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  1. /*
  2.  * Windows Media Audio Voice decoder.
  3.  * Copyright (c) 2009 Ronald S. Bultje
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * @brief Windows Media Audio Voice compatible decoder
  25.  * @author Ronald S. Bultje <rsbultje@gmail.com>
  26.  */
  27.  
  28. #include <math.h>
  29.  
  30. #include "libavutil/channel_layout.h"
  31. #include "libavutil/float_dsp.h"
  32. #include "libavutil/mem.h"
  33. #include "avcodec.h"
  34. #include "internal.h"
  35. #include "get_bits.h"
  36. #include "put_bits.h"
  37. #include "wmavoice_data.h"
  38. #include "celp_filters.h"
  39. #include "acelp_vectors.h"
  40. #include "acelp_filters.h"
  41. #include "lsp.h"
  42. #include "dct.h"
  43. #include "rdft.h"
  44. #include "sinewin.h"
  45.  
  46. #define MAX_BLOCKS           8   ///< maximum number of blocks per frame
  47. #define MAX_LSPS             16  ///< maximum filter order
  48. #define MAX_LSPS_ALIGN16     16  ///< same as #MAX_LSPS; needs to be multiple
  49.                                  ///< of 16 for ASM input buffer alignment
  50. #define MAX_FRAMES           3   ///< maximum number of frames per superframe
  51. #define MAX_FRAMESIZE        160 ///< maximum number of samples per frame
  52. #define MAX_SIGNAL_HISTORY   416 ///< maximum excitation signal history
  53. #define MAX_SFRAMESIZE       (MAX_FRAMESIZE * MAX_FRAMES)
  54.                                  ///< maximum number of samples per superframe
  55. #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
  56.                                  ///< was split over two packets
  57. #define VLC_NBITS            6   ///< number of bits to read per VLC iteration
  58.  
  59. /**
  60.  * Frame type VLC coding.
  61.  */
  62. static VLC frame_type_vlc;
  63.  
  64. /**
  65.  * Adaptive codebook types.
  66.  */
  67. enum {
  68.     ACB_TYPE_NONE       = 0, ///< no adaptive codebook (only hardcoded fixed)
  69.     ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
  70.                              ///< we interpolate to get a per-sample pitch.
  71.                              ///< Signal is generated using an asymmetric sinc
  72.                              ///< window function
  73.                              ///< @note see #wmavoice_ipol1_coeffs
  74.     ACB_TYPE_HAMMING    = 2  ///< Per-block pitch with signal generation using
  75.                              ///< a Hamming sinc window function
  76.                              ///< @note see #wmavoice_ipol2_coeffs
  77. };
  78.  
  79. /**
  80.  * Fixed codebook types.
  81.  */
  82. enum {
  83.     FCB_TYPE_SILENCE    = 0, ///< comfort noise during silence
  84.                              ///< generated from a hardcoded (fixed) codebook
  85.                              ///< with per-frame (low) gain values
  86.     FCB_TYPE_HARDCODED  = 1, ///< hardcoded (fixed) codebook with per-block
  87.                              ///< gain values
  88.     FCB_TYPE_AW_PULSES  = 2, ///< Pitch-adaptive window (AW) pulse signals,
  89.                              ///< used in particular for low-bitrate streams
  90.     FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
  91.                              ///< combinations of either single pulses or
  92.                              ///< pulse pairs
  93. };
  94.  
  95. /**
  96.  * Description of frame types.
  97.  */
  98. static const struct frame_type_desc {
  99.     uint8_t n_blocks;     ///< amount of blocks per frame (each block
  100.                           ///< (contains 160/#n_blocks samples)
  101.     uint8_t log_n_blocks; ///< log2(#n_blocks)
  102.     uint8_t acb_type;     ///< Adaptive codebook type (ACB_TYPE_*)
  103.     uint8_t fcb_type;     ///< Fixed codebook type (FCB_TYPE_*)
  104.     uint8_t dbl_pulses;   ///< how many pulse vectors have pulse pairs
  105.                           ///< (rather than just one single pulse)
  106.                           ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
  107.     uint16_t frame_size;  ///< the amount of bits that make up the block
  108.                           ///< data (per frame)
  109. } frame_descs[17] = {
  110.     { 1, 0, ACB_TYPE_NONE,       FCB_TYPE_SILENCE,    0,   0 },
  111.     { 2, 1, ACB_TYPE_NONE,       FCB_TYPE_HARDCODED,  0,  28 },
  112.     { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES,  0,  46 },
  113.     { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2,  80 },
  114.     { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
  115.     { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
  116.     { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
  117.     { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
  118.     { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0,  64 },
  119.     { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2,  80 },
  120.     { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 104 },
  121.     { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 108 },
  122.     { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 132 },
  123.     { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 168 },
  124.     { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 176 },
  125.     { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 208 },
  126.     { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 256 }
  127. };
  128.  
  129. /**
  130.  * WMA Voice decoding context.
  131.  */
  132. typedef struct {
  133.     /**
  134.      * @name Global values specified in the stream header / extradata or used all over.
  135.      * @{
  136.      */
  137.     GetBitContext gb;             ///< packet bitreader. During decoder init,
  138.                                   ///< it contains the extradata from the
  139.                                   ///< demuxer. During decoding, it contains
  140.                                   ///< packet data.
  141.     int8_t vbm_tree[25];          ///< converts VLC codes to frame type
  142.  
  143.     int spillover_bitsize;        ///< number of bits used to specify
  144.                                   ///< #spillover_nbits in the packet header
  145.                                   ///< = ceil(log2(ctx->block_align << 3))
  146.     int history_nsamples;         ///< number of samples in history for signal
  147.                                   ///< prediction (through ACB)
  148.  
  149.     /* postfilter specific values */
  150.     int do_apf;                   ///< whether to apply the averaged
  151.                                   ///< projection filter (APF)
  152.     int denoise_strength;         ///< strength of denoising in Wiener filter
  153.                                   ///< [0-11]
  154.     int denoise_tilt_corr;        ///< Whether to apply tilt correction to the
  155.                                   ///< Wiener filter coefficients (postfilter)
  156.     int dc_level;                 ///< Predicted amount of DC noise, based
  157.                                   ///< on which a DC removal filter is used
  158.  
  159.     int lsps;                     ///< number of LSPs per frame [10 or 16]
  160.     int lsp_q_mode;               ///< defines quantizer defaults [0, 1]
  161.     int lsp_def_mode;             ///< defines different sets of LSP defaults
  162.                                   ///< [0, 1]
  163.     int frame_lsp_bitsize;        ///< size (in bits) of LSPs, when encoded
  164.                                   ///< per-frame (independent coding)
  165.     int sframe_lsp_bitsize;       ///< size (in bits) of LSPs, when encoded
  166.                                   ///< per superframe (residual coding)
  167.  
  168.     int min_pitch_val;            ///< base value for pitch parsing code
  169.     int max_pitch_val;            ///< max value + 1 for pitch parsing
  170.     int pitch_nbits;              ///< number of bits used to specify the
  171.                                   ///< pitch value in the frame header
  172.     int block_pitch_nbits;        ///< number of bits used to specify the
  173.                                   ///< first block's pitch value
  174.     int block_pitch_range;        ///< range of the block pitch
  175.     int block_delta_pitch_nbits;  ///< number of bits used to specify the
  176.                                   ///< delta pitch between this and the last
  177.                                   ///< block's pitch value, used in all but
  178.                                   ///< first block
  179.     int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
  180.                                   ///< from -this to +this-1)
  181.     uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
  182.                                   ///< conversion
  183.  
  184.     /**
  185.      * @}
  186.      *
  187.      * @name Packet values specified in the packet header or related to a packet.
  188.      *
  189.      * A packet is considered to be a single unit of data provided to this
  190.      * decoder by the demuxer.
  191.      * @{
  192.      */
  193.     int spillover_nbits;          ///< number of bits of the previous packet's
  194.                                   ///< last superframe preceding this
  195.                                   ///< packet's first full superframe (useful
  196.                                   ///< for re-synchronization also)
  197.     int has_residual_lsps;        ///< if set, superframes contain one set of
  198.                                   ///< LSPs that cover all frames, encoded as
  199.                                   ///< independent and residual LSPs; if not
  200.                                   ///< set, each frame contains its own, fully
  201.                                   ///< independent, LSPs
  202.     int skip_bits_next;           ///< number of bits to skip at the next call
  203.                                   ///< to #wmavoice_decode_packet() (since
  204.                                   ///< they're part of the previous superframe)
  205.  
  206.     uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
  207.                                   ///< cache for superframe data split over
  208.                                   ///< multiple packets
  209.     int sframe_cache_size;        ///< set to >0 if we have data from an
  210.                                   ///< (incomplete) superframe from a previous
  211.                                   ///< packet that spilled over in the current
  212.                                   ///< packet; specifies the amount of bits in
  213.                                   ///< #sframe_cache
  214.     PutBitContext pb;             ///< bitstream writer for #sframe_cache
  215.  
  216.     /**
  217.      * @}
  218.      *
  219.      * @name Frame and superframe values
  220.      * Superframe and frame data - these can change from frame to frame,
  221.      * although some of them do in that case serve as a cache / history for
  222.      * the next frame or superframe.
  223.      * @{
  224.      */
  225.     double prev_lsps[MAX_LSPS];   ///< LSPs of the last frame of the previous
  226.                                   ///< superframe
  227.     int last_pitch_val;           ///< pitch value of the previous frame
  228.     int last_acb_type;            ///< frame type [0-2] of the previous frame
  229.     int pitch_diff_sh16;          ///< ((cur_pitch_val - #last_pitch_val)
  230.                                   ///< << 16) / #MAX_FRAMESIZE
  231.     float silence_gain;           ///< set for use in blocks if #ACB_TYPE_NONE
  232.  
  233.     int aw_idx_is_ext;            ///< whether the AW index was encoded in
  234.                                   ///< 8 bits (instead of 6)
  235.     int aw_pulse_range;           ///< the range over which #aw_pulse_set1()
  236.                                   ///< can apply the pulse, relative to the
  237.                                   ///< value in aw_first_pulse_off. The exact
  238.                                   ///< position of the first AW-pulse is within
  239.                                   ///< [pulse_off, pulse_off + this], and
  240.                                   ///< depends on bitstream values; [16 or 24]
  241.     int aw_n_pulses[2];           ///< number of AW-pulses in each block; note
  242.                                   ///< that this number can be negative (in
  243.                                   ///< which case it basically means "zero")
  244.     int aw_first_pulse_off[2];    ///< index of first sample to which to
  245.                                   ///< apply AW-pulses, or -0xff if unset
  246.     int aw_next_pulse_off_cache;  ///< the position (relative to start of the
  247.                                   ///< second block) at which pulses should
  248.                                   ///< start to be positioned, serves as a
  249.                                   ///< cache for pitch-adaptive window pulses
  250.                                   ///< between blocks
  251.  
