text: /*
   * Windows Media Audio Voice decoder.
   * Copyright (c) 2009 Ronald S. Bultje
   *
   * This file is part of FFmpeg.
   *
   * FFmpeg is free software; you can redistribute it and/or
   * modify it under the terms of the GNU Lesser General Public
   * License as published by the Free Software Foundation; either
   * version 2.1 of the License, or (at your option) any later version.
   *
   * FFmpeg is distributed in the hope that it will be useful,
   * but WITHOUT ANY WARRANTY; without even the implied warranty of
   * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
   * Lesser General Public License for more details.
   *
   * You should have received a copy of the GNU Lesser General Public
   * License along with FFmpeg; if not, write to the Free Software
   * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
   */
  
  /**
   * @file
   * @brief Windows Media Audio Voice compatible decoder
   * @author Ronald S. Bultje <rsbultje@gmail.com>
   */
  
  #include <math.h>
  
  #include "libavutil/channel_layout.h"
  #include "libavutil/float_dsp.h"
  #include "libavutil/mem.h"
  #include "avcodec.h"
  #include "internal.h"
  #include "get_bits.h"
  #include "put_bits.h"
  #include "wmavoice_data.h"
  #include "celp_filters.h"
  #include "acelp_vectors.h"
  #include "acelp_filters.h"
  #include "lsp.h"
  #include "dct.h"
  #include "rdft.h"
  #include "sinewin.h"
  
  #define MAX_BLOCKS           8   ///< maximum number of blocks per frame
  #define MAX_LSPS             16  ///< maximum filter order
  #define MAX_LSPS_ALIGN16     16  ///< same as #MAX_LSPS; needs to be multiple
                                   ///< of 16 for ASM input buffer alignment
  #define MAX_FRAMES           3   ///< maximum number of frames per superframe
  #define MAX_FRAMESIZE        160 ///< maximum number of samples per frame
  #define MAX_SIGNAL_HISTORY   416 ///< maximum excitation signal history
  #define MAX_SFRAMESIZE       (MAX_FRAMESIZE * MAX_FRAMES)
                                   ///< maximum number of samples per superframe
  #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
                                   ///< was split over two packets
  #define VLC_NBITS            6   ///< number of bits to read per VLC iteration
  
  /**
   * Frame type VLC coding.
   */
  static VLC frame_type_vlc;
  
  /**
   * Adaptive codebook types.
   */
  enum {
      ACB_TYPE_NONE       = 0, ///< no adaptive codebook (only hardcoded fixed)
      ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
                               ///< we interpolate to get a per-sample pitch.
                               ///< Signal is generated using an asymmetric sinc
                               ///< window function
                               ///< @note see #wmavoice_ipol1_coeffs
      ACB_TYPE_HAMMING    = 2  ///< Per-block pitch with signal generation using
                               ///< a Hamming sinc window function
                               ///< @note see #wmavoice_ipol2_coeffs
  };
  
  /**
   * Fixed codebook types.
   */
  enum {
      FCB_TYPE_SILENCE    = 0, ///< comfort noise during silence
                               ///< generated from a hardcoded (fixed) codebook
                               ///< with per-frame (low) gain values
      FCB_TYPE_HARDCODED  = 1, ///< hardcoded (fixed) codebook with per-block
                               ///< gain values
      FCB_TYPE_AW_PULSES  = 2, ///< Pitch-adaptive window (AW) pulse signals,
                               ///< used in particular for low-bitrate streams
      FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
                               ///< combinations of either single pulses or
                               ///< pulse pairs
  };
  
  /**
   * Description of frame types.
   */
  static const struct frame_type_desc {
      uint8_t n_blocks;     ///< amount of blocks per frame (each block
                            ///< (contains 160/#n_blocks samples)
      uint8_t log_n_blocks; ///< log2(#n_blocks)
      uint8_t acb_type;     ///< Adaptive codebook type (ACB_TYPE_*)
      uint8_t fcb_type;     ///< Fixed codebook type (FCB_TYPE_*)
      uint8_t dbl_pulses;   ///< how many pulse vectors have pulse pairs
                            ///< (rather than just one single pulse)
                            ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
      uint16_t frame_size;  ///< the amount of bits that make up the block
                            ///< data (per frame)
  } frame_descs[17] = {
      { 1, 0, ACB_TYPE_NONE,       FCB_TYPE_SILENCE,    0,   0 },
      { 2, 1, ACB_TYPE_NONE,       FCB_TYPE_HARDCODED,  0,  28 },
      { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES,  0,  46 },
      { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2,  80 },
      { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
      { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
      { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
      { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
      { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0,  64 },
      { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2,  80 },
      { 2, 1, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 104 },
      { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 108 },
      { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 132 },
      { 4, 2, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 168 },
      { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 0, 176 },
      { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 2, 208 },
      { 8, 3, ACB_TYPE_HAMMING,    FCB_TYPE_EXC_PULSES, 5, 256 }
  };
  
