdft, 7, IDFT_C2R);

359 
ff_dct_init(&s>dct, 6, DCT_I);

360 
ff_dct_init(&s>dst, 6, DST_I);

361  
362 
ff_sine_window_init(s>cos, 256);

363 
memcpy(&s>sin[255], s>cos, 256 * sizeof(s>cos[0])); 
364 
for (n = 0; n < 255; n++) { 
365 
s>sin[n] = s>sin[510  n];

366 
s>cos[510  n] = s>cos[n];

367 
} 
368 
} 
369 
s>denoise_strength = (flags >> 2) & 0xF; 
370 
if (s>denoise_strength >= 12) { 
371 
av_log(ctx, AV_LOG_ERROR, 
372 
"Invalid denoise filter strength %d (max=11)\n",

373 
s>denoise_strength); 
374 
return 1; 
375 
} 
376 
s>denoise_tilt_corr = !!(flags & 0x40);

377 
s>dc_level = (flags >> 7) & 0xF; 
378 
s>lsp_q_mode = !!(flags & 0x2000);

379 
s>lsp_def_mode = !!(flags & 0x4000);

380 
lsp16_flag = flags & 0x1000;

381 
if (lsp16_flag) {

382 
s>lsps = 16;

383 
s>frame_lsp_bitsize = 34;

384 
s>sframe_lsp_bitsize = 60;

385 
} else {

386 
s>lsps = 10;

387 
s>frame_lsp_bitsize = 24;

388 
s>sframe_lsp_bitsize = 48;

389 
} 
390 
for (n = 0; n < s>lsps; n++) 
391 
s>prev_lsps[n] = M_PI * (n + 1.0) / (s>lsps + 1.0); 
392  
393 
init_get_bits(&s>gb, ctx>extradata + 22, (ctx>extradata_size  22) << 3); 
394 
if (decode_vbmtree(&s>gb, s>vbm_tree) < 0) { 
395 
av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");

396 
return 1; 
397 
} 
398  
399 
s>min_pitch_val = ((ctx>sample_rate << 8) / 400 + 50) >> 8; 
400 
s>max_pitch_val = ((ctx>sample_rate << 8) * 37 / 2000 + 50) >> 8; 
401 
pitch_range = s>max_pitch_val  s>min_pitch_val; 
402 
s>pitch_nbits = av_ceil_log2(pitch_range); 
403 
s>last_pitch_val = 40;

404 
s>last_acb_type = ACB_TYPE_NONE; 
405 
s>history_nsamples = s>max_pitch_val + 8;

406  
407 
if (s>min_pitch_val < 1  s>history_nsamples > MAX_SIGNAL_HISTORY) { 
408 
int min_sr = ((((1 << 8)  50) * 400) + 0xFF) >> 8, 
409 
max_sr = ((((MAX_SIGNAL_HISTORY  8) << 8) + 205) * 2000 / 37) >> 8; 
410  
411 
av_log(ctx, AV_LOG_ERROR, 
412 
"Unsupported samplerate %d (min=%d, max=%d)\n",

413 
ctx>sample_rate, min_sr, max_sr); // 32222097 Hz

414  
415 
return 1; 
416 
} 
417  
418 
s>block_conv_table[0] = s>min_pitch_val;

419 
s>block_conv_table[1] = (pitch_range * 25) >> 6; 
420 
s>block_conv_table[2] = (pitch_range * 44) >> 6; 
421 
s>block_conv_table[3] = s>max_pitch_val  1; 
422 
s>block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; 
423 
s>block_delta_pitch_nbits = 1 + av_ceil_log2(s>block_delta_pitch_hrange);

424 
s>block_pitch_range = s>block_conv_table[2] +

425 
s>block_conv_table[3] + 1 + 
426 
2 * (s>block_conv_table[1]  2 * s>min_pitch_val); 
427 
s>block_pitch_nbits = av_ceil_log2(s>block_pitch_range); 
428  
429 
ctx>sample_fmt = AV_SAMPLE_FMT_FLT; 
430  
431 
return 0; 
432 
} 
433  
434 
/**

435 
* @defgroup postfilter Postfilter functions

436 
* Postfilter functions (gain control, wiener denoise filter, DC filter,

437 
* kalman smoothening, plus surrounding code to wrap it)

438 
* @{

439 
*/

440 
/**

441 
* Adaptive gain control (as used in postfilter).

442 
*

443 
* Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except

444 
* that the energy here is calculated using sum(abs(...)), whereas the

445 
* other codecs (e.g. AMRNB, SIPRO) use sqrt(dotproduct(...)).

446 
*

447 
* @param out output buffer for filtered samples

448 
* @param in input buffer containing the samples as they are after the

449 
* postfilter steps so far

450 
* @param speech_synth input buffer containing speech synth before postfilter

451 
* @param size input buffer size

452 
* @param alpha exponential filter factor

453 
* @param gain_mem pointer to filter memory (single float)

454 
*/

455 
static void adaptive_gain_control(float *out, const float *in, 
456 
const float *speech_synth, 
457 
int size, float alpha, float *gain_mem) 
458 
{ 
459 
int i;

460 
float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; 
461 
float mem = *gain_mem;

462  
463 
for (i = 0; i < size; i++) { 
464 
speech_energy += fabsf(speech_synth[i]); 
465 
postfilter_energy += fabsf(in[i]); 
466 
} 
467 
gain_scale_factor = (1.0  alpha) * speech_energy / postfilter_energy; 
468  
469 
for (i = 0; i < size; i++) { 
470 
mem = alpha * mem + gain_scale_factor; 
471 
out[i] = in[i] * mem; 
472 
} 
473  
474 
*gain_mem = mem; 
475 
} 
476  
477 
/**

478 
* Kalman smoothing function.

479 
*

480 
* This function looks back pitch +/ 3 samples back into history to find

481 
* the best fitting curve (that one giving the optimal gain of the two

482 
* signals, i.e. the highest dot product between the two), and then

483 
* uses that signal history to smoothen the output of the speech synthesis

484 
* filter.

485 
*

486 
* @param s WMA Voice decoding context

487 
* @param pitch pitch of the speech signal

488 
* @param in input speech signal

489 
* @param out output pointer for smoothened signal

490 
* @param size input/output buffer size

491 
*

492 
* @returns 1 if no smoothening took place, e.g. because no optimal

493 
* fit could be found, or 0 on success.

494 
*/

495 
static int kalman_smoothen(WMAVoiceContext *s, int pitch, 
496 
const float *in, float *out, int size) 
497 
{ 
498 
int n;

499 
float optimal_gain = 0, dot; 
500 
const float *ptr = &in[FFMAX(s>min_pitch_val, pitch  3)], 
501 
*end = &in[FFMIN(s>max_pitch_val, pitch + 3)],

502 
*best_hist_ptr; 
503  
504 
/* find best fitting point in history */

505 
do {

506 
dot = ff_dot_productf(in, ptr, size); 
507 
if (dot > optimal_gain) {

508 
optimal_gain = dot; 
509 
best_hist_ptr = ptr; 
510 
} 
511 
} while (ptr >= end);

512  
513 
if (optimal_gain <= 0) 
514 
return 1; 
515 
dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size); 
516 
if (dot <= 0) // would be 1.0 
517 
return 1; 
518  
519 
if (optimal_gain <= dot) {

520 
dot = dot / (dot + 0.6 * optimal_gain); // 0.6251.000 
521 
} else

522 
dot = 0.625; 
523  
524 
/* actual smoothing */

525 
for (n = 0; n < size; n++) 
526 
out[n] = best_hist_ptr[n] + dot * (in[n]  best_hist_ptr[n]); 
527  
528 
return 0; 
529 
} 
530  
531 
/**

532 
* Get the tilt factor of a formant filter from its transfer function

533 
* @see #tilt_factor() in amrnbdec.c, which does essentially the same,

534 
* but somehow (??) it does a speech synthesis filter in the

535 
* middle, which is missing here

536 
*

537 
* @param lpcs LPC coefficients

538 
* @param n_lpcs Size of LPC buffer

539 
* @returns the tilt factor

540 
*/

541 
static float tilt_factor(const float *lpcs, int n_lpcs) 
542 
{ 
543 
float rh0, rh1;

544  
545 
rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs); 
546 
rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs  1); 
547  
548 
return rh1 / rh0;

549 
} 
550  
551 
/**

552 
* Derive denoise filter coefficients (in real domain) from the LPCs.

553 
*/

554 
static void calc_input_response(WMAVoiceContext *s, float *lpcs, 
555 
int fcb_type, float *coeffs, int remainder) 
556 
{ 
557 
float last_coeff, min = 15.0, max = 15.0; 
558 
float irange, angle_mul, gain_mul, range, sq;

559 
int n, idx;

560  
561 
/* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */

562 
s>rdft.rdft_calc(&s>rdft, lpcs); 
563 
#define log_range(var, assign) do { \ 
564 
float tmp = log10f(assign); var = tmp; \

565 
max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ 
566 
} while (0) 
567 
log_range(last_coeff, lpcs[1] * lpcs[1]); 
568 
for (n = 1; n < 64; n++) 
569 
log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + 
570 
lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); 
571 
log_range(lpcs[0], lpcs[0] * lpcs[0]); 
572 
#undef log_range

573 
range = max  min; 
574 
lpcs[64] = last_coeff;

575  
576 
/* Now, use this spectrum to pick out these frequencies with higher

577 
* (relative) power/energy (which we then take to be "not noise"),

578 
* and set up a table (still in lpc[]) of (relative) gains per frequency.