  252.     int frame_cntr;               ///< current frame index [0 - 0xFFFE]; is
  253.                                   ///< only used for comfort noise in #pRNG()
  254.     float gain_pred_err[6];       ///< cache for gain prediction
  255.     float excitation_history[MAX_SIGNAL_HISTORY];
  256.                                   ///< cache of the signal of previous
  257.                                   ///< superframes, used as a history for
  258.                                   ///< signal generation
  259.     float synth_history[MAX_LSPS]; ///< see #excitation_history
  260.     /**
  261.      * @}
  262.      *
  263.      * @name Postfilter values
  264.      *
  265.      * Variables used for postfilter implementation, mostly history for
  266.      * smoothing and so on, and context variables for FFT/iFFT.
  267.      * @{
  268.      */
  269.     RDFTContext rdft, irdft;      ///< contexts for FFT-calculation in the
  270.                                   ///< postfilter (for denoise filter)
  271.     DCTContext dct, dst;          ///< contexts for phase shift (in Hilbert
  272.                                   ///< transform, part of postfilter)
  273.     float sin[511], cos[511];     ///< 8-bit cosine/sine windows over [-pi,pi]
  274.                                   ///< range
  275.     float postfilter_agc;         ///< gain control memory, used in
  276.                                   ///< #adaptive_gain_control()
  277.     float dcf_mem[2];             ///< DC filter history
  278.     float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
  279.                                   ///< zero filter output (i.e. excitation)
  280.                                   ///< by postfilter
  281.     float denoise_filter_cache[MAX_FRAMESIZE];
  282.     int   denoise_filter_cache_size; ///< samples in #denoise_filter_cache
  283.     DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
  284.                                   ///< aligned buffer for LPC tilting
  285.     DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
  286.                                   ///< aligned buffer for denoise coefficients
  287.     DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
  288.                                   ///< aligned buffer for postfilter speech
  289.                                   ///< synthesis
  290.     /**
  291.      * @}
  292.      */
  293. } WMAVoiceContext;
  294.  
  295. /**
  296.  * Set up the variable bit mode (VBM) tree from container extradata.
  297.  * @param gb bit I/O context.
  298.  *           The bit context (s->gb) should be loaded with byte 23-46 of the
  299.  *           container extradata (i.e. the ones containing the VBM tree).
  300.  * @param vbm_tree pointer to array to which the decoded VBM tree will be
  301.  *                 written.
  302.  * @return 0 on success, <0 on error.
  303.  */
  304. static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
  305. {
  306.     int cntr[8] = { 0 }, n, res;
  307.  
  308.     memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
  309.     for (n = 0; n < 17; n++) {
  310.         res = get_bits(gb, 3);
  311.         if (cntr[res] > 3) // should be >= 3 + (res == 7))
  312.             return -1;
  313.         vbm_tree[res * 3 + cntr[res]++] = n;
  314.     }
  315.     return 0;
  316. }
  317.  
  318. static av_cold void wmavoice_init_static_data(AVCodec *codec)
  319. {
  320.     static const uint8_t bits[] = {
  321.          2,  2,  2,  4,  4,  4,
  322.          6,  6,  6,  8,  8,  8,
  323.         10, 10, 10, 12, 12, 12,
  324.         14, 14, 14, 14
  325.     };
  326.     static const uint16_t codes[] = {
  327.           0x0000, 0x0001, 0x0002,        //              00/01/10
  328.           0x000c, 0x000d, 0x000e,        //           11+00/01/10
  329.           0x003c, 0x003d, 0x003e,        //         1111+00/01/10
  330.           0x00fc, 0x00fd, 0x00fe,        //       111111+00/01/10
  331.           0x03fc, 0x03fd, 0x03fe,        //     11111111+00/01/10
  332.           0x0ffc, 0x0ffd, 0x0ffe,        //   1111111111+00/01/10
  333.           0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
  334.     };
  335.  
  336.     INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
  337.                     bits, 1, 1, codes, 2, 2, 132);
  338. }
  339.  
  340. /**
  341.  * Set up decoder with parameters from demuxer (extradata etc.).
  342.  */
  343. static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
  344. {
  345.     int n, flags, pitch_range, lsp16_flag;
  346.     WMAVoiceContext *s = ctx->priv_data;
  347.  
  348.     /**
  349.      * Extradata layout:
  350.      * - byte  0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
  351.      * - byte 19-22: flags field (annoyingly in LE; see below for known
  352.      *               values),
  353.      * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
  354.      *               rest is 0).
  355.      */
  356.     if (ctx->extradata_size != 46) {
  357.         av_log(ctx, AV_LOG_ERROR,
  358.                "Invalid extradata size %d (should be 46)\n",
  359.                ctx->extradata_size);
  360.         return AVERROR_INVALIDDATA;
  361.     }
  362.     flags                = AV_RL32(ctx->extradata + 18);
  363.     s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
  364.     s->do_apf            =    flags & 0x1;
  365.     if (s->do_apf) {
  366.         ff_rdft_init(&s->rdft,  7, DFT_R2C);
  367.         ff_rdft_init(&s->irdft, 7, IDFT_C2R);
  368.         ff_dct_init(&s->dct,  6, DCT_I);
  369.         ff_dct_init(&s->dst,  6, DST_I);
  370.  
  371.         ff_sine_window_init(s->cos, 256);
  372.         memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
  373.         for (n = 0; n < 255; n++) {
  374.             s->sin[n]       = -s->sin[510 - n];
  375.             s->cos[510 - n] =  s->cos[n];
  376.         }
  377.     }
  378.     s->denoise_strength  =   (flags >> 2) & 0xF;
  379.     if (s->denoise_strength >= 12) {
  380.         av_log(ctx, AV_LOG_ERROR,
  381.                "Invalid denoise filter strength %d (max=11)\n",
  382.                s->denoise_strength);
  383.         return AVERROR_INVALIDDATA;
  384.     }
  385.     s->denoise_tilt_corr = !!(flags & 0x40);
  386.     s->dc_level          =   (flags >> 7) & 0xF;
  387.     s->lsp_q_mode        = !!(flags & 0x2000);
  388.     s->lsp_def_mode      = !!(flags & 0x4000);
  389.     lsp16_flag           =    flags & 0x1000;
  390.     if (lsp16_flag) {
  391.         s->lsps               = 16;
  392.         s->frame_lsp_bitsize  = 34;
  393.         s->sframe_lsp_bitsize = 60;
  394.     } else {
  395.         s->lsps               = 10;
  396.         s->frame_lsp_bitsize  = 24;
  397.         s->sframe_lsp_bitsize = 48;
  398.     }
  399.     for (n = 0; n < s->lsps; n++)
  400.         s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
  401.  
  402.     init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
  403.     if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
  404.         av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
  405.         return AVERROR_INVALIDDATA;
  406.     }
  407.  
  408.     s->min_pitch_val    = ((ctx->sample_rate << 8)      /  400 + 50) >> 8;
  409.     s->max_pitch_val    = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
  410.     pitch_range         = s->max_pitch_val - s->min_pitch_val;
  411.     if (pitch_range <= 0) {
  412.         av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
  413.         return AVERROR_INVALIDDATA;
  414.     }
  415.     s->pitch_nbits      = av_ceil_log2(pitch_range);
  416.     s->last_pitch_val   = 40;
  417.     s->last_acb_type    = ACB_TYPE_NONE;
  418.     s->history_nsamples = s->max_pitch_val + 8;
  419.  
  420.     if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
  421.         int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
  422.             max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
  423.  
  424.         av_log(ctx, AV_LOG_ERROR,
  425.                "Unsupported samplerate %d (min=%d, max=%d)\n",
  426.                ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
  427.  
  428.         return AVERROR(ENOSYS);
  429.     }
  430.  
  431.     s->block_conv_table[0]      = s->min_pitch_val;
  432.     s->block_conv_table[1]      = (pitch_range * 25) >> 6;
  433.     s->block_conv_table[2]      = (pitch_range * 44) >> 6;
  434.     s->block_conv_table[3]      = s->max_pitch_val - 1;
  435.     s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
  436.     if (s->block_delta_pitch_hrange <= 0) {
  437.         av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
  438.         return AVERROR_INVALIDDATA;
  439.     }
  440.     s->block_delta_pitch_nbits  = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
  441.     s->block_pitch_range        = s->block_conv_table[2] +
  442.                                   s->block_conv_table[3] + 1 +
  443.                                   2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
  444.     s->block_pitch_nbits        = av_ceil_log2(s->block_pitch_range);
  445.  
  446.     ctx->channels               = 1;
  447.     ctx->channel_layout         = AV_CH_LAYOUT_MONO;
  448.     ctx->sample_fmt             = AV_SAMPLE_FMT_FLT;
  449.  
  450.     return 0;
  451. }
  452.  
  453. /**
  454.  * @name Postfilter functions
  455.  * Postfilter functions (gain control, wiener denoise filter, DC filter,
  456.  * kalman smoothening, plus surrounding code to wrap it)
  457.  * @{
  458.  */
  459. /**
  460.  * Adaptive gain control (as used in postfilter).
  461.  *
  462.  * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
  463.  * that the energy here is calculated using sum(abs(...)), whereas the
  464.  * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
  465.  *
  466.  * @param out output buffer for filtered samples
  467.  * @param in input buffer containing the samples as they are after the
  468.  *           postfilter steps so far
  469.  * @param speech_synth input buffer containing speech synth before postfilter
  470.  * @param size input buffer size
  471.  * @param alpha exponential filter factor
  472.  * @param gain_mem pointer to filter memory (single float)
  473.  */
  474. static void adaptive_gain_control(float *out, const float *in,
  475.                                   const float *speech_synth,
  476.                                   int size, float alpha, float *gain_mem)
  477. {
  478.     int i;
  479.     float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
  480.     float mem = *gain_mem;
  481.  
  482.     for (i = 0; i < size; i++) {
  483.         speech_energy     += fabsf(speech_synth[i]);
  484.         postfilter_energy += fabsf(in[i]);
  485.     }
  486.     gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
  487.  
  488.     for (i = 0; i < size; i++) {
  489.         mem = alpha * mem + gain_scale_factor;
  490.         out[i] = in[i] * mem;
  491.     }
  492.  
  493.     *gain_mem = mem;
  494. }
  495.  
  496. /**
  497.  * Kalman smoothing function.
  498.  *
  499.  * This function looks back pitch +/- 3 samples back into history to find
  500.  * the best fitting curve (that one giving the optimal gain of the two
  501.  * signals, i.e. the highest dot product between the two), and then
  502.  * uses that signal history to smoothen the output of the speech synthesis
  503.  * filter.
  504.  *
  505.  * @param s WMA Voice decoding context
  506.  * @param pitch pitch of the speech signal
  507.  * @param in input speech signal
  508.  * @param out output pointer for smoothened signal
  509.  * @param size input/output buffer size
  510.  *
  511.  * @returns -1 if no smoothening took place, e.g. because no optimal
  512.  *          fit could be found, or 0 on success.