  /**
   * WMA Voice decoding context.
   */
  typedef struct {
      /**
       * @name Global values specified in the stream header / extradata or used all over.
       * @{
       */
      GetBitContext gb;             ///< packet bitreader. During decoder init,
                                    ///< it contains the extradata from the
                                    ///< demuxer. During decoding, it contains
                                    ///< packet data.
      int8_t vbm_tree[25];          ///< converts VLC codes to frame type
  
      int spillover_bitsize;        ///< number of bits used to specify
                                    ///< #spillover_nbits in the packet header
                                    ///< = ceil(log2(ctx->block_align << 3))
      int history_nsamples;         ///< number of samples in history for signal
                                    ///< prediction (through ACB)
  
      /* postfilter specific values */
      int do_apf;                   ///< whether to apply the averaged
                                    ///< projection filter (APF)
      int denoise_strength;         ///< strength of denoising in Wiener filter
                                    ///< [0-11]
      int denoise_tilt_corr;        ///< Whether to apply tilt correction to the
                                    ///< Wiener filter coefficients (postfilter)
      int dc_level;                 ///< Predicted amount of DC noise, based
                                    ///< on which a DC removal filter is used
  
      int lsps;                     ///< number of LSPs per frame [10 or 16]
      int lsp_q_mode;               ///< defines quantizer defaults [0, 1]
      int lsp_def_mode;             ///< defines different sets of LSP defaults
                                    ///< [0, 1]
      int frame_lsp_bitsize;        ///< size (in bits) of LSPs, when encoded
                                    ///< per-frame (independent coding)
      int sframe_lsp_bitsize;       ///< size (in bits) of LSPs, when encoded
                                    ///< per superframe (residual coding)
  
      int min_pitch_val;            ///< base value for pitch parsing code
      int max_pitch_val;            ///< max value + 1 for pitch parsing
      int pitch_nbits;              ///< number of bits used to specify the
                                    ///< pitch value in the frame header
      int block_pitch_nbits;        ///< number of bits used to specify the
                                    ///< first block's pitch value
      int block_pitch_range;        ///< range of the block pitch
      int block_delta_pitch_nbits;  ///< number of bits used to specify the
                                    ///< delta pitch between this and the last
                                    ///< block's pitch value, used in all but
                                    ///< first block
      int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
                                    ///< from -this to +this-1)
      uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
                                    ///< conversion
  
      /**
       * @}
       *
       * @name Packet values specified in the packet header or related to a packet.
       *
       * A packet is considered to be a single unit of data provided to this
       * decoder by the demuxer.
       * @{
       */
      int spillover_nbits;          ///< number of bits of the previous packet's
                                    ///< last superframe preceding this
                                    ///< packet's first full superframe (useful
                                    ///< for re-synchronization also)
      int has_residual_lsps;        ///< if set, superframes contain one set of
                                    ///< LSPs that cover all frames, encoded as
                                    ///< independent and residual LSPs; if not
                                    ///< set, each frame contains its own, fully
                                    ///< independent, LSPs
      int skip_bits_next;           ///< number of bits to skip at the next call
                                    ///< to #wmavoice_decode_packet() (since
                                    ///< they're part of the previous superframe)
  
      uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
                                    ///< cache for superframe data split over
                                    ///< multiple packets
      int sframe_cache_size;        ///< set to >0 if we have data from an
                                    ///< (incomplete) superframe from a previous
                                    ///< packet that spilled over in the current
                                    ///< packet; specifies the amount of bits in
                                    ///< #sframe_cache
      PutBitContext pb;             ///< bitstream writer for #sframe_cache
  