579 
* These frequencies will be maintained, while others ("noise") will be

580 
* decreased in the filter output. */

581 
irange = 64.0 / range; // so irange*(maxvalue) is in the range [0, 63] 
582 
gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : 
583 
(5.0 / 14.7)); 
584 
angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); 
585 
for (n = 0; n <= 64; n++) { 
586 
float pwr;

587  
588 
idx = FFMAX(0, lrint((max  lpcs[n]) * irange)  1); 
589 
pwr = wmavoice_denoise_power_table[s>denoise_strength][idx]; 
590 
lpcs[n] = angle_mul * pwr; 
591  
592 
/* 70.57 =~ 1/log10(1.0331663) */

593 
idx = (pwr * gain_mul  0.0295) * 70.570526123; 
594 
if (idx > 127) { // fallback if index falls outside table range 
595 
coeffs[n] = wmavoice_energy_table[127] *

596 
powf(1.0331663, idx  127); 
597 
} else

598 
coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];

599 
} 
600  
601 
/* calculate the Hilbert transform of the gains, which we do (since this

602 
* is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).

603 
* Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the

604 
* "moment" of the LPCs in this filter. */

605 
s>dct.dct_calc(&s>dct, lpcs); 
606 
s>dst.dct_calc(&s>dst, lpcs); 
607  
608 
/* Split out the coefficient indexes into phase/magnitude pairs */

609 
idx = 255 + av_clip(lpcs[64], 255, 255); 
610 
coeffs[0] = coeffs[0] * s>cos[idx]; 
611 
idx = 255 + av_clip(lpcs[64]  2 * lpcs[63], 255, 255); 
612 
last_coeff = coeffs[64] * s>cos[idx];

613 
for (n = 63;; n) { 
614 
idx = 255 + av_clip(lpcs[64]  2 * lpcs[n  1], 255, 255); 
615 
coeffs[n * 2 + 1] = coeffs[n] * s>sin[idx]; 
616 
coeffs[n * 2] = coeffs[n] * s>cos[idx];

617  
618 
if (!n) break; 
619  
620 
idx = 255 + av_clip( lpcs[64]  2 * lpcs[n  1], 255, 255); 
621 
coeffs[n * 2 + 1] = coeffs[n] * s>sin[idx]; 
622 
coeffs[n * 2] = coeffs[n] * s>cos[idx];

623 
} 
624 
coeffs[1] = last_coeff;

625  
626 
/* move into real domain */

627 
s>irdft.rdft_calc(&s>irdft, coeffs); 
628  
629 
/* tilt correction and normalize scale */

630 
memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128  remainder)); 
631 
if (s>denoise_tilt_corr) {

632 
float tilt_mem = 0; 
633  
634 
coeffs[remainder  1] = 0; 
635 
ff_tilt_compensation(&tilt_mem, 
636 
1.8 * tilt_factor(coeffs, remainder  1), 
637 
coeffs, remainder); 
638 
} 
639 
sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder)); 
640 
for (n = 0; n < remainder; n++) 
641 
coeffs[n] *= sq; 
642 
} 
643  
644 
/**

645 
* This function applies a Wiener filter on the (noisy) speech signal as

646 
* a means to denoise it.

647 
*

648 
*  take RDFT of LPCs to get the power spectrum of the noise + speech;

649 
*  using this power spectrum, calculate (for each frequency) the Wiener

650 
* filter gain, which depends on the frequency power and desired level

651 
* of noise subtraction (when set too high, this leads to artifacts)

652 
* We can do this symmetrically over the Xaxis (so 04kHz is the inverse

653 
* of 48kHz);

654 
*  by doing a phase shift, calculate the Hilbert transform of this array

655 
* of perfrequency filtergains to get the filtering coefficients;

656 
*  smoothen/normalize/detilt these filter coefficients as desired;

657 
*  take RDFT of noisy sound, apply the coefficients and take its IRDFT

658 
* to get the denoised speech signal;

659 
*  the leftover (i.e. output of the IRDFT on denoised speech data beyond

660 
* the frame boundary) are saved and applied to subsequent frames by an

661 
* overlapadd method (otherwise you get clickingartifacts).

662 
*

663 
* @param s WMA Voice decoding context

664 
* @param fcb_type Frame (codebook) type

665 
* @param synth_pf input: the noisy speech signal, output: denoised speech

666 
* data; should be 16byte aligned (for ASM purposes)

667 
* @param size size of the speech data

668 
* @param lpcs LPCs used to synthesize this frame's speech data

669 
*/

670 
static void wiener_denoise(WMAVoiceContext *s, int fcb_type, 
671 
float *synth_pf, int size, 
672 
const float *lpcs) 
673 
{ 
674 
int remainder, lim, n;

675  
676 
if (fcb_type != FCB_TYPE_SILENCE) {

677 
float *tilted_lpcs = s>tilted_lpcs_pf,

678 
*coeffs = s>denoise_coeffs_pf, tilt_mem = 0;

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

686  
687 
/* The IRDFT output (127 samples for 7bit filter) beyond the frame

688 
* size is applied to the next frame. All input beyond this is zero,

689 
* and thus all output beyond this will go towards zero, hence we can

690 
* limit to min(size1, 127size) as a performance consideration. */

691 
remainder = FFMIN(127  size, size  1); 
692 
calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); 
693  
694 
/* apply coefficients (in frequency spectrum domain), i.e. complex

695 
* number multiplication */

696 
memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128  size)); 
697 
s>rdft.rdft_calc(&s>rdft, synth_pf); 
698 
s>rdft.rdft_calc(&s>rdft, coeffs); 
699 
synth_pf[0] *= coeffs[0]; 
700 
synth_pf[1] *= coeffs[1]; 
701 
for (n = 1; n < 64; n++) { 
702 
float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; 
703 
synth_pf[n * 2] = v1 * coeffs[n * 2]  v2 * coeffs[n * 2 + 1]; 
704 
synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; 
705 
} 
706 
s>irdft.rdft_calc(&s>irdft, synth_pf); 
707 
} 
708  
709 
/* merge filter output with the history of previous runs */

710 
if (s>denoise_filter_cache_size) {

711 
lim = FFMIN(s>denoise_filter_cache_size, size); 
712 
for (n = 0; n < lim; n++) 
713 
synth_pf[n] += s>denoise_filter_cache[n]; 
714 
s>denoise_filter_cache_size = lim; 
715 
memmove(s>denoise_filter_cache, &s>denoise_filter_cache[size], 
716 
sizeof(s>denoise_filter_cache[0]) * s>denoise_filter_cache_size); 
717 
} 
718  
719 
/* move remainder of filter output into a cache for future runs */

720 
if (fcb_type != FCB_TYPE_SILENCE) {

721 
lim = FFMIN(remainder, s>denoise_filter_cache_size); 
722 
for (n = 0; n < lim; n++) 
723 
s>denoise_filter_cache[n] += synth_pf[size + n]; 
724 
if (lim < remainder) {

725 
memcpy(&s>denoise_filter_cache[lim], &synth_pf[size + lim], 
726 
sizeof(s>denoise_filter_cache[0]) * (remainder  lim)); 
727 
s>denoise_filter_cache_size = remainder; 
728 
} 
729 
} 
730 
} 
731  
732 
/**

733 
* Averaging projection filter, the postfilter used in WMAVoice.

734 
*

735 
* This uses the following steps:

736 
*  A zerosynthesis filter (generate excitation from synth signal)

737 
*  Kalman smoothing on excitation, based on pitch

738 
*  Resynthesized smoothened output

739 
*  Iterative Wiener denoise filter

740 
*  Adaptive gain filter

741 
*  DC filter

742 
*

743 
* @param s WMAVoice decoding context

744 
* @param synth Speech synthesis output (before postfilter)

745 
* @param samples Output buffer for filtered samples

746 
* @param size Buffer size of synth & samples

747 
* @param lpcs Generated LPCs used for speech synthesis

748 
* @param zero_exc_pf destination for zero synthesis filter (16byte aligned)

749 
* @param fcb_type Frame type (silence, hardcoded, AWpulses or FCBpulses)

750 
* @param pitch Pitch of the input signal

751 
*/

752 
static void postfilter(WMAVoiceContext *s, const float *synth, 
753 
float *samples, int size, 
754 
const float *lpcs, float *zero_exc_pf, 
755 
int fcb_type, int pitch) 
756 
{ 
757 
float synth_filter_in_buf[MAX_FRAMESIZE / 2], 
758 
*synth_pf = &s>synth_filter_out_buf[MAX_LSPS_ALIGN16], 
759 
*synth_filter_in = zero_exc_pf; 
760  
761 
assert(size <= MAX_FRAMESIZE / 2);

762  
763 
/* generate excitation from input signal */

764 
ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s>lsps); 
765  
766 
if (fcb_type >= FCB_TYPE_AW_PULSES &&

767 
!kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) 
768 
synth_filter_in = synth_filter_in_buf; 
769  
770 
/* resynthesize speech after smoothening, and keep history */

771 
ff_celp_lp_synthesis_filterf(synth_pf, lpcs, 
772 
synth_filter_in, size, s>lsps); 
773 
memcpy(&synth_pf[s>lsps], &synth_pf[size  s>lsps], 
774 
sizeof(synth_pf[0]) * s>lsps); 
775  
776 
wiener_denoise(s, fcb_type, synth_pf, size, lpcs); 
777  
778 
adaptive_gain_control(samples, synth_pf, synth, size, 0.99, 
779 
&s>postfilter_agc); 
780  
781 
if (s>dc_level > 8) { 
782 
/* remove ultralow frequency DC noise / highpass filter;