  513.  */
  514. static int kalman_smoothen(WMAVoiceContext *s, int pitch,
  515.                            const float *in, float *out, int size)
  516. {
  517.     int n;
  518.     float optimal_gain = 0, dot;
  519.     const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
  520.                 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
  521.                 *best_hist_ptr = NULL;
  522.  
  523.     /* find best fitting point in history */
  524.     do {
  525.         dot = avpriv_scalarproduct_float_c(in, ptr, size);
  526.         if (dot > optimal_gain) {
  527.             optimal_gain  = dot;
  528.             best_hist_ptr = ptr;
  529.         }
  530.     } while (--ptr >= end);
  531.  
  532.     if (optimal_gain <= 0)
  533.         return -1;
  534.     dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
  535.     if (dot <= 0) // would be 1.0
  536.         return -1;
  537.  
  538.     if (optimal_gain <= dot) {
  539.         dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
  540.     } else
  541.         dot = 0.625;
  542.  
  543.     /* actual smoothing */
  544.     for (n = 0; n < size; n++)
  545.         out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
  546.  
  547.     return 0;
  548. }
  549.  
  550. /**
  551.  * Get the tilt factor of a formant filter from its transfer function
  552.  * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
  553.  *      but somehow (??) it does a speech synthesis filter in the
  554.  *      middle, which is missing here
  555.  *
  556.  * @param lpcs LPC coefficients
  557.  * @param n_lpcs Size of LPC buffer
  558.  * @returns the tilt factor
  559.  */
  560. static float tilt_factor(const float *lpcs, int n_lpcs)
  561. {
  562.     float rh0, rh1;
  563.  
  564.     rh0 = 1.0     + avpriv_scalarproduct_float_c(lpcs,  lpcs,    n_lpcs);
  565.     rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
  566.  
  567.     return rh1 / rh0;
  568. }
  569.  
  570. /**
  571.  * Derive denoise filter coefficients (in real domain) from the LPCs.
  572.  */
  573. static void calc_input_response(WMAVoiceContext *s, float *lpcs,
  574.                                 int fcb_type, float *coeffs, int remainder)
  575. {
  576.     float last_coeff, min = 15.0, max = -15.0;
  577.     float irange, angle_mul, gain_mul, range, sq;
  578.     int n, idx;
  579.  
  580.     /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
  581.     s->rdft.rdft_calc(&s->rdft, lpcs);
  582. #define log_range(var, assign) do { \
  583.         float tmp = log10f(assign);  var = tmp; \
  584.         max       = FFMAX(max, tmp); min = FFMIN(min, tmp); \
  585.     } while (0)
  586.     log_range(last_coeff,  lpcs[1]         * lpcs[1]);
  587.     for (n = 1; n < 64; n++)
  588.         log_range(lpcs[n], lpcs[n * 2]     * lpcs[n * 2] +
  589.                            lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
  590.     log_range(lpcs[0],     lpcs[0]         * lpcs[0]);
  591. #undef log_range
  592.     range    = max - min;
  593.     lpcs[64] = last_coeff;
  594.  
  595.     /* Now, use this spectrum to pick out these frequencies with higher
  596.      * (relative) power/energy (which we then take to be "not noise"),
  597.      * and set up a table (still in lpc[]) of (relative) gains per frequency.
  598.      * These frequencies will be maintained, while others ("noise") will be
  599.      * decreased in the filter output. */
  600.     irange    = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
  601.     gain_mul  = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
  602.                                                           (5.0 / 14.7));
  603.     angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
  604.     for (n = 0; n <= 64; n++) {
  605.         float pwr;
  606.  
  607.         idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
  608.         pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
  609.         lpcs[n] = angle_mul * pwr;
  610.  
  611.         /* 70.57 =~ 1/log10(1.0331663) */
  612.         idx = (pwr * gain_mul - 0.0295) * 70.570526123;
  613.         if (idx > 127) { // fall back if index falls outside table range
  614.             coeffs[n] = wmavoice_energy_table[127] *
  615.                         powf(1.0331663, idx - 127);
  616.         } else
  617.             coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
  618.     }
  619.  
  620.     /* calculate the Hilbert transform of the gains, which we do (since this
  621.      * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
  622.      * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
  623.      * "moment" of the LPCs in this filter. */
  624.     s->dct.dct_calc(&s->dct, lpcs);
  625.     s->dst.dct_calc(&s->dst, lpcs);
  626.  
  627.     /* Split out the coefficient indexes into phase/magnitude pairs */
  628.     idx = 255 + av_clip(lpcs[64],               -255, 255);
  629.     coeffs[0]  = coeffs[0]  * s->cos[idx];
  630.     idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
  631.     last_coeff = coeffs[64] * s->cos[idx];
  632.     for (n = 63;; n--) {
  633.         idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
  634.         coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
  635.         coeffs[n * 2]     = coeffs[n] * s->cos[idx];
  636.  
  637.         if (!--n) break;
  638.  
  639.         idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
  640.         coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
  641.         coeffs[n * 2]     = coeffs[n] * s->cos[idx];
  642.     }
  643.     coeffs[1] = last_coeff;
  644.  
  645.     /* move into real domain */
  646.     s->irdft.rdft_calc(&s->irdft, coeffs);
  647.  
  648.     /* tilt correction and normalize scale */
  649.     memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
  650.     if (s->denoise_tilt_corr) {
  651.         float tilt_mem = 0;
  652.  
  653.         coeffs[remainder - 1] = 0;
  654.         ff_tilt_compensation(&tilt_mem,
  655.                              -1.8 * tilt_factor(coeffs, remainder - 1),
  656.                              coeffs, remainder);
  657.     }
  658.     sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
  659.                                                                remainder));
  660.     for (n = 0; n < remainder; n++)
  661.         coeffs[n] *= sq;
  662. }
  663.  
  664. /**
  665.  * This function applies a Wiener filter on the (noisy) speech signal as
  666.  * a means to denoise it.
  667.  *
  668.  * - take RDFT of LPCs to get the power spectrum of the noise + speech;
  669.  * - using this power spectrum, calculate (for each frequency) the Wiener
  670.  *    filter gain, which depends on the frequency power and desired level
  671.  *    of noise subtraction (when set too high, this leads to artifacts)
  672.  *    We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
  673.  *    of 4-8kHz);
  674.  * - by doing a phase shift, calculate the Hilbert transform of this array
  675.  *    of per-frequency filter-gains to get the filtering coefficients;
  676.  * - smoothen/normalize/de-tilt these filter coefficients as desired;
  677.  * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
  678.  *    to get the denoised speech signal;
  679.  * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
  680.  *    the frame boundary) are saved and applied to subsequent frames by an
  681.  *    overlap-add method (otherwise you get clicking-artifacts).
  682.  *
  683.  * @param s WMA Voice decoding context
  684.  * @param fcb_type Frame (codebook) type
  685.  * @param synth_pf input: the noisy speech signal, output: denoised speech
  686.  *                 data; should be 16-byte aligned (for ASM purposes)
  687.  * @param size size of the speech data
  688.  * @param lpcs LPCs used to synthesize this frame's speech data
  689.  */
  690. static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
  691.                            float *synth_pf, int size,
  692.                            const float *lpcs)
  693. {
  694.     int remainder, lim, n;
  695.  
  696.     if (fcb_type != FCB_TYPE_SILENCE) {
  697.         float *tilted_lpcs = s->tilted_lpcs_pf,
  698.               *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
  699.  
  700.         tilted_lpcs[0]           = 1.0;
  701.         memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
  702.         memset(&tilted_lpcs[s->lsps + 1], 0,
  703.                sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
  704.         ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
  705.                              tilted_lpcs, s->lsps + 2);
  706.  
  707.         /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
  708.          * size is applied to the next frame. All input beyond this is zero,
  709.          * and thus all output beyond this will go towards zero, hence we can
  710.          * limit to min(size-1, 127-size) as a performance consideration. */
  711.         remainder = FFMIN(127 - size, size - 1);
  712.         calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
  713.  
  714.         /* apply coefficients (in frequency spectrum domain), i.e. complex
  715.          * number multiplication */
  716.         memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
  717.         s->rdft.rdft_calc(&s->rdft, synth_pf);
  718.         s->rdft.rdft_calc(&s->rdft, coeffs);
  719.         synth_pf[0] *= coeffs[0];
  720.         synth_pf[1] *= coeffs[1];
  721.         for (n = 1; n < 64; n++) {
  722.             float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
  723.             synth_pf[n * 2]     = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
  724.             synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
  725.         }
  726.         s->irdft.rdft_calc(&s->irdft, synth_pf);
  727.     }
  728.  
  729.     /* merge filter output with the history of previous runs */
  730.     if (s->denoise_filter_cache_size) {
  731.         lim = FFMIN(s->denoise_filter_cache_size, size);
  732.         for (n = 0; n < lim; n++)
  733.             synth_pf[n] += s->denoise_filter_cache[n];
  734.         s->denoise_filter_cache_size -= lim;
  735.         memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
  736.                 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
  737.     }
  738.  
  739.     /* move remainder of filter output into a cache for future runs */
  740.     if (fcb_type != FCB_TYPE_SILENCE) {
  741.         lim = FFMIN(remainder, s->denoise_filter_cache_size);
  742.         for (n = 0; n < lim; n++)
  743.             s->denoise_filter_cache[n] += synth_pf[size + n];
  744.         if (lim < remainder) {
  745.             memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
  746.                    sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
  747.             s->denoise_filter_cache_size = remainder;
  748.         }
  749.     }
  750. }
  751.  
  752. /**
  753.  * Averaging projection filter, the postfilter used in WMAVoice.
  754.  *
  755.  * This uses the following steps:
  756.  * - A zero-synthesis filter (generate excitation from synth signal)
  757.  * - Kalman smoothing on excitation, based on pitch
  758.  * - Re-synthesized smoothened output
  759.  * - Iterative Wiener denoise filter
  760.  * - Adaptive gain filter
  761.  * - DC filter
  762.  *
  763.  * @param s WMAVoice decoding context
  764.  * @param synth Speech synthesis output (before postfilter)
  765.  * @param samples Output buffer for filtered samples
  766.  * @param size Buffer size of synth & samples
  767.  * @param lpcs Generated LPCs used for speech synthesis
  768.  * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
  769.  * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
  770.  * @param pitch Pitch of the input signal
  771.  */
  772. static void postfilter(WMAVoiceContext *s, const float *synth,
  773.                        float *samples,    int size,
  774.                        const float *lpcs, float *zero_exc_pf,
  775.                        int fcb_type,      int pitch)
  776. {
  777.     float synth_filter_in_buf[MAX_FRAMESIZE / 2],
  778.           *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
  779.           *synth_filter_in = zero_exc_pf;
  780.  
  781.     av_assert0(size <= MAX_FRAMESIZE / 2);
  782.  
  783.     /* generate excitation from input signal */
  784.     ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
  785.  
  786.     if (fcb_type >= FCB_TYPE_AW_PULSES &&
  787.         !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
  788.         synth_filter_in = synth_filter_in_buf;
  789.  