      /**
       * @}
       *
       * @name Frame and superframe values
       * Superframe and frame data - these can change from frame to frame,
       * although some of them do in that case serve as a cache / history for
       * the next frame or superframe.
       * @{
       */
      double prev_lsps[MAX_LSPS];   ///< LSPs of the last frame of the previous
                                    ///< superframe
      int last_pitch_val;           ///< pitch value of the previous frame
      int last_acb_type;            ///< frame type [0-2] of the previous frame
      int pitch_diff_sh16;          ///< ((cur_pitch_val - #last_pitch_val)
                                    ///< << 16) / #MAX_FRAMESIZE
      float silence_gain;           ///< set for use in blocks if #ACB_TYPE_NONE
  
      int aw_idx_is_ext;            ///< whether the AW index was encoded in
                                    ///< 8 bits (instead of 6)
      int aw_pulse_range;           ///< the range over which #aw_pulse_set1()
                                    ///< can apply the pulse, relative to the
                                    ///< value in aw_first_pulse_off. The exact
                                    ///< position of the first AW-pulse is within
                                    ///< [pulse_off, pulse_off + this], and
                                    ///< depends on bitstream values; [16 or 24]
      int aw_n_pulses[2];           ///< number of AW-pulses in each block; note
                                    ///< that this number can be negative (in
                                    ///< which case it basically means "zero")
      int aw_first_pulse_off[2];    ///< index of first sample to which to
                                    ///< apply AW-pulses, or -0xff if unset
      int aw_next_pulse_off_cache;  ///< the position (relative to start of the
                                    ///< second block) at which pulses should
                                    ///< start to be positioned, serves as a
                                    ///< cache for pitch-adaptive window pulses
                                    ///< between blocks
  
      int frame_cntr;               ///< current frame index [0 - 0xFFFE]; is
                                    ///< only used for comfort noise in #pRNG()
      float gain_pred_err[6];       ///< cache for gain prediction
      float excitation_history[MAX_SIGNAL_HISTORY];
                                    ///< cache of the signal of previous
                                    ///< superframes, used as a history for
                                    ///< signal generation
      float synth_history[MAX_LSPS]; ///< see #excitation_history
      /**
       * @}
       *
       * @name Postfilter values
       *
       * Variables used for postfilter implementation, mostly history for
       * smoothing and so on, and context variables for FFT/iFFT.
       * @{
       */
      RDFTContext rdft, irdft;      ///< contexts for FFT-calculation in the
                                    ///< postfilter (for denoise filter)
      DCTContext dct, dst;          ///< contexts for phase shift (in Hilbert
                                    ///< transform, part of postfilter)
      float sin[511], cos[511];     ///< 8-bit cosine/sine windows over [-pi,pi]
                                    ///< range
      float postfilter_agc;         ///< gain control memory, used in
                                    ///< #adaptive_gain_control()
      float dcf_mem[2];             ///< DC filter history
      float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
                                    ///< zero filter output (i.e. excitation)
                                    ///< by postfilter
      float denoise_filter_cache[MAX_FRAMESIZE];
      int   denoise_filter_cache_size; ///< samples in #denoise_filter_cache
      DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
                                    ///< aligned buffer for LPC tilting
      DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
                                    ///< aligned buffer for denoise coefficients
      DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
                                    ///< aligned buffer for postfilter speech
                                    ///< synthesis
      /**
       * @}
       */
  } WMAVoiceContext;
  
  /**
   * Set up the variable bit mode (VBM) tree from container extradata.
   * @param gb bit I/O context.
   *           The bit context (s->gb) should be loaded with byte 23-46 of the
   *           container extradata (i.e. the ones containing the VBM tree).
   * @param vbm_tree pointer to array to which the decoded VBM tree will be
   *                 written.
   * @return 0 on success, <0 on error.
   */
  static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
  {
      static const uint8_t bits[] = {
           2,  2,  2,  4,  4,  4,
           6,  6,  6,  8,  8,  8,
          10, 10, 10, 12, 12, 12,
          14, 14, 14, 14
      };
      static const uint16_t codes[] = {
            0x0000, 0x0001, 0x0002,        //              00/01/10
            0x000c, 0x000d, 0x000e,        //           11+00/01/10
            0x003c, 0x003d, 0x003e,        //         1111+00/01/10
            0x00fc, 0x00fd, 0x00fe,        //       111111+00/01/10
            0x03fc, 0x03fd, 0x03fe,        //     11111111+00/01/10
            0x0ffc, 0x0ffd, 0x0ffe,        //   1111111111+00/01/10
            0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
      };
      int cntr[8] = { 0 }, n, res;
  
      memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
      for (n = 0; n < 17; n++) {
          res = get_bits(gb, 3);
          if (cntr[res] > 3) // should be >= 3 + (res == 7))
              return -1;
          vbm_tree[res * 3 + cntr[res]++] = n;
      }
      INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
                      bits, 1, 1, codes, 2, 2, 132);
      return 0;
  }
  
  /**
   * Set up decoder with parameters from demuxer (extradata etc.).
   */
  static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
  {
      int n, flags, pitch_range, lsp16_flag;
      WMAVoiceContext *s = ctx->priv_data;
  