783 
* coefficients are identical to those used in SIPR decoding,

784 
* and very closely resemble those used in AMRNB decoding. */

785 
ff_acelp_apply_order_2_transfer_function(samples, samples, 
786 
(const float[2]) { 1.99997, 1.0 }, 
787 
(const float[2]) { 1.9330735188, 0.93589198496 }, 
788 
0.93980580475, s>dcf_mem, size); 
789 
} 
790 
} 
791 
/**

792 
* @}

793 
*/

794  
795 
/**

796 
* Dequantize LSPs

797 
* @param lsps output pointer to the array that will hold the LSPs

798 
* @param num number of LSPs to be dequantized

799 
* @param values quantized values, contains n_stages values

800 
* @param sizes range (i.e. max value) of each quantized value

801 
* @param n_stages number of dequantization runs

802 
* @param table dequantization table to be used

803 
* @param mul_q LSF multiplier

804 
* @param base_q base (lowest) LSF values

805 
*/

806 
static void dequant_lsps(double *lsps, int num, 
807 
const uint16_t *values,

808 
const uint16_t *sizes,

809 
int n_stages, const uint8_t *table, 
810 
const double *mul_q, 
811 
const double *base_q) 
812 
{ 
813 
int n, m;

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

818 
double base = base_q[n], mul = mul_q[n];

819  
820 
for (m = 0; m < num; m++) 
821 
lsps[m] += base + mul * t_off[m]; 
822  
823 
table += sizes[n] * num; 
824 
} 
825 
} 
826  
827 
/**

828 
* @defgroup lsp_dequant LSP dequantization routines

829 
* LSP dequantization routines, for 10/16LSPs and independent/residual coding.

830 
* @note we assume enough bits are available, caller should check.

831 
* lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;

832 
* lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.

833 
* @{

834 
*/

835 
/**

836 
* Parse 10 independentlycoded LSPs.

837 
*/

838 
static void dequant_lsp10i(GetBitContext *gb, double *lsps) 
839 
{ 
840 
static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; 
841 
static const double mul_lsf[4] = { 
842 
5.2187144800e3, 1.4626986422e3, 
843 
9.6179549166e4, 1.1325736225e3 
844 
}; 
845 
static const double base_lsf[4] = { 
846 
M_PI * 2.15522e1, M_PI * 6.1646e2, 
847 
M_PI * 3.3486e2, M_PI * 5.7408e2 
848 
}; 
849 
uint16_t v[4];

850  
851 
v[0] = get_bits(gb, 8); 
852 
v[1] = get_bits(gb, 6); 
853 
v[2] = get_bits(gb, 5); 
854 
v[3] = get_bits(gb, 5); 
855  
856 
dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, 
857 
mul_lsf, base_lsf); 
858 
} 
859  
860 
/**

861 
* Parse 10 independentlycoded LSPs, and then derive the tables to

862 
* generate LSPs for the other frames from them (residual coding).

863 
*/

864 
static void dequant_lsp10r(GetBitContext *gb, 
865 
double *i_lsps, const double *old, 
866 
double *a1, double *a2, int q_mode) 
867 
{ 
868 
static const uint16_t vec_sizes[3] = { 128, 64, 64 }; 
869 
static const double mul_lsf[3] = { 
870 
2.5807601174e3, 1.2354460219e3, 1.1763821673e3 
871 
}; 
872 
static const double base_lsf[3] = { 
873 
M_PI * 1.07448e1, M_PI * 5.2706e2, M_PI * 5.1634e2 
874 
}; 
875 
const float (*ipol_tab)[2][10] = q_mode ? 
876 
wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; 
877 
uint16_t interpol, v[3];

878 
int n;

879  
880 
dequant_lsp10i(gb, i_lsps); 
881  
882 
interpol = get_bits(gb, 5);

883 
v[0] = get_bits(gb, 7); 
884 
v[1] = get_bits(gb, 6); 
885 
v[2] = get_bits(gb, 6); 
886  
887 
for (n = 0; n < 10; n++) { 
888 
double delta = old[n]  i_lsps[n];

889 
a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];

890 
a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; 
891 
} 
892  
893 
dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, 
894 
mul_lsf, base_lsf); 
895 
} 
896  
897 
/**

898 
* Parse 16 independentlycoded LSPs.

899 
*/

900 
static void dequant_lsp16i(GetBitContext *gb, double *lsps) 
901 
{ 
902 
static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; 
903 
static const double mul_lsf[5] = { 
904 
3.3439586280e3, 6.9908173703e4, 
905 
3.3216608306e3, 1.0334960326e3, 
906 
3.1899104283e3 
907 
}; 
908 
static const double base_lsf[5] = { 
909 
M_PI * 1.27576e1, M_PI * 2.4292e2, 
910 
M_PI * 1.28094e1, M_PI * 3.2128e2, 
911 
M_PI * 1.29816e1 
912 
}; 
913 
uint16_t v[5];

914  
915 
v[0] = get_bits(gb, 8); 
916 
v[1] = get_bits(gb, 6); 
917 
v[2] = get_bits(gb, 7); 
918 
v[3] = get_bits(gb, 6); 
919 
v[4] = get_bits(gb, 7); 
920  
921 
dequant_lsps( lsps, 5, v, vec_sizes, 2, 
922 
wmavoice_dq_lsp16i1, mul_lsf, base_lsf); 
923 
dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, 
924 
wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); 
925 
dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, 
926 
wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); 
927 
} 
928  
929 
/**

930 
* Parse 16 independentlycoded LSPs, and then derive the tables to

931 
* generate LSPs for the other frames from them (residual coding).

932 
*/

933 
static void dequant_lsp16r(GetBitContext *gb, 
934 
double *i_lsps, const double *old, 
935 
double *a1, double *a2, int q_mode) 
936 
{ 
937 
static const uint16_t vec_sizes[3] = { 128, 128, 128 }; 
938 
static const double mul_lsf[3] = { 
939 
1.2232979501e3, 1.4062241527e3, 1.6114744851e3 
940 
}; 
941 
static const double base_lsf[3] = { 
942 
M_PI * 5.5830e2, M_PI * 5.2908e2, M_PI * 5.4776e2 
943 
}; 
944 
const float (*ipol_tab)[2][16] = q_mode ? 
945 
wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; 
946 
uint16_t interpol, v[3];

947 
int n;

948  
949 
dequant_lsp16i(gb, i_lsps); 
950  
951 
interpol = get_bits(gb, 5);

952 
v[0] = get_bits(gb, 7); 
953 
v[1] = get_bits(gb, 7); 
954 
v[2] = get_bits(gb, 7); 
955  
956 
for (n = 0; n < 16; n++) { 
957 
double delta = old[n]  i_lsps[n];

958 
a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];

959 
a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; 
960 
} 
961  
962 
dequant_lsps( a2, 10, v, vec_sizes, 1, 
963 
wmavoice_dq_lsp16r1, mul_lsf, base_lsf); 
964 
dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, 
965 
wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); 
966 
dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, 
967 
wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); 
968 
} 
969  
970 
/**

971 
* @}

972 
* @defgroup aw Pitchadaptive window coding functions

973 
* The next few functions are for pitchadaptive window coding.

974 
* @{

975 
*/

976 
/**

977 
* Parse the offset of the first pitchadaptive window pulses, and

978 
* the distribution of pulses between the two blocks in this frame.

979 
* @param s WMA Voice decoding context private data

980 
* @param gb bit I/O context

981 
* @param pitch pitch for each block in this frame

982 
*/

983 
static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, 
984 
const int *pitch) 
985 
{ 
986 
static const int16_t start_offset[94] = { 
987 
11, 9, 7, 5, 3, 1, 1, 3, 5, 7, 9, 11, 
988 
13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, 
989 
27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, 
990 
45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, 
991 
69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, 
992 
93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, 
993 
117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, 
994 
141, 143, 145, 147, 149, 151, 153, 155, 157, 159 
995 
}; 
996 
int bits, offset;

997  
998 
/* position of pulse */

999 
s>aw_idx_is_ext = 0;

1000 
if ((bits = get_bits(gb, 6)) >= 54) { 
1001 
s>aw_idx_is_ext = 1;

1002 
bits += (bits  54) * 3 + get_bits(gb, 2); 
1003 
} 
1004  
1005 
/* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count

1006 
* the distribution of the pulses in each block contained in this frame. */

1007 
s>aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; 
1008 
for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; 
1009 
s>aw_n_pulses[0] = (pitch[0]  1 + MAX_FRAMESIZE / 2  offset) / pitch[0]; 
1010 
s>aw_first_pulse_off[0] = offset  s>aw_pulse_range / 2; 
1011 
offset += s>aw_n_pulses[0] * pitch[0]; 
1012 
s>aw_n_pulses[1] = (pitch[1]  1 + MAX_FRAMESIZE  offset) / pitch[1]; 
1013 
s>aw_first_pulse_off[1] = offset  (MAX_FRAMESIZE + s>aw_pulse_range) / 2; 
1014  
1015 
/* if continuing from a position before the block, reset position to

1016 
* start of block (when corrected for the range over which it can be

1017 
* spread in aw_pulse_set1()). */

1018 
if (start_offset[bits] < MAX_FRAMESIZE / 2) { 
1019 
while (s>aw_first_pulse_off[1]  pitch[1] + s>aw_pulse_range > 0) 
1020 
s>aw_first_pulse_off[1] = pitch[1]; 
1021 
if (start_offset[bits] < 0) 
1022 
while (s>aw_first_pulse_off[0]  pitch[0] + s>aw_pulse_range > 0) 
1023 
s>aw_first_pulse_off[0] = pitch[0]; 
1024 
} 
1025 
} 
1026  
1027 
/**

1028 
* Apply second set of pitchadaptive window pulses.