  790.     /* re-synthesize speech after smoothening, and keep history */
  791.     ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
  792.                                  synth_filter_in, size, s->lsps);
  793.     memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
  794.            sizeof(synth_pf[0]) * s->lsps);
  795.  
  796.     wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
  797.  
  798.     adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
  799.                           &s->postfilter_agc);
  800.  
  801.     if (s->dc_level > 8) {
  802.         /* remove ultra-low frequency DC noise / highpass filter;
  803.          * coefficients are identical to those used in SIPR decoding,
  804.          * and very closely resemble those used in AMR-NB decoding. */
  805.         ff_acelp_apply_order_2_transfer_function(samples, samples,
  806.             (const float[2]) { -1.99997,      1.0 },
  807.             (const float[2]) { -1.9330735188, 0.93589198496 },
  808.             0.93980580475, s->dcf_mem, size);
  809.     }
  810. }
  811. /**
  812.  * @}
  813.  */
  814.  
  815. /**
  816.  * Dequantize LSPs
  817.  * @param lsps output pointer to the array that will hold the LSPs
  818.  * @param num number of LSPs to be dequantized
  819.  * @param values quantized values, contains n_stages values
  820.  * @param sizes range (i.e. max value) of each quantized value
  821.  * @param n_stages number of dequantization runs
  822.  * @param table dequantization table to be used
  823.  * @param mul_q LSF multiplier
  824.  * @param base_q base (lowest) LSF values
  825.  */
  826. static void dequant_lsps(double *lsps, int num,
  827.                          const uint16_t *values,
  828.                          const uint16_t *sizes,
  829.                          int n_stages, const uint8_t *table,
  830.                          const double *mul_q,
  831.                          const double *base_q)
  832. {
  833.     int n, m;
  834.  
  835.     memset(lsps, 0, num * sizeof(*lsps));
  836.     for (n = 0; n < n_stages; n++) {
  837.         const uint8_t *t_off = &table[values[n] * num];
  838.         double base = base_q[n], mul = mul_q[n];
  839.  
  840.         for (m = 0; m < num; m++)
  841.             lsps[m] += base + mul * t_off[m];
  842.  
  843.         table += sizes[n] * num;
  844.     }
  845. }
  846.  
  847. /**
  848.  * @name LSP dequantization routines
  849.  * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
  850.  * @note we assume enough bits are available, caller should check.
  851.  * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
  852.  * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
  853.  * @{
  854.  */
  855. /**
  856.  * Parse 10 independently-coded LSPs.
  857.  */
  858. static void dequant_lsp10i(GetBitContext *gb, double *lsps)
  859. {
  860.     static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
  861.     static const double mul_lsf[4] = {
  862.         5.2187144800e-3,    1.4626986422e-3,
  863.         9.6179549166e-4,    1.1325736225e-3
  864.     };
  865.     static const double base_lsf[4] = {
  866.         M_PI * -2.15522e-1, M_PI * -6.1646e-2,
  867.         M_PI * -3.3486e-2,  M_PI * -5.7408e-2
  868.     };
  869.     uint16_t v[4];
  870.  
  871.     v[0] = get_bits(gb, 8);
  872.     v[1] = get_bits(gb, 6);
  873.     v[2] = get_bits(gb, 5);
  874.     v[3] = get_bits(gb, 5);
  875.  
  876.     dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
  877.                  mul_lsf, base_lsf);
  878. }
  879.  
  880. /**
  881.  * Parse 10 independently-coded LSPs, and then derive the tables to
  882.  * generate LSPs for the other frames from them (residual coding).
  883.  */
  884. static void dequant_lsp10r(GetBitContext *gb,
  885.                            double *i_lsps, const double *old,
  886.                            double *a1, double *a2, int q_mode)
  887. {
  888.     static const uint16_t vec_sizes[3] = { 128, 64, 64 };
  889.     static const double mul_lsf[3] = {
  890.         2.5807601174e-3,    1.2354460219e-3,   1.1763821673e-3
  891.     };
  892.     static const double base_lsf[3] = {
  893.         M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
  894.     };
  895.     const float (*ipol_tab)[2][10] = q_mode ?
  896.         wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
  897.     uint16_t interpol, v[3];
  898.     int n;
  899.  
  900.     dequant_lsp10i(gb, i_lsps);
  901.  
  902.     interpol = get_bits(gb, 5);
  903.     v[0]     = get_bits(gb, 7);
  904.     v[1]     = get_bits(gb, 6);
  905.     v[2]     = get_bits(gb, 6);
  906.  
  907.     for (n = 0; n < 10; n++) {
  908.         double delta = old[n] - i_lsps[n];
  909.         a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
  910.         a1[10 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
  911.     }
  912.  
  913.     dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
  914.                  mul_lsf, base_lsf);
  915. }
  916.  
  917. /**
  918.  * Parse 16 independently-coded LSPs.
  919.  */
  920. static void dequant_lsp16i(GetBitContext *gb, double *lsps)
  921. {
  922.     static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
  923.     static const double mul_lsf[5] = {
  924.         3.3439586280e-3,    6.9908173703e-4,
  925.         3.3216608306e-3,    1.0334960326e-3,
  926.         3.1899104283e-3
  927.     };
  928.     static const double base_lsf[5] = {
  929.         M_PI * -1.27576e-1, M_PI * -2.4292e-2,
  930.         M_PI * -1.28094e-1, M_PI * -3.2128e-2,
  931.         M_PI * -1.29816e-1
  932.     };
  933.     uint16_t v[5];
  934.  
  935.     v[0] = get_bits(gb, 8);
  936.     v[1] = get_bits(gb, 6);
  937.     v[2] = get_bits(gb, 7);
  938.     v[3] = get_bits(gb, 6);
  939.     v[4] = get_bits(gb, 7);
  940.  
  941.     dequant_lsps( lsps,     5,  v,     vec_sizes,    2,
  942.                  wmavoice_dq_lsp16i1,  mul_lsf,     base_lsf);
  943.     dequant_lsps(&lsps[5],  5, &v[2], &vec_sizes[2], 2,
  944.                  wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
  945.     dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
  946.                  wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
  947. }
  948.  
  949. /**
  950.  * Parse 16 independently-coded LSPs, and then derive the tables to
  951.  * generate LSPs for the other frames from them (residual coding).
  952.  */
  953. static void dequant_lsp16r(GetBitContext *gb,
  954.                            double *i_lsps, const double *old,
  955.                            double *a1, double *a2, int q_mode)
  956. {
  957.     static const uint16_t vec_sizes[3] = { 128, 128, 128 };
  958.     static const double mul_lsf[3] = {
  959.         1.2232979501e-3,   1.4062241527e-3,   1.6114744851e-3
  960.     };
  961.     static const double base_lsf[3] = {
  962.         M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
  963.     };
  964.     const float (*ipol_tab)[2][16] = q_mode ?
  965.         wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
  966.     uint16_t interpol, v[3];
  967.     int n;
  968.  
  969.     dequant_lsp16i(gb, i_lsps);
  970.  
  971.     interpol = get_bits(gb, 5);
  972.     v[0]     = get_bits(gb, 7);
  973.     v[1]     = get_bits(gb, 7);
  974.     v[2]     = get_bits(gb, 7);
  975.  
  976.     for (n = 0; n < 16; n++) {
  977.         double delta = old[n] - i_lsps[n];
  978.         a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
  979.         a1[16 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
  980.     }
  981.  
  982.     dequant_lsps( a2,     10,  v,     vec_sizes,    1,
  983.                  wmavoice_dq_lsp16r1,  mul_lsf,     base_lsf);
  984.     dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
  985.                  wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
  986.     dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
  987.                  wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
  988. }
  989.  
  990. /**
  991.  * @}
  992.  * @name Pitch-adaptive window coding functions
  993.  * The next few functions are for pitch-adaptive window coding.
  994.  * @{
  995.  */
  996. /**
  997.  * Parse the offset of the first pitch-adaptive window pulses, and
  998.  * the distribution of pulses between the two blocks in this frame.
  999.  * @param s WMA Voice decoding context private data
  1000.  * @param gb bit I/O context
  1001.  * @param pitch pitch for each block in this frame
  1002.  */
  1003. static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
  1004.                             const int *pitch)
  1005. {
  1006.     static const int16_t start_offset[94] = {
  1007.         -11,  -9,  -7,  -5,  -3,  -1,   1,   3,   5,   7,   9,  11,
  1008.          13,  15,  18,  17,  19,  20,  21,  22,  23,  24,  25,  26,
  1009.          27,  28,  29,  30,  31,  32,  33,  35,  37,  39,  41,  43,
  1010.          45,  47,  49,  51,  53,  55,  57,  59,  61,  63,  65,  67,
  1011.          69,  71,  73,  75,  77,  79,  81,  83,  85,  87,  89,  91,
  1012.          93,  95,  97,  99, 101, 103, 105, 107, 109, 111, 113, 115,
  1013.         117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
  1014.         141, 143, 145, 147, 149, 151, 153, 155, 157, 159
  1015.     };
  1016.     int bits, offset;
  1017.  
  1018.     /* position of pulse */
  1019.     s->aw_idx_is_ext = 0;
  1020.     if ((bits = get_bits(gb, 6)) >= 54) {
  1021.         s->aw_idx_is_ext = 1;
  1022.         bits += (bits - 54) * 3 + get_bits(gb, 2);
  1023.     }
  1024.  
  1025.     /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
  1026.      * the distribution of the pulses in each block contained in this frame. */
  1027.     s->aw_pulse_range        = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
  1028.     for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
  1029.     s->aw_n_pulses[0]        = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
  1030.     s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
  1031.     offset                  += s->aw_n_pulses[0] * pitch[0];
  1032.     s->aw_n_pulses[1]        = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
  1033.     s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
  1034.  
  1035.     /* if continuing from a position before the block, reset position to
  1036.      * start of block (when corrected for the range over which it can be
  1037.      * spread in aw_pulse_set1()). */
  1038.     if (start_offset[bits] < MAX_FRAMESIZE / 2) {
  1039.         while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
  1040.             s->aw_first_pulse_off[1] -= pitch[1];
  1041.         if (start_offset[bits] < 0)
  1042.             while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
  1043.                 s->aw_first_pulse_off[0] -= pitch[0];
  1044.     }
  1045. }
  1046.  
  1047. /**
  1048.  * Apply second set of pitch-adaptive window pulses.
  1049.  * @param s WMA Voice decoding context private data
  1050.  * @param gb bit I/O context
  1051.  * @param block_idx block index in frame [0, 1]
  1052.  * @param fcb structure containing fixed codebook vector info
  1053.  * @return -1 on error, 0 otherwise
  1054.  */
  1055. static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
  1056.                          int block_idx, AMRFixed *fcb)
  1057. {
  1058.     uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
  1059.     uint16_t *use_mask = use_mask_mem + 2;
  1060.     /* in this function, idx is the index in the 80-bit (+ padding) use_mask
  1061.      * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
  1062.      * of idx are the position of the bit within a particular item in the
  1063.      * array (0 being the most significant bit, and 15 being the least
  1064.      * significant bit), and the remainder (>> 4) is the index in the
  1065.      * use_mask[]-array. This is faster and uses less memory than using a
  1066.      * 80-byte/80-int array. */
  1067.     int pulse_off = s->aw_first_pulse_off[block_idx],
  1068.         pulse_start, n, idx, range, aidx, start_off = 0;
  1069.  