      /**
       * Extradata layout:
       * - byte  0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
       * - byte 19-22: flags field (annoyingly in LE; see below for known
       *               values),
       * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
       *               rest is 0).
       */
      if (ctx->extradata_size != 46) {
          av_log(ctx, AV_LOG_ERROR,
                 "Invalid extradata size %d (should be 46)\n",
                 ctx->extradata_size);
          return -1;
      }
      flags                = AV_RL32(ctx->extradata + 18);
      s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
      s->do_apf            =    flags & 0x1;
      if (s->do_apf) {
          ff_rdft_init(&s->rdft,  7, DFT_R2C);
          ff_rdft_init(&s->irdft, 7, IDFT_C2R);
          ff_dct_init(&s->dct,  6, DCT_I);
          ff_dct_init(&s->dst,  6, DST_I);
  
          ff_sine_window_init(s->cos, 256);
          memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
          for (n = 0; n < 255; n++) {
              s->sin[n]       = -s->sin[510 - n];
              s->cos[510 - n] =  s->cos[n];
          }
      }
      s->denoise_strength  =   (flags >> 2) & 0xF;
      if (s->denoise_strength >= 12) {
          av_log(ctx, AV_LOG_ERROR,
                 "Invalid denoise filter strength %d (max=11)\n",
                 s->denoise_strength);
          return -1;
      }
      s->denoise_tilt_corr = !!(flags & 0x40);
      s->dc_level          =   (flags >> 7) & 0xF;
      s->lsp_q_mode        = !!(flags & 0x2000);
      s->lsp_def_mode      = !!(flags & 0x4000);
      lsp16_flag           =    flags & 0x1000;
      if (lsp16_flag) {
          s->lsps               = 16;
          s->frame_lsp_bitsize  = 34;
          s->sframe_lsp_bitsize = 60;
      } else {
          s->lsps               = 10;
          s->frame_lsp_bitsize  = 24;
          s->sframe_lsp_bitsize = 48;
      }
      for (n = 0; n < s->lsps; n++)
          s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
  
      init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
      if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
          av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
          return -1;
      }
  
      s->min_pitch_val    = ((ctx->sample_rate << 8)      /  400 + 50) >> 8;
      s->max_pitch_val    = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
      pitch_range         = s->max_pitch_val - s->min_pitch_val;
      if (pitch_range <= 0) {
          av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
          return -1;
      }
      s->pitch_nbits      = av_ceil_log2(pitch_range);
      s->last_pitch_val   = 40;
      s->last_acb_type    = ACB_TYPE_NONE;
      s->history_nsamples = s->max_pitch_val + 8;
  
      if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
          int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
              max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
  
          av_log(ctx, AV_LOG_ERROR,
                 "Unsupported samplerate %d (min=%d, max=%d)\n",
                 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
  
          return -1;
      }
  
      s->block_conv_table[0]      = s->min_pitch_val;
      s->block_conv_table[1]      = (pitch_range * 25) >> 6;
      s->block_conv_table[2]      = (pitch_range * 44) >> 6;
      s->block_conv_table[3]      = s->max_pitch_val - 1;
      s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
      if (s->block_delta_pitch_hrange <= 0) {
          av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
          return -1;
      }
      s->block_delta_pitch_nbits  = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
      s->block_pitch_range        = s->block_conv_table[2] +
                                    s->block_conv_table[3] + 1 +
                                    2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
      s->block_pitch_nbits        = av_ceil_log2(s->block_pitch_range);
  
      ctx->channels               = 1;
      ctx->channel_layout         = AV_CH_LAYOUT_MONO;
      ctx->sample_fmt             = AV_SAMPLE_FMT_FLT;
  
      return 0;
  }
  
  /**
   * @name Postfilter functions
   * Postfilter functions (gain control, wiener denoise filter, DC filter,
   * kalman smoothening, plus surrounding code to wrap it)
   * @{
   */
  /**
   * Adaptive gain control (as used in postfilter).
   *
   * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
   * that the energy here is calculated using sum(abs(...)), whereas the
   * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
   *
   * @param out output buffer for filtered samples
   * @param in input buffer containing the samples as they are after the
   *           postfilter steps so far
   * @param speech_synth input buffer containing speech synth before postfilter
   * @param size input buffer size
   * @param alpha exponential filter factor
   * @param gain_mem pointer to filter memory (single float)
   */
  static void adaptive_gain_control(float *out, const float *in,
                                    const float *speech_synth,
                                    int size, float alpha, float *gain_mem)
  {
      int i;
      float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
      float mem = *gain_mem;
  
      for (i = 0; i < size; i++) {
          speech_energy     += fabsf(speech_synth[i]);
          postfilter_energy += fabsf(in[i]);
      }
      gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
  
      for (i = 0; i < size; i++) {
          mem = alpha * mem + gain_scale_factor;
          out[i] = in[i] * mem;
      }
  