1029 
* @param s WMA Voice decoding context private data

1030 
* @param gb bit I/O context

1031 
* @param block_idx block index in frame [0, 1]

1032 
* @param fcb structure containing fixed codebook vector info

1033 
*/

1034 
static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, 
1035 
int block_idx, AMRFixed *fcb)

1036 
{ 
1037 
uint16_t use_mask_mem[9]; // only 5 are used, rest is padding 
1038 
uint16_t *use_mask = use_mask_mem + 2;

1039 
/* in this function, idx is the index in the 80bit (+ padding) use_mask

1040 
* bitarray. Since use_mask consists of 16bit values, the lower 4 bits

1041 
* of idx are the position of the bit within a particular item in the

1042 
* array (0 being the most significant bit, and 15 being the least

1043 
* significant bit), and the remainder (>> 4) is the index in the

1044 
* use_mask[]array. This is faster and uses less memory than using a

1045 
* 80byte/80int array. */

1046 
int pulse_off = s>aw_first_pulse_off[block_idx],

1047 
pulse_start, n, idx, range, aidx, start_off = 0;

1048  
1049 
/* set offset of first pulse to within this block */

1050 
if (s>aw_n_pulses[block_idx] > 0) 
1051 
while (pulse_off + s>aw_pulse_range < 1) 
1052 
pulse_off += fcb>pitch_lag; 
1053  
1054 
/* find range per pulse */

1055 
if (s>aw_n_pulses[0] > 0) { 
1056 
if (block_idx == 0) { 
1057 
range = 32;

1058 
} else /* block_idx = 1 */ { 
1059 
range = 8;

1060 
if (s>aw_n_pulses[block_idx] > 0) 
1061 
pulse_off = s>aw_next_pulse_off_cache; 
1062 
} 
1063 
} else

1064 
range = 16;

1065 
pulse_start = s>aw_n_pulses[block_idx] > 0 ? pulse_off  range / 2 : 0; 
1066  
1067 
/* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,

1068 
* in the range of [pulse_off, pulse_off + s>aw_pulse_range], and thus

1069 
* we exclude that range from being pulsed again in this function. */

1070 
memset(&use_mask[2], 0, 2 * sizeof(use_mask[0])); 
1071 
memset( use_mask, 1, 5 * sizeof(use_mask[0])); 
1072 
memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); 
1073 
if (s>aw_n_pulses[block_idx] > 0) 
1074 
for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb>pitch_lag) { 
1075 
int excl_range = s>aw_pulse_range; // always 16 or 24 
1076 
uint16_t *use_mask_ptr = &use_mask[idx >> 4];

1077 
int first_sh = 16  (idx & 15); 
1078 
*use_mask_ptr++ &= 0xFFFF << first_sh;

1079 
excl_range = first_sh; 
1080 
if (excl_range >= 16) { 
1081 
*use_mask_ptr++ = 0;

1082 
*use_mask_ptr &= 0xFFFF >> (excl_range  16); 
1083 
} else

1084 
*use_mask_ptr &= 0xFFFF >> excl_range;

1085 
} 
1086  
1087 
/* find the 'aidx'th offset that is not excluded */

1088 
aidx = get_bits(gb, s>aw_n_pulses[0] > 0 ? 5  2 * block_idx : 4); 
1089 
for (n = 0; n <= aidx; pulse_start++) { 
1090 
for (idx = pulse_start; idx < 0; idx += fcb>pitch_lag) ; 
1091 
if (idx >= MAX_FRAMESIZE / 2) { // find from zero 
1092 
if (use_mask[0]) idx = 0x0F; 
1093 
else if (use_mask[1]) idx = 0x1F; 
1094 
else if (use_mask[2]) idx = 0x2F; 
1095 
else if (use_mask[3]) idx = 0x3F; 
1096 
else if (use_mask[4]) idx = 0x4F; 
1097 
else return; 
1098 
idx = av_log2_16bit(use_mask[idx >> 4]);

1099 
} 
1100 
if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { 
1101 
use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); 
1102 
n++; 
1103 
start_off = idx; 
1104 
} 
1105 
} 
1106  
1107 
fcb>x[fcb>n] = start_off; 
1108 
fcb>y[fcb>n] = get_bits1(gb) ? 1.0 : 1.0; 
1109 
fcb>n++; 
1110  
1111 
/* set offset for next block, relative to start of that block */

1112 
n = (MAX_FRAMESIZE / 2  start_off) % fcb>pitch_lag;

1113 
s>aw_next_pulse_off_cache = n ? fcb>pitch_lag  n : 0;

1114 
} 
1115  
1116 
/**

1117 
* Apply first set of pitchadaptive window pulses.

1118 
* @param s WMA Voice decoding context private data

1119 
* @param gb bit I/O context

1120 
* @param block_idx block index in frame [0, 1]

1121 
* @param fcb storage location for fixed codebook pulse info

1122 
*/

1123 
static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, 
1124 
int block_idx, AMRFixed *fcb)

1125 
{ 
1126 
int val = get_bits(gb, 12  2 * (s>aw_idx_is_ext && !block_idx)); 
1127 
float v;

1128  
1129 
if (s>aw_n_pulses[block_idx] > 0) { 
1130 
int n, v_mask, i_mask, sh, n_pulses;

1131  
1132 
if (s>aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each 
1133 
n_pulses = 3;

1134 
v_mask = 8;

1135 
i_mask = 7;

1136 
sh = 4;

1137 
} else { // 4 pulses, 1:sign + 2:index each 
1138 
n_pulses = 4;

1139 
v_mask = 4;

1140 
i_mask = 3;

1141 
sh = 3;

1142 
} 
1143  
1144 
for (n = n_pulses  1; n >= 0; n, val >>= sh) { 
1145 
fcb>y[fcb>n] = (val & v_mask) ? 1.0 : 1.0; 
1146 
fcb>x[fcb>n] = (val & i_mask) * n_pulses + n + 
1147 
s>aw_first_pulse_off[block_idx]; 
1148 
while (fcb>x[fcb>n] < 0) 
1149 
fcb>x[fcb>n] += fcb>pitch_lag; 
1150 
if (fcb>x[fcb>n] < MAX_FRAMESIZE / 2) 
1151 
fcb>n++; 
1152 
} 
1153 
} else {

1154 
int num2 = (val & 0x1FF) >> 1, delta, idx; 
1155  
1156 
if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } 
1157 
else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1  1 * 77; } 
1158 
else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1  2 * 76; } 
1159 
else { delta = 7; idx = num2 + 1  3 * 75; } 
1160 
v = (val & 0x200) ? 1.0 : 1.0; 
1161  
1162 
fcb>no_repeat_mask = 3 << fcb>n;

1163 
fcb>x[fcb>n] = idx  delta; 
1164 
fcb>y[fcb>n] = v; 
1165 
fcb>x[fcb>n + 1] = idx;

1166 
fcb>y[fcb>n + 1] = (val & 1) ? v : v; 
1167 
fcb>n += 2;

1168 
} 
1169 
} 
1170  
1171 
/**

1172 
* @}

1173 
*

1174 
* Generate a random number from frame_cntr and block_idx, which will lief

1175 
* in the range [0, 1000  block_size] (so it can be used as an index in a

1176 
* table of size 1000 of which you want to read block_size entries).

1177 
*

1178 
* @param frame_cntr current frame number

1179 
* @param block_num current block index

1180 
* @param block_size amount of entries we want to read from a table

1181 
* that has 1000 entries

1182 
* @return a (non)random number in the [0, 1000  block_size] range.

1183 
*/

1184 
static int pRNG(int frame_cntr, int block_num, int block_size) 
1185 
{ 
1186 
/* array to simplify the calculation of z:

1187 
* y = (x % 9) * 5 + 6;

1188 
* z = (49995 * x) / y;

1189 
* Since y only has 9 values, we can remove the division by using a

1190 
* LUT and using FASTDIVstyle divisions. For each of the 9 values

1191 
* of y, we can rewrite z as:

1192 
* z = x * (49995 / y) + x * ((49995 % y) / y)

1193 
* In this table, each col represents one possible value of y, the

1194 
* first number is 49995 / y, and the second is the FASTDIV variant

1195 
* of 49995 % y / y. */

1196 
static const unsigned int div_tbl[9][2] = { 
1197 
{ 8332, 3 * 715827883U }, // y = 6 
1198 
{ 4545, 0 * 390451573U }, // y = 11 
1199 
{ 3124, 11 * 268435456U }, // y = 16 
1200 
{ 2380, 15 * 204522253U }, // y = 21 
1201 
{ 1922, 23 * 165191050U }, // y = 26 
1202 
{ 1612, 23 * 138547333U }, // y = 31 
1203 
{ 1388, 27 * 119304648U }, // y = 36 
1204 
{ 1219, 16 * 104755300U }, // y = 41 
1205 
{ 1086, 39 * 93368855U } // y = 46 
1206 
}; 
1207 
unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; 
1208 
if (x >= 0xFFFF) x = 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, 
1209 
// so this is effectively a modulo (%)

1210 
y = x  9 * MULH(477218589, x); // x % 9 
1211 
z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); 
1212 
// z = x * 49995 / (y * 5 + 6)

1213 
return z % (1000  block_size); 
1214 
} 
1215  
1216 
/**

1217 
* Parse hardcoded signal for a single block.

1218 
* @note see #synth_block().