  1070.     /* set offset of first pulse to within this block */
  1071.     if (s->aw_n_pulses[block_idx] > 0)
  1072.         while (pulse_off + s->aw_pulse_range < 1)
  1073.             pulse_off += fcb->pitch_lag;
  1074.  
  1075.     /* find range per pulse */
  1076.     if (s->aw_n_pulses[0] > 0) {
  1077.         if (block_idx == 0) {
  1078.             range = 32;
  1079.         } else /* block_idx = 1 */ {
  1080.             range = 8;
  1081.             if (s->aw_n_pulses[block_idx] > 0)
  1082.                 pulse_off = s->aw_next_pulse_off_cache;
  1083.         }
  1084.     } else
  1085.         range = 16;
  1086.     pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
  1087.  
  1088.     /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
  1089.      * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
  1090.      * we exclude that range from being pulsed again in this function. */
  1091.     memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
  1092.     memset( use_mask,   -1, 5 * sizeof(use_mask[0]));
  1093.     memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
  1094.     if (s->aw_n_pulses[block_idx] > 0)
  1095.         for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
  1096.             int excl_range         = s->aw_pulse_range; // always 16 or 24
  1097.             uint16_t *use_mask_ptr = &use_mask[idx >> 4];
  1098.             int first_sh           = 16 - (idx & 15);
  1099.             *use_mask_ptr++       &= 0xFFFFu << first_sh;
  1100.             excl_range            -= first_sh;
  1101.             if (excl_range >= 16) {
  1102.                 *use_mask_ptr++    = 0;
  1103.                 *use_mask_ptr     &= 0xFFFF >> (excl_range - 16);
  1104.             } else
  1105.                 *use_mask_ptr     &= 0xFFFF >> excl_range;
  1106.         }
  1107.  
  1108.     /* find the 'aidx'th offset that is not excluded */
  1109.     aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
  1110.     for (n = 0; n <= aidx; pulse_start++) {
  1111.         for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
  1112.         if (idx >= MAX_FRAMESIZE / 2) { // find from zero
  1113.             if (use_mask[0])      idx = 0x0F;
  1114.             else if (use_mask[1]) idx = 0x1F;
  1115.             else if (use_mask[2]) idx = 0x2F;
  1116.             else if (use_mask[3]) idx = 0x3F;
  1117.             else if (use_mask[4]) idx = 0x4F;
  1118.             else return -1;
  1119.             idx -= av_log2_16bit(use_mask[idx >> 4]);
  1120.         }
  1121.         if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
  1122.             use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
  1123.             n++;
  1124.             start_off = idx;
  1125.         }
  1126.     }
  1127.  
  1128.     fcb->x[fcb->n] = start_off;
  1129.     fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
  1130.     fcb->n++;
  1131.  
  1132.     /* set offset for next block, relative to start of that block */
  1133.     n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
  1134.     s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
  1135.     return 0;
  1136. }
  1137.  
  1138. /**
  1139.  * Apply first set of pitch-adaptive window pulses.
  1140.  * @param s WMA Voice decoding context private data
  1141.  * @param gb bit I/O context
  1142.  * @param block_idx block index in frame [0, 1]
  1143.  * @param fcb storage location for fixed codebook pulse info
  1144.  */
  1145. static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
  1146.                           int block_idx, AMRFixed *fcb)
  1147. {
  1148.     int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
  1149.     float v;
  1150.  
  1151.     if (s->aw_n_pulses[block_idx] > 0) {
  1152.         int n, v_mask, i_mask, sh, n_pulses;
  1153.  
  1154.         if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
  1155.             n_pulses = 3;
  1156.             v_mask   = 8;
  1157.             i_mask   = 7;
  1158.             sh       = 4;
  1159.         } else { // 4 pulses, 1:sign + 2:index each
  1160.             n_pulses = 4;
  1161.             v_mask   = 4;
  1162.             i_mask   = 3;
  1163.             sh       = 3;
  1164.         }
  1165.  
  1166.         for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
  1167.             fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
  1168.             fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
  1169.                                  s->aw_first_pulse_off[block_idx];
  1170.             while (fcb->x[fcb->n] < 0)
  1171.                 fcb->x[fcb->n] += fcb->pitch_lag;
  1172.             if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
  1173.                 fcb->n++;
  1174.         }
  1175.     } else {
  1176.         int num2 = (val & 0x1FF) >> 1, delta, idx;
  1177.  
  1178.         if (num2 < 1 * 79)      { delta = 1; idx = num2 + 1; }
  1179.         else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
  1180.         else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
  1181.         else                    { delta = 7; idx = num2 + 1 - 3 * 75; }
  1182.         v = (val & 0x200) ? -1.0 : 1.0;
  1183.  
  1184.         fcb->no_repeat_mask |= 3 << fcb->n;
  1185.         fcb->x[fcb->n]       = idx - delta;
  1186.         fcb->y[fcb->n]       = v;
  1187.         fcb->x[fcb->n + 1]   = idx;
  1188.         fcb->y[fcb->n + 1]   = (val & 1) ? -v : v;
  1189.         fcb->n              += 2;
  1190.     }
  1191. }
  1192.  
  1193. /**
  1194.  * @}
  1195.  *
  1196.  * Generate a random number from frame_cntr and block_idx, which will lief
  1197.  * in the range [0, 1000 - block_size] (so it can be used as an index in a
  1198.  * table of size 1000 of which you want to read block_size entries).
  1199.  *
  1200.  * @param frame_cntr current frame number
  1201.  * @param block_num current block index
  1202.  * @param block_size amount of entries we want to read from a table
  1203.  *                   that has 1000 entries
  1204.  * @return a (non-)random number in the [0, 1000 - block_size] range.
  1205.  */
  1206. static int pRNG(int frame_cntr, int block_num, int block_size)
  1207. {
  1208.     /* array to simplify the calculation of z:
  1209.      * y = (x % 9) * 5 + 6;
  1210.      * z = (49995 * x) / y;
  1211.      * Since y only has 9 values, we can remove the division by using a
  1212.      * LUT and using FASTDIV-style divisions. For each of the 9 values
  1213.      * of y, we can rewrite z as:
  1214.      * z = x * (49995 / y) + x * ((49995 % y) / y)
  1215.      * In this table, each col represents one possible value of y, the
  1216.      * first number is 49995 / y, and the second is the FASTDIV variant
  1217.      * of 49995 % y / y. */
  1218.     static const unsigned int div_tbl[9][2] = {
  1219.         { 8332,  3 * 715827883U }, // y =  6
  1220.         { 4545,  0 * 390451573U }, // y = 11
  1221.         { 3124, 11 * 268435456U }, // y = 16
  1222.         { 2380, 15 * 204522253U }, // y = 21
  1223.         { 1922, 23 * 165191050U }, // y = 26
  1224.         { 1612, 23 * 138547333U }, // y = 31
  1225.         { 1388, 27 * 119304648U }, // y = 36
  1226.         { 1219, 16 * 104755300U }, // y = 41
  1227.         { 1086, 39 *  93368855U }  // y = 46
  1228.     };
  1229.     unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
  1230.     if (x >= 0xFFFF) x -= 0xFFFF;   // max value of x is 8*1877+0xFFFE=0x13AA6,
  1231.                                     // so this is effectively a modulo (%)
  1232.     y = x - 9 * MULH(477218589, x); // x % 9
  1233.     z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
  1234.                                     // z = x * 49995 / (y * 5 + 6)
  1235.     return z % (1000 - block_size);
  1236. }
  1237.  
  1238. /**
  1239.  * Parse hardcoded signal for a single block.
  1240.  * @note see #synth_block().
  1241.  */
  1242. static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
  1243.                                  int block_idx, int size,
  1244.                                  const struct frame_type_desc *frame_desc,
  1245.                                  float *excitation)
  1246. {
  1247.     float gain;
  1248.     int n, r_idx;
  1249.  
  1250.     av_assert0(size <= MAX_FRAMESIZE);
  1251.  
  1252.     /* Set the offset from which we start reading wmavoice_std_codebook */
  1253.     if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
  1254.         r_idx = pRNG(s->frame_cntr, block_idx, size);
  1255.         gain  = s->silence_gain;
  1256.     } else /* FCB_TYPE_HARDCODED */ {
  1257.         r_idx = get_bits(gb, 8);
  1258.         gain  = wmavoice_gain_universal[get_bits(gb, 6)];
  1259.     }
  1260.  
  1261.     /* Clear gain prediction parameters */
  1262.     memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
  1263.  
  1264.     /* Apply gain to hardcoded codebook and use that as excitation signal */
  1265.     for (n = 0; n < size; n++)
  1266.         excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
  1267. }
  1268.  
  1269. /**
  1270.  * Parse FCB/ACB signal for a single block.
  1271.  * @note see #synth_block().
  1272.  */
  1273. static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
  1274.                                 int block_idx, int size,
  1275.                                 int block_pitch_sh2,
  1276.                                 const struct frame_type_desc *frame_desc,
  1277.                                 float *excitation)
  1278. {
  1279.     static const float gain_coeff[6] = {
  1280.         0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
  1281.     };
  1282.     float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
  1283.     int n, idx, gain_weight;
  1284.     AMRFixed fcb;
  1285.  
  1286.     av_assert0(size <= MAX_FRAMESIZE / 2);
  1287.     memset(pulses, 0, sizeof(*pulses) * size);
  1288.  
  1289.     fcb.pitch_lag      = block_pitch_sh2 >> 2;
  1290.     fcb.pitch_fac      = 1.0;
  1291.     fcb.no_repeat_mask = 0;
  1292.     fcb.n              = 0;
  1293.  
  1294.     /* For the other frame types, this is where we apply the innovation
  1295.      * (fixed) codebook pulses of the speech signal. */
  1296.     if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
  1297.         aw_pulse_set1(s, gb, block_idx, &fcb);
  1298.         if (aw_pulse_set2(s, gb, block_idx, &fcb)) {
  1299.             /* Conceal the block with silence and return.
  1300.              * Skip the correct amount of bits to read the next
  1301.              * block from the correct offset. */
  1302.             int r_idx = pRNG(s->frame_cntr, block_idx, size);
  1303.  
  1304.             for (n = 0; n < size; n++)
  1305.                 excitation[n] =
  1306.                     wmavoice_std_codebook[r_idx + n] * s->silence_gain;
  1307.             skip_bits(gb, 7 + 1);
  1308.             return;
  1309.         }
  1310.     } else /* FCB_TYPE_EXC_PULSES */ {
  1311.         int offset_nbits = 5 - frame_desc->log_n_blocks;
  1312.  
  1313.         fcb.no_repeat_mask = -1;
  1314.         /* similar to ff_decode_10_pulses_35bits(), but with single pulses
  1315.          * (instead of double) for a subset of pulses */
  1316.         for (n = 0; n < 5; n++) {
  1317.             float sign;
  1318.             int pos1, pos2;
  1319.  
  1320.             sign           = get_bits1(gb) ? 1.0 : -1.0;
  1321.             pos1           = get_bits(gb, offset_nbits);
  1322.             fcb.x[fcb.n]   = n + 5 * pos1;
  1323.             fcb.y[fcb.n++] = sign;
  1324.             if (n < frame_desc->dbl_pulses) {
  1325.                 pos2           = get_bits(gb, offset_nbits);
  1326.                 fcb.x[fcb.n]   = n + 5 * pos2;
  1327.                 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
  1328.             }
  1329.         }
  1330.     }
  1331.     ff_set_fixed_vector(pulses, &fcb, 1.0, size);
  1332.  
  1333.     /* Calculate gain for adaptive & fixed codebook signal.
  1334.      * see ff_amr_set_fixed_gain(). */
  1335.     idx = get_bits(gb, 7);
  1336.     fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
  1337.                                                  gain_coeff, 6) -
  1338.                     5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
  1339.     acb_gain = wmavoice_gain_codebook_acb[idx];
  1340.     pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
  1341.                         -2.9957322736 /* log(0.05) */,
  1342.                          1.6094379124 /* log(5.0)  */);
  1343.  
  1344.     gain_weight = 8 >> frame_desc->log_n_blocks;
  1345.     memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
  1346.             sizeof(*s->gain_pred_err) * (6 - gain_weight));
  1347.     for (n = 0; n < gain_weight; n++)
  1348.         s->gain_pred_err[n] = pred_err;
  1349.  
  1350.     /* Calculation of adaptive codebook */
  1351.     if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
  1352.         int len;
  1353.         for (n = 0; n < size; n += len) {
  1354.             int next_idx_sh16;
  1355.             int abs_idx    = block_idx * size + n;
  1356.             int pitch_sh16 = (s->last_pitch_val << 16) +
  1357.                              s->pitch_diff_sh16 * abs_idx;
  1358.             int pitch      = (pitch_sh16 + 0x6FFF) >> 16;
  1359.             int idx_sh16   = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
  1360.             idx            = idx_sh16 >> 16;
  1361.             if (s->pitch_diff_sh16) {
  1362.                 if (s->pitch_diff_sh16 > 0) {
  1363.                     next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
  1364.                 } else
  1365.                     next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
  1366.                 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
  1367.                               1, size - n);
  1368.             } else
  1369.                 len = size;
  1370.  
  1371.             ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
  1372.                                   wmavoice_ipol1_coeffs, 17,
  1373.                                   idx, 9, len);
  1374.         }
  1375.     } else /* ACB_TYPE_HAMMING */ {
  1376.         int block_pitch = block_pitch_sh2 >> 2;
  1377.         idx             = block_pitch_sh2 & 3;
  1378.         if (idx) {
  1379.             ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
  1380.                                   wmavoice_ipol2_coeffs, 4,
  1381.                                   idx, 8, size);
  1382.         } else
  1383.             av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
  1384.                               sizeof(float) * size);
  1385.     }
  1386.  
  1387.     /* Interpolate ACB/FCB and use as excitation signal */
  1388.     ff_weighted_vector_sumf(excitation, excitation, pulses,
  1389.                             acb_gain, fcb_gain, size);
  1390. }
  1391.  
  1392. /**
  1393.  * Parse data in a single block.
  1394.  * @note we assume enough bits are available, caller should check.
  1395.  *
  1396.  * @param s WMA Voice decoding context private data
  1397.  * @param gb bit I/O context
  1398.  * @param block_idx index of the to-be-read block
  1399.  * @param size amount of samples to be read in this block
  1400.  * @param block_pitch_sh2 pitch for this block << 2
  1401.  * @param lsps LSPs for (the end of) this frame
  1402.  * @param prev_lsps LSPs for the last frame
  1403.  * @param frame_desc frame type descriptor
  1404.  * @param excitation target memory for the ACB+FCB interpolated signal
  1405.  * @param synth target memory for the speech synthesis filter output
  1406.  * @return 0 on success, <0 on error.
  1407.  */
  1408. static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
  1409.                         int block_idx, int size,
  1410.                         int block_pitch_sh2,
  1411.                         const double *lsps, const double *prev_lsps,
  1412.                         const struct frame_type_desc *frame_desc,
  1413.                         float *excitation, float *synth)
  1414. {
  1415.     double i_lsps[MAX_LSPS];
  1416.     float lpcs[MAX_LSPS];
  1417.     float fac;
  1418.     int n;
  1419.  
  1420.     if (frame_desc->acb_type == ACB_TYPE_NONE)
  1421.         synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
  1422.     else
  1423.         synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
  1424.                             frame_desc, excitation);
  1425.  
  1426.     /* convert interpolated LSPs to LPCs */
  1427.     fac = (block_idx + 0.5) / frame_desc->n_blocks;
  1428.     for (n = 0; n < s->lsps; n++) // LSF -> LSP
  1429.         i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
  1430.     ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
  1431.  
  1432.     /* Speech synthesis */
  1433.     ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
  1434. }
  1435.  
  1436. /**
  1437.  * Synthesize output samples for a single frame.
  1438.  * @note we assume enough bits are available, caller should check.
  1439.  *
  1440.  * @param ctx WMA Voice decoder context
  1441.  * @param gb bit I/O context (s->gb or one for cross-packet superframes)
  1442.  * @param frame_idx Frame number within superframe [0-2]
  1443.  * @param samples pointer to output sample buffer, has space for at least 160
  1444.  *                samples
  1445.  * @param lsps LSP array
  1446.  * @param prev_lsps array of previous frame's LSPs
  1447.  * @param excitation target buffer for excitation signal
  1448.  * @param synth target buffer for synthesized speech data
  1449.  * @return 0 on success, <0 on error.
  1450.  */
  1451. static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
  1452.                        float *samples,
  1453.                        const double *lsps, const double *prev_lsps,
  1454.                        float *excitation, float *synth)
  1455. {
  1456.     WMAVoiceContext *s = ctx->priv_data;
  1457.     int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val);
  1458.     int pitch[MAX_BLOCKS], av_uninit(last_block_pitch);
  1459.  
  1460.     /* Parse frame type ("frame header"), see frame_descs */
  1461.     int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
  1462.  
  1463.     if (bd_idx < 0) {
  1464.         av_log(ctx, AV_LOG_ERROR,
  1465.                "Invalid frame type VLC code, skipping\n");
  1466.         return AVERROR_INVALIDDATA;
  1467.     }
  1468.  
  1469.     block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
  1470.  
  1471.     /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
  1472.     if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
  1473.         /* Pitch is provided per frame, which is interpreted as the pitch of
  1474.          * the last sample of the last block of this frame. We can interpolate
  1475.          * the pitch of other blocks (and even pitch-per-sample) by gradually
  1476.          * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
  1477.         n_blocks_x2      = frame_descs[bd_idx].n_blocks << 1;
  1478.         log_n_blocks_x2  = frame_descs[bd_idx].log_n_blocks + 1;
  1479.         cur_pitch_val    = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
  1480.         cur_pitch_val    = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
  1481.         if (s->last_acb_type == ACB_TYPE_NONE ||
  1482.             20 * abs(cur_pitch_val - s->last_pitch_val) >
  1483.                 (cur_pitch_val + s->last_pitch_val))
  1484.             s->last_pitch_val = cur_pitch_val;
  1485.  
  1486.         /* pitch per block */
  1487.         for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
  1488.             int fac = n * 2 + 1;
  1489.  
  1490.             pitch[n] = (MUL16(fac,                 cur_pitch_val) +
  1491.                         MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
  1492.                         frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
  1493.         }
  1494.  
  1495.         /* "pitch-diff-per-sample" for calculation of pitch per sample */
  1496.         s->pitch_diff_sh16 =
  1497.             ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
  1498.     }
  1499.  
  1500.     /* Global gain (if silence) and pitch-adaptive window coordinates */
  1501.     switch (frame_descs[bd_idx].fcb_type) {
  1502.     case FCB_TYPE_SILENCE:
  1503.         s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
  1504.         break;
  1505.     case FCB_TYPE_AW_PULSES:
  1506.         aw_parse_coords(s, gb, pitch);
  1507.         break;
  1508.     }
  1509.  
  1510.     for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
  1511.         int bl_pitch_sh2;
  1512.  
  1513.         /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
  1514.         switch (frame_descs[bd_idx].acb_type) {
  1515.         case ACB_TYPE_HAMMING: {
  1516.             /* Pitch is given per block. Per-block pitches are encoded as an
  1517.              * absolute value for the first block, and then delta values
  1518.              * relative to this value) for all subsequent blocks. The scale of
  1519.              * this pitch value is semi-logaritmic compared to its use in the
  1520.              * decoder, so we convert it to normal scale also. */
  1521.             int block_pitch,
  1522.                 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
  1523.                 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
  1524.                 t3 =  s->block_conv_table[3] - s->block_conv_table[2] + 1;
  1525.  
  1526.             if (n == 0) {
  1527.                 block_pitch = get_bits(gb, s->block_pitch_nbits);
  1528.             } else
  1529.                 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
  1530.                                  get_bits(gb, s->block_delta_pitch_nbits);
  1531.             /* Convert last_ so that any next delta is within _range */
  1532.             last_block_pitch = av_clip(block_pitch,
  1533.                                        s->block_delta_pitch_hrange,
  1534.                                        s->block_pitch_range -
  1535.                                            s->block_delta_pitch_hrange);
  1536.  
  1537.             /* Convert semi-log-style scale back to normal scale */
  1538.             if (block_pitch < t1) {
  1539.                 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
  1540.             } else {
  1541.                 block_pitch -= t1;
  1542.                 if (block_pitch < t2) {
  1543.                     bl_pitch_sh2 =
  1544.                         (s->block_conv_table[1] << 2) + (block_pitch << 1);
  1545.                 } else {
  1546.                     block_pitch -= t2;
  1547.                     if (block_pitch < t3) {
  1548.                         bl_pitch_sh2 =
  1549.                             (s->block_conv_table[2] + block_pitch) << 2;
  1550.                     } else
  1551.                         bl_pitch_sh2 = s->block_conv_table[3] << 2;
  1552.                 }
  1553.             }
  1554.             pitch[n] = bl_pitch_sh2 >> 2;
  1555.             break;
  1556.         }
  1557.  
  1558.         case ACB_TYPE_ASYMMETRIC: {
  1559.             bl_pitch_sh2 = pitch[n] << 2;
  1560.             break;
  1561.         }
  1562.  
  1563.         default: // ACB_TYPE_NONE has no pitch
  1564.             bl_pitch_sh2 = 0;
  1565.             break;
  1566.         }
  1567.  