1219 
*/

1220 
static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, 
1221 
int block_idx, int size, 
1222 
const struct frame_type_desc *frame_desc, 
1223 
float *excitation)

1224 
{ 
1225 
float gain;

1226 
int n, r_idx;

1227  
1228 
assert(size <= MAX_FRAMESIZE); 
1229  
1230 
/* Set the offset from which we start reading wmavoice_std_codebook */

1231 
if (frame_desc>fcb_type == FCB_TYPE_SILENCE) {

1232 
r_idx = pRNG(s>frame_cntr, block_idx, size); 
1233 
gain = s>silence_gain; 
1234 
} else /* FCB_TYPE_HARDCODED */ { 
1235 
r_idx = get_bits(gb, 8);

1236 
gain = wmavoice_gain_universal[get_bits(gb, 6)];

1237 
} 
1238  
1239 
/* Clear gain prediction parameters */

1240 
memset(s>gain_pred_err, 0, sizeof(s>gain_pred_err)); 
1241  
1242 
/* Apply gain to hardcoded codebook and use that as excitation signal */

1243 
for (n = 0; n < size; n++) 
1244 
excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; 
1245 
} 
1246  
1247 
/**

1248 
* Parse FCB/ACB signal for a single block.

1249 
* @note see #synth_block().

1250 
*/

1251 
static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, 
1252 
int block_idx, int size, 
1253 
int block_pitch_sh2,

1254 
const struct frame_type_desc *frame_desc, 
1255 
float *excitation)

1256 
{ 
1257 
static const float gain_coeff[6] = { 
1258 
0.8169, 0.06545, 0.1726, 0.0185, 0.0359, 0.0458 
1259 
}; 
1260 
float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; 
1261 
int n, idx, gain_weight;

1262 
AMRFixed fcb; 
1263  
1264 
assert(size <= MAX_FRAMESIZE / 2);

1265 
memset(pulses, 0, sizeof(*pulses) * size); 
1266  
1267 
fcb.pitch_lag = block_pitch_sh2 >> 2;

1268 
fcb.pitch_fac = 1.0; 
1269 
fcb.no_repeat_mask = 0;

1270 
fcb.n = 0;

1271  
1272 
/* For the other frame types, this is where we apply the innovation

1273 
* (fixed) codebook pulses of the speech signal. */

1274 
if (frame_desc>fcb_type == FCB_TYPE_AW_PULSES) {

1275 
aw_pulse_set1(s, gb, block_idx, &fcb); 
1276 
aw_pulse_set2(s, gb, block_idx, &fcb); 
1277 
} else /* FCB_TYPE_EXC_PULSES */ { 
1278 
int offset_nbits = 5  frame_desc>log_n_blocks; 
1279  
1280 
fcb.no_repeat_mask = 1;

1281 
/* similar to ff_decode_10_pulses_35bits(), but with single pulses

1282 
* (instead of double) for a subset of pulses */

1283 
for (n = 0; n < 5; n++) { 
1284 
float sign;

1285 
int pos1, pos2;

1286  
1287 
sign = get_bits1(gb) ? 1.0 : 1.0; 
1288 
pos1 = get_bits(gb, offset_nbits); 
1289 
fcb.x[fcb.n] = n + 5 * pos1;

1290 
fcb.y[fcb.n++] = sign; 
1291 
if (n < frame_desc>dbl_pulses) {

1292 
pos2 = get_bits(gb, offset_nbits); 
1293 
fcb.x[fcb.n] = n + 5 * pos2;

1294 
fcb.y[fcb.n++] = (pos1 < pos2) ? sign : sign; 
1295 
} 
1296 
} 
1297 
} 
1298 
ff_set_fixed_vector(pulses, &fcb, 1.0, size); 
1299  
1300 
/* Calculate gain for adaptive & fixed codebook signal.

1301 
* see ff_amr_set_fixed_gain(). */

1302 
idx = get_bits(gb, 7);

1303 
fcb_gain = expf(ff_dot_productf(s>gain_pred_err, gain_coeff, 6) 

1304 
5.2409161640 + wmavoice_gain_codebook_fcb[idx]); 
1305 
acb_gain = wmavoice_gain_codebook_acb[idx]; 
1306 
pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], 
1307 
2.9957322736 /* log(0.05) */, 
1308 
1.6094379124 /* log(5.0) */); 
1309  
1310 
gain_weight = 8 >> frame_desc>log_n_blocks;

1311 
memmove(&s>gain_pred_err[gain_weight], s>gain_pred_err, 
1312 
sizeof(*s>gain_pred_err) * (6  gain_weight)); 
1313 
for (n = 0; n < gain_weight; n++) 
1314 
s>gain_pred_err[n] = pred_err; 
1315  
1316 
/* Calculation of adaptive codebook */

1317 
if (frame_desc>acb_type == ACB_TYPE_ASYMMETRIC) {

1318 
int len;

1319 
for (n = 0; n < size; n += len) { 
1320 
int next_idx_sh16;

1321 
int abs_idx = block_idx * size + n;

1322 
int pitch_sh16 = (s>last_pitch_val << 16) + 
1323 
s>pitch_diff_sh16 * abs_idx; 
1324 
int pitch = (pitch_sh16 + 0x6FFF) >> 16; 
1325 
int idx_sh16 = ((pitch << 16)  pitch_sh16) * 8 + 0x58000; 
1326 
idx = idx_sh16 >> 16;

1327 
if (s>pitch_diff_sh16) {

1328 
if (s>pitch_diff_sh16 > 0) { 
1329 
next_idx_sh16 = (idx_sh16) &~ 0xFFFF;

1330 
} else

1331 
next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; 
1332 
len = av_clip((idx_sh16  next_idx_sh16) / s>pitch_diff_sh16 / 8,

1333 
1, size  n);

1334 
} else

1335 
len = size; 
1336  
1337 
ff_acelp_interpolatef(&excitation[n], &excitation[n  pitch], 
1338 
wmavoice_ipol1_coeffs, 17,

1339 
idx, 9, len);

1340 
} 
1341 
} else /* ACB_TYPE_HAMMING */ { 
1342 
int block_pitch = block_pitch_sh2 >> 2; 
1343 
idx = block_pitch_sh2 & 3;

1344 
if (idx) {

1345 
ff_acelp_interpolatef(excitation, &excitation[block_pitch], 
1346 
wmavoice_ipol2_coeffs, 4,

1347 
idx, 8, size);

1348 
} else

1349 
av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, 
1350 
sizeof(float) * size); 
1351 
} 
1352  
1353 
/* Interpolate ACB/FCB and use as excitation signal */

1354 
ff_weighted_vector_sumf(excitation, excitation, pulses, 
1355 
acb_gain, fcb_gain, size); 
1356 
} 
1357  
1358 
/**

1359 
* Parse data in a single block.

1360 
* @note we assume enough bits are available, caller should check.

1361 
*

1362 
* @param s WMA Voice decoding context private data

1363 
* @param gb bit I/O context

1364 
* @param block_idx index of the toberead block

1365 
* @param size amount of samples to be read in this block

1366 
* @param block_pitch_sh2 pitch for this block << 2

1367 
* @param lsps LSPs for (the end of) this frame

1368 
* @param prev_lsps LSPs for the last frame

1369 
* @param frame_desc frame type descriptor

1370 
* @param excitation target memory for the ACB+FCB interpolated signal

1371 
* @param synth target memory for the speech synthesis filter output

1372 
* @return 0 on success, <0 on error.

1373 
*/

1374 
static void synth_block(WMAVoiceContext *s, GetBitContext *gb, 
1375 
int block_idx, int size, 
1376 
int block_pitch_sh2,

1377 
const double *lsps, const double *prev_lsps, 
1378 
const struct frame_type_desc *frame_desc, 
1379 
float *excitation, float *synth) 
1380 
{ 
1381 
double i_lsps[MAX_LSPS];

1382 
float lpcs[MAX_LSPS];

1383 
float fac;

1384 
int n;

1385  
1386 
if (frame_desc>acb_type == ACB_TYPE_NONE)

1387 
synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); 
1388 
else

1389 
synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, 
1390 
frame_desc, excitation); 
1391  
1392 
/* convert interpolated LSPs to LPCs */

1393 
fac = (block_idx + 0.5) / frame_desc>n_blocks; 
1394 
for (n = 0; n < s>lsps; n++) // LSF > LSP 
1395 
i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n]  prev_lsps[n])); 
1396 
ff_acelp_lspd2lpc(i_lsps, lpcs, s>lsps >> 1);

1397  
1398 
/* Speech synthesis */

1399 
ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s>lsps); 
1400 
} 
1401  
1402 
/**

1403 
* Synthesize output samples for a single frame.

1404 
* @note we assume enough bits are available, caller should check.

1405 
*

1406 
* @param ctx WMA Voice decoder context

1407 
* @param gb bit I/O context (s>gb or one for crosspacket superframes)

1408 
* @param frame_idx Frame number within superframe [02]

1409 
* @param samples pointer to output sample buffer, has space for at least 160

1410 
* samples

1411 
* @param lsps LSP array

1412 
* @param prev_lsps array of previous frame's LSPs

1413 
* @param excitation target buffer for excitation signal

1414 
* @param synth target buffer for synthesized speech data

1415 
* @return 0 on success, <0 on error.