  1568.         synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
  1569.                     lsps, prev_lsps, &frame_descs[bd_idx],
  1570.                     &excitation[n * block_nsamples],
  1571.                     &synth[n * block_nsamples]);
  1572.     }
  1573.  
  1574.     /* Averaging projection filter, if applicable. Else, just copy samples
  1575.      * from synthesis buffer */
  1576.     if (s->do_apf) {
  1577.         double i_lsps[MAX_LSPS];
  1578.         float lpcs[MAX_LSPS];
  1579.  
  1580.         for (n = 0; n < s->lsps; n++) // LSF -> LSP
  1581.             i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
  1582.         ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
  1583.         postfilter(s, synth, samples, 80, lpcs,
  1584.                    &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
  1585.                    frame_descs[bd_idx].fcb_type, pitch[0]);
  1586.  
  1587.         for (n = 0; n < s->lsps; n++) // LSF -> LSP
  1588.             i_lsps[n] = cos(lsps[n]);
  1589.         ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
  1590.         postfilter(s, &synth[80], &samples[80], 80, lpcs,
  1591.                    &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
  1592.                    frame_descs[bd_idx].fcb_type, pitch[0]);
  1593.     } else
  1594.         memcpy(samples, synth, 160 * sizeof(synth[0]));
  1595.  
  1596.     /* Cache values for next frame */
  1597.     s->frame_cntr++;
  1598.     if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
  1599.     s->last_acb_type = frame_descs[bd_idx].acb_type;
  1600.     switch (frame_descs[bd_idx].acb_type) {
  1601.     case ACB_TYPE_NONE:
  1602.         s->last_pitch_val = 0;
  1603.         break;
  1604.     case ACB_TYPE_ASYMMETRIC:
  1605.         s->last_pitch_val = cur_pitch_val;
  1606.         break;
  1607.     case ACB_TYPE_HAMMING:
  1608.         s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
  1609.         break;
  1610.     }
  1611.  
  1612.     return 0;
  1613. }
  1614.  
  1615. /**
  1616.  * Ensure minimum value for first item, maximum value for last value,
  1617.  * proper spacing between each value and proper ordering.
  1618.  *
  1619.  * @param lsps array of LSPs
  1620.  * @param num size of LSP array
  1621.  *
  1622.  * @note basically a double version of #ff_acelp_reorder_lsf(), might be
  1623.  *       useful to put in a generic location later on. Parts are also
  1624.  *       present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
  1625.  *       which is in float.
  1626.  */
  1627. static void stabilize_lsps(double *lsps, int num)
  1628. {
  1629.     int n, m, l;
  1630.  
  1631.     /* set minimum value for first, maximum value for last and minimum
  1632.      * spacing between LSF values.
  1633.      * Very similar to ff_set_min_dist_lsf(), but in double. */
  1634.     lsps[0]       = FFMAX(lsps[0],       0.0015 * M_PI);
  1635.     for (n = 1; n < num; n++)
  1636.         lsps[n]   = FFMAX(lsps[n],       lsps[n - 1] + 0.0125 * M_PI);
  1637.     lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
  1638.  
  1639.     /* reorder (looks like one-time / non-recursed bubblesort).
  1640.      * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
  1641.     for (n = 1; n < num; n++) {
  1642.         if (lsps[n] < lsps[n - 1]) {
  1643.             for (m = 1; m < num; m++) {
  1644.                 double tmp = lsps[m];
  1645.                 for (l = m - 1; l >= 0; l--) {
  1646.                     if (lsps[l] <= tmp) break;
  1647.                     lsps[l + 1] = lsps[l];
  1648.                 }
  1649.                 lsps[l + 1] = tmp;
  1650.             }
  1651.             break;
  1652.         }
  1653.     }
  1654. }
  1655.  
  1656. /**
  1657.  * Test if there's enough bits to read 1 superframe.
  1658.  *
  1659.  * @param orig_gb bit I/O context used for reading. This function
  1660.  *                does not modify the state of the bitreader; it
  1661.  *                only uses it to copy the current stream position
  1662.  * @param s WMA Voice decoding context private data
  1663.  * @return < 0 on error, 1 on not enough bits or 0 if OK.
  1664.  */
  1665. static int check_bits_for_superframe(GetBitContext *orig_gb,
  1666.                                      WMAVoiceContext *s)
  1667. {
  1668.     GetBitContext s_gb, *gb = &s_gb;
  1669.     int n, need_bits, bd_idx;
  1670.     const struct frame_type_desc *frame_desc;
  1671.  
  1672.     /* initialize a copy */
  1673.     init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
  1674.     skip_bits_long(gb, get_bits_count(orig_gb));
  1675.     av_assert1(get_bits_left(gb) == get_bits_left(orig_gb));
  1676.  
  1677.     /* superframe header */
  1678.     if (get_bits_left(gb) < 14)
  1679.         return 1;
  1680.     if (!get_bits1(gb))
  1681.         return AVERROR(ENOSYS);           // WMAPro-in-WMAVoice superframe
  1682.     if (get_bits1(gb)) skip_bits(gb, 12); // number of  samples in superframe
  1683.     if (s->has_residual_lsps) {           // residual LSPs (for all frames)
  1684.         if (get_bits_left(gb) < s->sframe_lsp_bitsize)
  1685.             return 1;
  1686.         skip_bits_long(gb, s->sframe_lsp_bitsize);
  1687.     }
  1688.  
  1689.     /* frames */
  1690.     for (n = 0; n < MAX_FRAMES; n++) {
  1691.         int aw_idx_is_ext = 0;
  1692.  
  1693.         if (!s->has_residual_lsps) {     // independent LSPs (per-frame)
  1694.            if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
  1695.            skip_bits_long(gb, s->frame_lsp_bitsize);
  1696.         }
  1697.         bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
  1698.         if (bd_idx < 0)
  1699.             return AVERROR_INVALIDDATA; // invalid frame type VLC code
  1700.         frame_desc = &frame_descs[bd_idx];
  1701.         if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
  1702.             if (get_bits_left(gb) < s->pitch_nbits)
  1703.                 return 1;
  1704.             skip_bits_long(gb, s->pitch_nbits);
  1705.         }
  1706.         if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
  1707.             skip_bits(gb, 8);
  1708.         } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
  1709.             int tmp = get_bits(gb, 6);
  1710.             if (tmp >= 0x36) {
  1711.                 skip_bits(gb, 2);
  1712.                 aw_idx_is_ext = 1;
  1713.             }
  1714.         }
  1715.  
  1716.         /* blocks */
  1717.         if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
  1718.             need_bits = s->block_pitch_nbits +
  1719.                 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
  1720.         } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
  1721.             need_bits = 2 * !aw_idx_is_ext;
  1722.         } else
  1723.             need_bits = 0;
  1724.         need_bits += frame_desc->frame_size;
  1725.         if (get_bits_left(gb) < need_bits)
  1726.             return 1;
  1727.         skip_bits_long(gb, need_bits);
  1728.     }
  1729.  
  1730.     return 0;
  1731. }
  1732.  
  1733. /**
  1734.  * Synthesize output samples for a single superframe. If we have any data
  1735.  * cached in s->sframe_cache, that will be used instead of whatever is loaded
  1736.  * in s->gb.
  1737.  *
  1738.  * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
  1739.  * to give a total of 480 samples per frame. See #synth_frame() for frame
  1740.  * parsing. In addition to 3 frames, superframes can also contain the LSPs
  1741.  * (if these are globally specified for all frames (residually); they can
  1742.  * also be specified individually per-frame. See the s->has_residual_lsps
  1743.  * option), and can specify the number of samples encoded in this superframe
  1744.  * (if less than 480), usually used to prevent blanks at track boundaries.
  1745.  *
  1746.  * @param ctx WMA Voice decoder context
  1747.  * @return 0 on success, <0 on error or 1 if there was not enough data to
  1748.  *         fully parse the superframe
  1749.  */
  1750. static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
  1751.                             int *got_frame_ptr)
  1752. {
  1753.     WMAVoiceContext *s = ctx->priv_data;
  1754.     GetBitContext *gb = &s->gb, s_gb;
  1755.     int n, res, n_samples = 480;
  1756.     double lsps[MAX_FRAMES][MAX_LSPS];
  1757.     const double *mean_lsf = s->lsps == 16 ?
  1758.         wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
  1759.     float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
  1760.     float synth[MAX_LSPS + MAX_SFRAMESIZE];
  1761.     float *samples;
  1762.  
  1763.     memcpy(synth,      s->synth_history,
  1764.            s->lsps             * sizeof(*synth));
  1765.     memcpy(excitation, s->excitation_history,
  1766.            s->history_nsamples * sizeof(*excitation));
  1767.  
  1768.     if (s->sframe_cache_size > 0) {
  1769.         gb = &s_gb;
  1770.         init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
  1771.         s->sframe_cache_size = 0;
  1772.     }
  1773.  
  1774.     if ((res = check_bits_for_superframe(gb, s)) == 1) {
  1775.         *got_frame_ptr = 0;
  1776.         return 1;
  1777.     } else if (res < 0)
  1778.         return res;
  1779.  
  1780.     /* First bit is speech/music bit, it differentiates between WMAVoice
  1781.      * speech samples (the actual codec) and WMAVoice music samples, which
  1782.      * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
  1783.      * the wild yet. */
  1784.     if (!get_bits1(gb)) {
  1785.         avpriv_request_sample(ctx, "WMAPro-in-WMAVoice");
  1786.         return AVERROR_PATCHWELCOME;
  1787.     }
  1788.  
  1789.     /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
  1790.     if (get_bits1(gb)) {
  1791.         if ((n_samples = get_bits(gb, 12)) > 480) {
  1792.             av_log(ctx, AV_LOG_ERROR,
  1793.                    "Superframe encodes >480 samples (%d), not allowed\n",
  1794.                    n_samples);
  1795.             return AVERROR_INVALIDDATA;
  1796.         }
  1797.     }
  1798.     /* Parse LSPs, if global for the superframe (can also be per-frame). */
  1799.     if (s->has_residual_lsps) {
  1800.         double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
  1801.  
  1802.         for (n = 0; n < s->lsps; n++)
  1803.             prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
  1804.  
  1805.         if (s->lsps == 10) {
  1806.             dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
  1807.         } else /* s->lsps == 16 */
  1808.             dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
  1809.  
  1810.         for (n = 0; n < s->lsps; n++) {
  1811.             lsps[0][n]  = mean_lsf[n] + (a1[n]           - a2[n * 2]);
  1812.             lsps[1][n]  = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
  1813.             lsps[2][n] += mean_lsf[n];
  1814.         }
  1815.         for (n = 0; n < 3; n++)
  1816.             stabilize_lsps(lsps[n], s->lsps);
  1817.     }
  1818.  
  1819.     /* get output buffer */
  1820.     frame->nb_samples = 480;
  1821.     if ((res = ff_get_buffer(ctx, frame, 0)) < 0)
  1822.         return res;
  1823.     frame->nb_samples = n_samples;
  1824.     samples = (float *)frame->data[0];
  1825.  
  1826.     /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
  1827.     for (n = 0; n < 3; n++) {
  1828.         if (!s->has_residual_lsps) {
  1829.             int m;
  1830.  
  1831.             if (s->lsps == 10) {
  1832.                 dequant_lsp10i(gb, lsps[n]);
  1833.             } else /* s->lsps == 16 */
  1834.                 dequant_lsp16i(gb, lsps[n]);
  1835.  
  1836.             for (m = 0; m < s->lsps; m++)
  1837.                 lsps[n][m] += mean_lsf[m];
  1838.             stabilize_lsps(lsps[n], s->lsps);
  1839.         }
  1840.  
  1841.         if ((res = synth_frame(ctx, gb, n,
  1842.                                &samples[n * MAX_FRAMESIZE],
  1843.                                lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
  1844.                                &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
  1845.                                &synth[s->lsps + n * MAX_FRAMESIZE]))) {
  1846.             *got_frame_ptr = 0;
  1847.             return res;
  1848.         }
  1849.     }
  1850.  
  1851.     /* Statistics? FIXME - we don't check for length, a slight overrun
  1852.      * will be caught by internal buffer padding, and anything else
  1853.      * will be skipped, not read. */
  1854.     if (get_bits1(gb)) {
  1855.         res = get_bits(gb, 4);
  1856.         skip_bits(gb, 10 * (res + 1));
  1857.     }
  1858.  
  1859.     *got_frame_ptr = 1;
  1860.  
  1861.     /* Update history */
  1862.     memcpy(s->prev_lsps,           lsps[2],
  1863.            s->lsps             * sizeof(*s->prev_lsps));
  1864.     memcpy(s->synth_history,      &synth[MAX_SFRAMESIZE],
  1865.            s->lsps             * sizeof(*synth));
  1866.     memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
  1867.            s->history_nsamples * sizeof(*excitation));
  1868.     if (s->do_apf)
  1869.         memmove(s->zero_exc_pf,       &s->zero_exc_pf[MAX_SFRAMESIZE],
  1870.                 s->history_nsamples * sizeof(*s->zero_exc_pf));
  1871.  
  1872.     return 0;
  1873. }
  1874.  
  1875. /**
  1876.  * Parse the packet header at the start of each packet (input data to this
  1877.  * decoder).
  1878.  *
  1879.  * @param s WMA Voice decoding context private data
  1880.  * @return 1 if not enough bits were available, or 0 on success.
  1881.  */
  1882. static int parse_packet_header(WMAVoiceContext *s)
  1883. {
  1884.     GetBitContext *gb = &s->gb;
  1885.     unsigned int res;
  1886.  
  1887.     if (get_bits_left(gb) < 11)
  1888.         return 1;
  1889.     skip_bits(gb, 4);          // packet sequence number
  1890.     s->has_residual_lsps = get_bits1(gb);
  1891.     do {
  1892.         res = get_bits(gb, 6); // number of superframes per packet
  1893.                                // (minus first one if there is spillover)
  1894.         if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
  1895.             return 1;
  1896.     } while (res == 0x3F);
  1897.     s->spillover_nbits   = get_bits(gb, s->spillover_bitsize);
  1898.  
  1899.     return 0;
  1900. }
  1901.  
  1902. /**
  1903.  * Copy (unaligned) bits from gb/data/size to pb.
  1904.  *
  1905.  * @param pb target buffer to copy bits into
  1906.  * @param data source buffer to copy bits from
  1907.  * @param size size of the source data, in bytes
  1908.  * @param gb bit I/O context specifying the current position in the source.
  1909.  *           data. This function might use this to align the bit position to
  1910.  *           a whole-byte boundary before calling #avpriv_copy_bits() on aligned
  1911.  *           source data
  1912.  * @param nbits the amount of bits to copy from source to target
  1913.  *
  1914.  * @note after calling this function, the current position in the input bit
  1915.  *       I/O context is undefined.
  1916.  */
  1917. static void copy_bits(PutBitContext *pb,
  1918.                       const uint8_t *data, int size,
  1919.                       GetBitContext *gb, int nbits)
  1920. {
  1921.     int rmn_bytes, rmn_bits;
  1922.  
  1923.     rmn_bits = rmn_bytes = get_bits_left(gb);
  1924.     if (rmn_bits < nbits)
  1925.         return;
  1926.     if (nbits > pb->size_in_bits - put_bits_count(pb))
  1927.         return;
  1928.     rmn_bits &= 7; rmn_bytes >>= 3;
  1929.     if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
  1930.         put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
  1931.     avpriv_copy_bits(pb, data + size - rmn_bytes,
  1932.                  FFMIN(nbits - rmn_bits, rmn_bytes << 3));
  1933. }
  1934.  
  1935. /**
  1936.  * Packet decoding: a packet is anything that the (ASF) demuxer contains,
  1937.  * and we expect that the demuxer / application provides it to us as such
  1938.  * (else you'll probably get garbage as output). Every packet has a size of
  1939.  * ctx->block_align bytes, starts with a packet header (see
  1940.  * #parse_packet_header()), and then a series of superframes. Superframe
  1941.  * boundaries may exceed packets, i.e. superframes can split data over
  1942.  * multiple (two) packets.
  1943.  *
  1944.  * For more information about frames, see #synth_superframe().
  1945.  */
  1946. static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
  1947.                                   int *got_frame_ptr, AVPacket *avpkt)
  1948. {
  1949.     WMAVoiceContext *s = ctx->priv_data;
  1950.     GetBitContext *gb = &s->gb;
  1951.     int size, res, pos;
  1952.  
  1953.     /* Packets are sometimes a multiple of ctx->block_align, with a packet
  1954.      * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
  1955.      * feeds us ASF packets, which may concatenate multiple "codec" packets
  1956.      * in a single "muxer" packet, so we artificially emulate that by
  1957.      * capping the packet size at ctx->block_align. */
  1958.     for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
  1959.     if (!size) {
  1960.         *got_frame_ptr = 0;
  1961.         return 0;
  1962.     }
  1963.     init_get_bits(&s->gb, avpkt->data, size << 3);
  1964.  
  1965.     /* size == ctx->block_align is used to indicate whether we are dealing with
  1966.      * a new packet or a packet of which we already read the packet header
  1967.      * previously. */
  1968.     if (size == ctx->block_align) { // new packet header
  1969.         if ((res = parse_packet_header(s)) < 0)
  1970.             return res;
  1971.  
  1972.         /* If the packet header specifies a s->spillover_nbits, then we want
  1973.          * to push out all data of the previous packet (+ spillover) before
  1974.          * continuing to parse new superframes in the current packet. */
  1975.         if (s->spillover_nbits > 0) {
  1976.             if (s->sframe_cache_size > 0) {
  1977.                 int cnt = get_bits_count(gb);
  1978.                 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
  1979.                 flush_put_bits(&s->pb);
  1980.                 s->sframe_cache_size += s->spillover_nbits;
  1981.                 if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 &&
  1982.                     *got_frame_ptr) {
  1983.                     cnt += s->spillover_nbits;
  1984.                     s->skip_bits_next = cnt & 7;
  1985.                     return cnt >> 3;
  1986.                 } else
  1987.                     skip_bits_long (gb, s->spillover_nbits - cnt +
  1988.                                     get_bits_count(gb)); // resync
  1989.             } else
  1990.                 skip_bits_long(gb, s->spillover_nbits);  // resync
  1991.         }
  1992.     } else if (s->skip_bits_next)
  1993.         skip_bits(gb, s->skip_bits_next);
  1994.  
  1995.     /* Try parsing superframes in current packet */
  1996.     s->sframe_cache_size = 0;
  1997.     s->skip_bits_next = 0;
  1998.     pos = get_bits_left(gb);
  1999.     if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
  2000.         return res;
  2001.     } else if (*got_frame_ptr) {
  2002.         int cnt = get_bits_count(gb);
  2003.         s->skip_bits_next = cnt & 7;
  2004.         return cnt >> 3;
  2005.     } else if ((s->sframe_cache_size = pos) > 0) {
  2006.         /* rewind bit reader to start of last (incomplete) superframe... */
  2007.         init_get_bits(gb, avpkt->data, size << 3);
  2008.         skip_bits_long(gb, (size << 3) - pos);
  2009.         av_assert1(get_bits_left(gb) == pos);
  2010.  
  2011.         /* ...and cache it for spillover in next packet */
  2012.         init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
  2013.         copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
  2014.         // FIXME bad - just copy bytes as whole and add use the
  2015.         // skip_bits_next field
  2016.     }
  2017.  
  2018.     return size;
  2019. }
  2020.  
  2021. static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
  2022. {
  2023.     WMAVoiceContext *s = ctx->priv_data;
  2024.  
  2025.     if (s->do_apf) {
  2026.         ff_rdft_end(&s->rdft);
  2027.         ff_rdft_end(&s->irdft);
  2028.         ff_dct_end(&s->dct);
  2029.         ff_dct_end(&s->dst);
  2030.     }
  2031.  
  2032.     return 0;
  2033. }
  2034.  
  2035. static av_cold void wmavoice_flush(AVCodecContext *ctx)
  2036. {
  2037.     WMAVoiceContext *s = ctx->priv_data;
  2038.     int n;
  2039.  
  2040.     s->postfilter_agc    = 0;
  2041.     s->sframe_cache_size = 0;
  2042.     s->skip_bits_next    = 0;
  2043.     for (n = 0; n < s->lsps; n++)
  2044.         s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
  2045.     memset(s->excitation_history, 0,
  2046.            sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
  2047.     memset(s->synth_history,      0,
  2048.            sizeof(*s->synth_history)      * MAX_LSPS);
  2049.     memset(s->gain_pred_err,      0,
  2050.            sizeof(s->gain_pred_err));
  2051.  
  2052.     if (s->do_apf) {
  2053.         memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
  2054.                sizeof(*s->synth_filter_out_buf) * s->lsps);
  2055.         memset(s->dcf_mem,              0,
  2056.                sizeof(*s->dcf_mem)              * 2);
  2057.         memset(s->zero_exc_pf,          0,
  2058.                sizeof(*s->zero_exc_pf)          * s->history_nsamples);
  2059.         memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
  2060.     }
  2061. }
  2062.  
  2063. AVCodec ff_wmavoice_decoder = {
  2064.     .name             = "wmavoice",
  2065.     .long_name        = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
  2066.     .type             = AVMEDIA_TYPE_AUDIO,
  2067.     .id               = AV_CODEC_ID_WMAVOICE,
  2068.     .priv_data_size   = sizeof(WMAVoiceContext),
  2069.     .init             = wmavoice_decode_init,
  2070.     .init_static_data = wmavoice_init_static_data,
  2071.     .close            = wmavoice_decode_end,
  2072.     .decode           = wmavoice_decode_packet,
  2073.     .capabilities     = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
  2074.     .flush            = wmavoice_flush,
  2075. };
  2076.