1416 
*/

1417 
static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, 
1418 
float *samples,

1419 
const double *lsps, const double *prev_lsps, 
1420 
float *excitation, float *synth) 
1421 
{ 
1422 
WMAVoiceContext *s = ctx>priv_data; 
1423 
int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;

1424 
int pitch[MAX_BLOCKS], last_block_pitch;

1425  
1426 
/* Parse frame type ("frame header"), see frame_descs */

1427 
int bd_idx = s>vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], 
1428 
block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; 
1429  
1430 
if (bd_idx < 0) { 
1431 
av_log(ctx, AV_LOG_ERROR, 
1432 
"Invalid frame type VLC code, skipping\n");

1433 
return 1; 
1434 
} 
1435  
1436 
/* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitchperframe") */

1437 
if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {

1438 
/* Pitch is provided per frame, which is interpreted as the pitch of

1439 
* the last sample of the last block of this frame. We can interpolate

1440 
* the pitch of other blocks (and even pitchpersample) by gradually

1441 
* incrementing/decrementing prev_frame_pitch to cur_pitch_val. */

1442 
n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;

1443 
log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;

1444 
cur_pitch_val = s>min_pitch_val + get_bits(gb, s>pitch_nbits); 
1445 
cur_pitch_val = FFMIN(cur_pitch_val, s>max_pitch_val  1);

1446 
if (s>last_acb_type == ACB_TYPE_NONE 

1447 
20 * abs(cur_pitch_val  s>last_pitch_val) >

1448 
(cur_pitch_val + s>last_pitch_val)) 
1449 
s>last_pitch_val = cur_pitch_val; 
1450  
1451 
/* pitch per block */

1452 
for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { 
1453 
int fac = n * 2 + 1; 
1454  
1455 
pitch[n] = (MUL16(fac, cur_pitch_val) + 
1456 
MUL16((n_blocks_x2  fac), s>last_pitch_val) + 
1457 
frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; 
1458 
} 
1459  
1460 
/* "pitchdiffpersample" for calculation of pitch per sample */

1461 
s>pitch_diff_sh16 = 
1462 
((cur_pitch_val  s>last_pitch_val) << 16) / MAX_FRAMESIZE;

1463 
} 
1464  
1465 
/* Global gain (if silence) and pitchadaptive window coordinates */

1466 
switch (frame_descs[bd_idx].fcb_type) {

1467 
case FCB_TYPE_SILENCE:

1468 
s>silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];

1469 
break;

1470 
case FCB_TYPE_AW_PULSES:

1471 
aw_parse_coords(s, gb, pitch); 
1472 
break;

1473 
} 
1474  
1475 
for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { 
1476 
int bl_pitch_sh2;

1477  
1478 
/* Pitch calculation for ACB_TYPE_HAMMING ("pitchperblock") */

1479 
switch (frame_descs[bd_idx].acb_type) {

1480 
case ACB_TYPE_HAMMING: {

1481 
/* Pitch is given per block. Perblock pitches are encoded as an

1482 
* absolute value for the first block, and then delta values

1483 
* relative to this value) for all subsequent blocks. The scale of

1484 
* this pitch value is semilogaritmic compared to its use in the

1485 
* decoder, so we convert it to normal scale also. */

1486 
int block_pitch,

1487 
t1 = (s>block_conv_table[1]  s>block_conv_table[0]) << 2, 
1488 
t2 = (s>block_conv_table[2]  s>block_conv_table[1]) << 1, 
1489 
t3 = s>block_conv_table[3]  s>block_conv_table[2] + 1; 
1490  
1491 
if (n == 0) { 
1492 
block_pitch = get_bits(gb, s>block_pitch_nbits); 
1493 
} else

1494 
block_pitch = last_block_pitch  s>block_delta_pitch_hrange + 
1495 
get_bits(gb, s>block_delta_pitch_nbits); 
1496 
/* Convert last_ so that any next delta is within _range */

1497 
last_block_pitch = av_clip(block_pitch, 
1498 
s>block_delta_pitch_hrange, 
1499 
s>block_pitch_range  
1500 
s>block_delta_pitch_hrange); 
1501  
1502 
/* Convert semilogstyle scale back to normal scale */

1503 
if (block_pitch < t1) {

1504 
bl_pitch_sh2 = (s>block_conv_table[0] << 2) + block_pitch; 
1505 
} else {

1506 
block_pitch = t1; 
1507 
if (block_pitch < t2) {

1508 
bl_pitch_sh2 = 
1509 
(s>block_conv_table[1] << 2) + (block_pitch << 1); 
1510 
} else {

1511 
block_pitch = t2; 
1512 
if (block_pitch < t3) {

1513 
bl_pitch_sh2 = 
1514 
(s>block_conv_table[2] + block_pitch) << 2; 
1515 
} else

1516 
bl_pitch_sh2 = s>block_conv_table[3] << 2; 
1517 
} 
1518 
} 
1519 
pitch[n] = bl_pitch_sh2 >> 2;

1520 
break;

1521 
} 
1522  
1523 
case ACB_TYPE_ASYMMETRIC: {

1524 
bl_pitch_sh2 = pitch[n] << 2;

1525 
break;

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

1530 
break;

1531 
} 
1532  
1533 
synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, 
1534 
lsps, prev_lsps, &frame_descs[bd_idx], 
1535 
&excitation[n * block_nsamples], 
1536 
&synth[n * block_nsamples]); 
1537 
} 
1538  
1539 
/* Averaging projection filter, if applicable. Else, just copy samples

1540 
* from synthesis buffer */

1541 
if (s>do_apf) {

1542 
double i_lsps[MAX_LSPS];

1543 
float lpcs[MAX_LSPS];

1544  
1545 
for (n = 0; n < s>lsps; n++) // LSF > LSP 
1546 
i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); 
1547 
ff_acelp_lspd2lpc(i_lsps, lpcs, s>lsps >> 1);

1548 
postfilter(s, synth, samples, 80, lpcs,

1549 
&s>zero_exc_pf[s>history_nsamples + MAX_FRAMESIZE * frame_idx], 
1550 
frame_descs[bd_idx].fcb_type, pitch[0]);

1551  
1552 
for (n = 0; n < s>lsps; n++) // LSF > LSP 
1553 
i_lsps[n] = cos(lsps[n]); 
1554 
ff_acelp_lspd2lpc(i_lsps, lpcs, s>lsps >> 1);

1555 
postfilter(s, &synth[80], &samples[80], 80, lpcs, 
1556 
&s>zero_exc_pf[s>history_nsamples + MAX_FRAMESIZE * frame_idx + 80],

1557 
frame_descs[bd_idx].fcb_type, pitch[0]);

1558 
} else

1559 
memcpy(samples, synth, 160 * sizeof(synth[0])); 
1560  
1561 
/* Cache values for next frame */

1562 
s>frame_cntr++; 
1563 
if (s>frame_cntr >= 0xFFFF) s>frame_cntr = 0xFFFF; // i.e. modulo (%) 
1564 
s>last_acb_type = frame_descs[bd_idx].acb_type; 
1565 
switch (frame_descs[bd_idx].acb_type) {

1566 
case ACB_TYPE_NONE:

1567 
s>last_pitch_val = 0;

1568 
break;

1569 
case ACB_TYPE_ASYMMETRIC:

1570 
s>last_pitch_val = cur_pitch_val; 
1571 
break;

1572 
case ACB_TYPE_HAMMING:

1573 
s>last_pitch_val = pitch[frame_descs[bd_idx].n_blocks  1];

1574 
break;

1575 
} 
1576  
1577 
return 0; 
1578 
} 
1579  
1580 
/**

1581 
* Ensure minimum value for first item, maximum value for last value,

1582 
* proper spacing between each value and proper ordering.

1583 
*

1584 
* @param lsps array of LSPs

1585 
* @param num size of LSP array

1586 
*

1587 
* @note basically a double version of #ff_acelp_reorder_lsf(), might be

1588 
* useful to put in a generic location later on. Parts are also

1589 
* present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),

1590 
* which is in float.

1591 
*/

1592 
static void stabilize_lsps(double *lsps, int num) 
1593 
{ 
1594 
int n, m, l;

1595  
1596 
/* set minimum value for first, maximum value for last and minimum

1597 
* spacing between LSF values.

1598 
* Very similar to ff_set_min_dist_lsf(), but in double. */

1599 
lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); 
1600 
for (n = 1; n < num; n++) 
1601 
lsps[n] = FFMAX(lsps[n], lsps[n  1] + 0.0125 * M_PI); 
1602 
lsps[num  1] = FFMIN(lsps[num  1], 0.9985 * M_PI); 
1603  
1604 
/* reorder (looks like onetime / nonrecursed bubblesort).

1605 
* Very similar to ff_sort_nearly_sorted_floats(), but in double. */

1606 
for (n = 1; n < num; n++) { 
1607 
if (lsps[n] < lsps[n  1]) { 
1608 
for (m = 1; m < num; m++) { 
1609 
double tmp = lsps[m];

1610 
for (l = m  1; l >= 0; l) { 
1611 
if (lsps[l] <= tmp) break; 
1612 
lsps[l + 1] = lsps[l];

1613 
} 
1614 
lsps[l + 1] = tmp;

1615 
} 
1616 
break;

1617 
} 
1618 
} 
1619 
} 
1620  
1621 
/**

1622 
* Test if there's enough bits to read 1 superframe.

1623 
*

1624 
* @param orig_gb bit I/O context used for reading. This function

1625 
* does not modify the state of the bitreader; it

1626 
* only uses it to copy the current stream position

1627 
* @param s WMA Voice decoding context private data

1628 
* @return 1 if unsupported, 1 on not enough bits or 0 if OK.

1629 
*/

1630 
static int check_bits_for_superframe(GetBitContext *orig_gb, 
1631 
WMAVoiceContext *s) 
1632 
{ 
1633 
GetBitContext s_gb, *gb = &s_gb; 
1634 
int n, need_bits, bd_idx;

1635 
const struct frame_type_desc *frame_desc; 
1636  
1637 
/* initialize a copy */

1638 
init_get_bits(gb, orig_gb>buffer, orig_gb>size_in_bits); 
1639 
skip_bits_long(gb, get_bits_count(orig_gb)); 
1640 
assert(get_bits_left(gb) == get_bits_left(orig_gb)); 
1641  
1642 
/* superframe header */

1643 
if (get_bits_left(gb) < 14) 
1644 
return 1; 
1645 
if (!get_bits1(gb))

1646 
return 1; // WMAProinWMAVoice superframe 
1647 
if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe 
1648 
if (s>has_residual_lsps) { // residual LSPs (for all frames) 
1649 
if (get_bits_left(gb) < s>sframe_lsp_bitsize)

1650 
return 1; 
1651 
skip_bits_long(gb, s>sframe_lsp_bitsize); 
1652 
} 
1653  
1654 
/* frames */

1655 
for (n = 0; n < MAX_FRAMES; n++) { 
1656 
int aw_idx_is_ext = 0; 
1657  
1658 
if (!s>has_residual_lsps) { // independent LSPs (perframe) 
1659 
if (get_bits_left(gb) < s>frame_lsp_bitsize) return 1; 
1660 
skip_bits_long(gb, s>frame_lsp_bitsize); 
1661 
} 
1662 
bd_idx = s>vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; 
1663 
if (bd_idx < 0) 
1664 
return 1; // invalid frame type VLC code 
1665 
frame_desc = &frame_descs[bd_idx]; 
1666 
if (frame_desc>acb_type == ACB_TYPE_ASYMMETRIC) {

1667 
if (get_bits_left(gb) < s>pitch_nbits)

1668 
return 1; 
1669 
skip_bits_long(gb, s>pitch_nbits); 
1670 
} 
1671 
if (frame_desc>fcb_type == FCB_TYPE_SILENCE) {

1672 
skip_bits(gb, 8);

1673 
} else if (frame_desc>fcb_type == FCB_TYPE_AW_PULSES) { 
1674 
int tmp = get_bits(gb, 6); 
1675 
if (tmp >= 0x36) { 
1676 
skip_bits(gb, 2);

1677 
aw_idx_is_ext = 1;

1678 
} 
1679 
} 
1680  
1681 
/* blocks */

1682 
if (frame_desc>acb_type == ACB_TYPE_HAMMING) {

1683 
need_bits = s>block_pitch_nbits + 
1684 
(frame_desc>n_blocks  1) * s>block_delta_pitch_nbits;

1685 
} else if (frame_desc>fcb_type == FCB_TYPE_AW_PULSES) { 
1686 
need_bits = 2 * !aw_idx_is_ext;

1687 
} else

1688 
need_bits = 0;

1689 
need_bits += frame_desc>frame_size; 
1690 
if (get_bits_left(gb) < need_bits)

1691 
return 1; 
1692 
skip_bits_long(gb, need_bits); 
1693 
} 
1694  
1695 
return 0; 
1696 
} 
1697  
1698 
/**

1699 
* Synthesize output samples for a single superframe. If we have any data

1700 
* cached in s>sframe_cache, that will be used instead of whatever is loaded

1701 
* in s>gb.

1702 
*

1703 
* WMA Voice superframes contain 3 frames, each containing 160 audio samples,

1704 
* to give a total of 480 samples per frame. See #synth_frame() for frame

1705 
* parsing. In addition to 3 frames, superframes can also contain the LSPs

1706 
* (if these are globally specified for all frames (residually); they can

1707 
* also be specified individually perframe. See the s>has_residual_lsps

1708 
* option), and can specify the number of samples encoded in this superframe

1709 
* (if less than 480), usually used to prevent blanks at track boundaries.

1710 
*

1711 
* @param ctx WMA Voice decoder context

1712 
* @param samples pointer to output buffer for voice samples

1713 
* @param data_size pointer containing the size of #samples on input, and the

1714 
* amount of #samples filled on output

1715 
* @return 0 on success, <0 on error or 1 if there was not enough data to

1716 
* fully parse the superframe

1717 
*/

1718 
static int synth_superframe(AVCodecContext *ctx, 
1719 
float *samples, int *data_size) 
1720 
{ 
1721 
WMAVoiceContext *s = ctx>priv_data; 
1722 
GetBitContext *gb = &s>gb, s_gb; 
1723 
int n, res, n_samples = 480; 
1724 
double lsps[MAX_FRAMES][MAX_LSPS];

1725 
const double *mean_lsf = s>lsps == 16 ? 
1726 
wmavoice_mean_lsf16[s>lsp_def_mode] : wmavoice_mean_lsf10[s>lsp_def_mode]; 
1727 
float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; 
1728 
float synth[MAX_LSPS + MAX_SFRAMESIZE];

1729  
1730 
memcpy(synth, s>synth_history, 
1731 
s>lsps * sizeof(*synth));

1732 
memcpy(excitation, s>excitation_history, 
1733 
s>history_nsamples * sizeof(*excitation));

1734  
1735 
if (s>sframe_cache_size > 0) { 
1736 
gb = &s_gb; 
1737 
init_get_bits(gb, s>sframe_cache, s>sframe_cache_size); 
1738 
s>sframe_cache_size = 0;

1739 
} 
1740  
1741 
if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; 
1742  
1743 
/* First bit is speech/music bit, it differentiates between WMAVoice

1744 
* speech samples (the actual codec) and WMAVoice music samples, which

1745 
* are really WMAProinWMAVoicesuperframes. I've never seen those in

1746 
* the wild yet. */

1747 
if (!get_bits1(gb)) {

1748 
av_log_missing_feature(ctx, "WMAProinWMAVoice support", 1); 
1749 
return 1; 
1750 
} 
1751  
1752 
/* (optional) nr. of samples in superframe; always <= 480 and >= 0 */

1753 
if (get_bits1(gb)) {

1754 
if ((n_samples = get_bits(gb, 12)) > 480) { 
1755 
av_log(ctx, AV_LOG_ERROR, 
1756 
"Superframe encodes >480 samples (%d), not allowed\n",

1757 
n_samples); 
1758 
return 1; 
1759 
} 
1760 
} 
1761 
/* Parse LSPs, if global for the superframe (can also be perframe). */

1762 
if (s>has_residual_lsps) {

1763 
double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; 
1764  
1765 
for (n = 0; n < s>lsps; n++) 
1766 
prev_lsps[n] = s>prev_lsps[n]  mean_lsf[n]; 
1767  
1768 
if (s>lsps == 10) { 
1769 
dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s>lsp_q_mode);

1770 
} else /* s>lsps == 16 */ 
1771 
dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s>lsp_q_mode);

1772  
1773 
for (n = 0; n < s>lsps; n++) { 
1774 
lsps[0][n] = mean_lsf[n] + (a1[n]  a2[n * 2]); 
1775 
lsps[1][n] = mean_lsf[n] + (a1[s>lsps + n]  a2[n * 2 + 1]); 
1776 
lsps[2][n] += mean_lsf[n];

1777 
} 
1778 
for (n = 0; n < 3; n++) 
1779 
stabilize_lsps(lsps[n], s>lsps); 
1780 
} 
1781  
1782 
/* Parse frames, optionally preceeded by perframe (independent) LSPs. */

1783 
for (n = 0; n < 3; n++) { 
1784 
if (!s>has_residual_lsps) {

1785 
int m;

1786  
1787 
if (s>lsps == 10) { 
1788 
dequant_lsp10i(gb, lsps[n]); 
1789 
} else /* s>lsps == 16 */ 
1790 
dequant_lsp16i(gb, lsps[n]); 
1791  
1792 
for (m = 0; m < s>lsps; m++) 
1793 
lsps[n][m] += mean_lsf[m]; 
1794 
stabilize_lsps(lsps[n], s>lsps); 
1795 
} 
1796  
1797 
if ((res = synth_frame(ctx, gb, n,

1798 
&samples[n * MAX_FRAMESIZE], 
1799 
lsps[n], n == 0 ? s>prev_lsps : lsps[n  1], 
1800 
&excitation[s>history_nsamples + n * MAX_FRAMESIZE], 
1801 
&synth[s>lsps + n * MAX_FRAMESIZE]))) 
1802 
return res;

1803 
} 
1804  
1805 
/* Statistics? FIXME  we don't check for length, a slight overrun

1806 
* will be caught by internal buffer padding, and anything else

1807 
* will be skipped, not read. */

1808 
if (get_bits1(gb)) {

1809 
res = get_bits(gb, 4);

1810 
skip_bits(gb, 10 * (res + 1)); 
1811 
} 
1812  
1813 
/* Specify nr. of output samples */

1814 
*data_size = n_samples * sizeof(float); 
1815  
1816 
/* Update history */

1817 
memcpy(s>prev_lsps, lsps[2],

1818 
s>lsps * sizeof(*s>prev_lsps));

1819 
memcpy(s>synth_history, &synth[MAX_SFRAMESIZE], 
1820 
s>lsps * sizeof(*synth));

1821 
memcpy(s>excitation_history, &excitation[MAX_SFRAMESIZE], 
1822 
s>history_nsamples * sizeof(*excitation));

1823 
if (s>do_apf)

1824 
memmove(s>zero_exc_pf, &s>zero_exc_pf[MAX_SFRAMESIZE], 
1825 
s>history_nsamples * sizeof(*s>zero_exc_pf));

1826  
1827 
return 0; 
1828 
} 
1829  
1830 
/**

1831 
* Parse the packet header at the start of each packet (input data to this

1832 
* decoder).

1833 
*

1834 
* @param s WMA Voice decoding context private data

1835 
* @return 1 if not enough bits were available, or 0 on success.

1836 
*/

1837 
static int parse_packet_header(WMAVoiceContext *s) 
1838 
{ 
1839 
GetBitContext *gb = &s>gb; 
1840 
unsigned int res; 
1841  
1842 
if (get_bits_left(gb) < 11) 
1843 
return 1; 
1844 
skip_bits(gb, 4); // packet sequence number 
1845 
s>has_residual_lsps = get_bits1(gb); 
1846 
do {

1847 
res = get_bits(gb, 6); // number of superframes per packet 
1848 
// (minus first one if there is spillover)

1849 
if (get_bits_left(gb) < 6 * (res == 0x3F) + s>spillover_bitsize) 
1850 
return 1; 
1851 
} while (res == 0x3F); 
1852 
s>spillover_nbits = get_bits(gb, s>spillover_bitsize); 
1853  
1854 
return 0; 
1855 
} 
1856  
1857 
/**

1858 
* Copy (unaligned) bits from gb/data/size to pb.

1859 
*

1860 
* @param pb target buffer to copy bits into

1861 
* @param data source buffer to copy bits from

1862 
* @param size size of the source data, in bytes

1863 
* @param gb bit I/O context specifying the current position in the source.

1864 
* data. This function might use this to align the bit position to

1865 
* a wholebyte boundary before calling #ff_copy_bits() on aligned

1866 
* source data

1867 
* @param nbits the amount of bits to copy from source to target

1868 
*

1869 
* @note after calling this function, the current position in the input bit

1870 
* I/O context is undefined.

1871 
*/

1872 
static void copy_bits(PutBitContext *pb, 
1873 
const uint8_t *data, int size, 
1874 
GetBitContext *gb, int nbits)

1875 
{ 
1876 
int rmn_bytes, rmn_bits;

1877  
1878 
rmn_bits = rmn_bytes = get_bits_left(gb); 
1879 
if (rmn_bits < nbits)

1880 
return;

1881 
rmn_bits &= 7; rmn_bytes >>= 3; 
1882 
if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) 
1883 
put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); 
1884 
ff_copy_bits(pb, data + size  rmn_bytes, 
1885 
FFMIN(nbits  rmn_bits, rmn_bytes << 3));

1886 
} 
1887  
1888 
/**

1889 
* Packet decoding: a packet is anything that the (ASF) demuxer contains,

1890 
* and we expect that the demuxer / application provides it to us as such

1891 
* (else you'll probably get garbage as output). Every packet has a size of

1892 
* ctx>block_align bytes, starts with a packet header (see

1893 
* #parse_packet_header()), and then a series of superframes. Superframe

1894 
* boundaries may exceed packets, i.e. superframes can split data over

1895 
* multiple (two) packets.

1896 
*

1897 
* For more information about frames, see #synth_superframe().

1898 
*/

1899 
static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, 
1900 
int *data_size, AVPacket *avpkt)

1901 
{ 
1902 
WMAVoiceContext *s = ctx>priv_data; 
1903 
GetBitContext *gb = &s>gb; 
1904 
int size, res, pos;

1905  
1906 
if (*data_size < 480 * sizeof(float)) { 
1907 
av_log(ctx, AV_LOG_ERROR, 
1908 
"Output buffer too small (%d given  %zu needed)\n",

1909 
*data_size, 480 * sizeof(float)); 
1910 
return 1; 
1911 
} 
1912 
*data_size = 0;

1913  
1914 
/* Packets are sometimes a multiple of ctx>block_align, with a packet

1915 
* header at each ctx>block_align bytes. However, Libav's ASF demuxer

1916 
* feeds us ASF packets, which may concatenate multiple "codec" packets

1917 
* in a single "muxer" packet, so we artificially emulate that by

1918 
* capping the packet size at ctx>block_align. */

1919 
for (size = avpkt>size; size > ctx>block_align; size = ctx>block_align);

1920 
if (!size)

1921 
return 0; 
1922 
init_get_bits(&s>gb, avpkt>data, size << 3);

1923  
1924 
/* size == ctx>block_align is used to indicate whether we are dealing with

1925 
* a new packet or a packet of which we already read the packet header

1926 
* previously. */

1927 
if (size == ctx>block_align) { // new packet header 
1928 
if ((res = parse_packet_header(s)) < 0) 
1929 
return res;

1930  
1931 
/* If the packet header specifies a s>spillover_nbits, then we want

1932 
* to push out all data of the previous packet (+ spillover) before

1933 
* continuing to parse new superframes in the current packet. */

1934 
if (s>spillover_nbits > 0) { 
1935 
if (s>sframe_cache_size > 0) { 
1936 
int cnt = get_bits_count(gb);

1937 
copy_bits(&s>pb, avpkt>data, size, gb, s>spillover_nbits); 
1938 
flush_put_bits(&s>pb); 
1939 
s>sframe_cache_size += s>spillover_nbits; 
1940 
if ((res = synth_superframe(ctx, data, data_size)) == 0 && 
1941 
*data_size > 0) {

1942 
cnt += s>spillover_nbits; 
1943 
s>skip_bits_next = cnt & 7;

1944 
return cnt >> 3; 
1945 
} else

1946 
skip_bits_long (gb, s>spillover_nbits  cnt + 
1947 
get_bits_count(gb)); // resync

1948 
} else

1949 
skip_bits_long(gb, s>spillover_nbits); // resync

1950 
} 
1951 
} else if (s>skip_bits_next) 
1952 
skip_bits(gb, s>skip_bits_next); 
1953  
1954 
/* Try parsing superframes in current packet */

1955 
s>sframe_cache_size = 0;

1956 
s>skip_bits_next = 0;

1957 
pos = get_bits_left(gb); 
1958 
if ((res = synth_superframe(ctx, data, data_size)) < 0) { 
1959 
return res;

1960 
} else if (*data_size > 0) { 
1961 
int cnt = get_bits_count(gb);

1962 
s>skip_bits_next = cnt & 7;

1963 
return cnt >> 3; 
1964 
} else if ((s>sframe_cache_size = pos) > 0) { 
1965 
/* rewind bit reader to start of last (incomplete) superframe... */

1966 
init_get_bits(gb, avpkt>data, size << 3);

1967 
skip_bits_long(gb, (size << 3)  pos);

1968 
assert(get_bits_left(gb) == pos); 
1969  
1970 
/* ...and cache it for spillover in next packet */

1971 
init_put_bits(&s>pb, s>sframe_cache, SFRAME_CACHE_MAXSIZE); 
1972 
copy_bits(&s>pb, avpkt>data, size, gb, s>sframe_cache_size); 
1973 
// FIXME bad  just copy bytes as whole and add use the

1974 
// skip_bits_next field

1975 
} 
1976  
1977 
return size;

1978 
} 
1979  
1980 
static av_cold int wmavoice_decode_end(AVCodecContext *ctx) 
1981 
{ 
1982 
WMAVoiceContext *s = ctx>priv_data; 
1983  
1984 
if (s>do_apf) {

1985 
ff_rdft_end(&s>rdft); 
1986 
ff_rdft_end(&s>irdft); 
1987 
ff_dct_end(&s>dct); 
1988 
ff_dct_end(&s>dst); 
1989 
} 
1990  
1991 
return 0; 
1992 
} 
1993  
1994 
static av_cold void wmavoice_flush(AVCodecContext *ctx) 
1995 
{ 
1996 
WMAVoiceContext *s = ctx>priv_data; 
1997 
int n;

1998  
1999 
s>postfilter_agc = 0;

2000 
s>sframe_cache_size = 0;

2001 
s>skip_bits_next = 0;

2002 
for (n = 0; n < s>lsps; n++) 
2003 
s>prev_lsps[n] = M_PI * (n + 1.0) / (s>lsps + 1.0); 
2004 
memset(s>excitation_history, 0,

2005 
sizeof(*s>excitation_history) * MAX_SIGNAL_HISTORY);

2006 
memset(s>synth_history, 0,

2007 
sizeof(*s>synth_history) * MAX_LSPS);

2008 
memset(s>gain_pred_err, 0,

2009 
sizeof(s>gain_pred_err));

2010  
2011 
if (s>do_apf) {

2012 
memset(&s>synth_filter_out_buf[MAX_LSPS_ALIGN16  s>lsps], 0,

2013 
sizeof(*s>synth_filter_out_buf) * s>lsps);

2014 
memset(s>dcf_mem, 0,

2015 
sizeof(*s>dcf_mem) * 2); 
2016 
memset(s>zero_exc_pf, 0,

2017 
sizeof(*s>zero_exc_pf) * s>history_nsamples);

2018 
memset(s>denoise_filter_cache, 0, sizeof(s>denoise_filter_cache)); 
2019 
} 
2020 
} 
2021  
2022 
AVCodec ff_wmavoice_decoder = { 
2023 
"wmavoice",

2024 
AVMEDIA_TYPE_AUDIO, 
2025 
CODEC_ID_WMAVOICE, 
2026 
sizeof(WMAVoiceContext),

2027 
wmavoice_decode_init, 
2028 
NULL,

2029 
wmavoice_decode_end, 
2030 
wmavoice_decode_packet, 
2031 
CODEC_CAP_SUBFRAMES, 
2032 
.flush = wmavoice_flush, 
2033 
.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),

2034 
}; 