      *gain_mem = mem;
  }
  
  /**
   * Kalman smoothing function.
   *
   * This function looks back pitch +/- 3 samples back into history to find
   * the best fitting curve (that one giving the optimal gain of the two
   * signals, i.e. the highest dot product between the two), and then
   * uses that signal history to smoothen the output of the speech synthesis
   * filter.
   *
   * @param s WMA Voice decoding context
   * @param pitch pitch of the speech signal
   * @param in input speech signal
   * @param out output pointer for smoothened signal
   * @param size input/output buffer size
   *
   * @returns -1 if no smoothening took place, e.g. because no optimal
   *          fit could be found, or 0 on success.
   */
  static int kalman_smoothen(WMAVoiceContext *s, int pitch,
                             const float *in, float *out, int size)
  {
      int n;
      float optimal_gain = 0, dot;
      const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
                  *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
                  *best_hist_ptr = NULL;
  
      /* find best fitting point in history */
      do {
          dot = avpriv_scalarproduct_float_c(in, ptr, size);
          if (dot > optimal_gain) {
              optimal_gain  = dot;
              best_hist_ptr = ptr;
          }
      } while (--ptr >= end);
  
      if (optimal_gain <= 0)
          return -1;
      dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
      if (dot <= 0) // would be 1.0
          return -1;
  
      if (optimal_gain <= dot) {
          dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
      } else
          dot = 0.625;
  
      /* actual smoothing */
      for (n = 0; n < size; n++)
          out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
  
      return 0;
  }
  
  /**
   * Get the tilt factor of a formant filter from its transfer function
   * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
   *      but somehow (??) it does a speech synthesis filter in the
   *      middle, which is missing here
   *
   * @param lpcs LPC coefficients
   * @param n_lpcs Size of LPC buffer
   * @returns the tilt factor
   */
  static float tilt_factor(const float *lpcs, int n_lpcs)
  {
      float rh0, rh1;
  
      rh0 = 1.0     + avpriv_scalarproduct_float_c(lpcs,  lpcs,    n_lpcs);
      rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
  
      return rh1 / rh0;
  }
  
  /**
   * Derive denoise filter coefficients (in real domain) from the LPCs.
   */
  static void calc_input_response(WMAVoiceContext *s, float *lpcs,
                                  int fcb_type, float *coeffs, int remainder)
  {
      float last_coeff, min = 15.0, max = -15.0;
      float irange, angle_mul, gain_mul, range, sq;
      int n, idx;
  
      /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
      s->rdft.rdft_calc(&s->rdft, lpcs);
  #define log_range(var, assign) do { \
          float tmp = log10f(assign);  var = tmp; \
          max       = FFMAX(max, tmp); min = FFMIN(min, tmp); \
      } while (0)
      log_range(last_coeff,  lpcs[1]         * lpcs[1]);
      for (n = 1; n < 64; n++)
          log_range(lpcs[n], lpcs[n * 2]     * lpcs[n * 2] +
                             lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
      log_range(lpcs[0],     lpcs[0]         * lpcs[0]);
  #undef log_range
      range    = max - min;
      lpcs[64] = last_coeff;
  
      /* Now, use this spectrum to pick out these frequencies with higher
       * (relative) power/energy (which we then take to be "not noise"),
       * and set up a table (still in lpc[]) of (relative) gains per frequency.
       * These frequencies will be maintained, while others ("noise") will be
       * decreased in the filter output. */
      irange    = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
      gain_mul  = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
                                                            (5.0 / 14.7));
      angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
      for (n = 0; n <= 64; n++) {
          float pwr;
  
          idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
          pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
          lpcs[n] = angle_mul * pwr;
  
          /* 70.57 =~ 1/log10(1.0331663) */
          idx = (pwr * gain_mul - 0.0295) * 70.570526123;
          if (idx > 127) { // fallback if index falls outside table range
              coeffs[n] = wmavoice_energy_table[127] *
                          powf(1.0331663, idx - 127);
          } else
              coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
      }
  
      /* calculate the Hilbert transform of the gains, which we do (since this
       * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
       * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
       * "moment" of the LPCs in this filter. */
      s->dct.dct_calc(&s->dct, lpcs);
      s->dst.dct_calc(&s->dst, lpcs);
  
      /* Split out the coefficient indexes into phase/magnitude pairs */
      idx = 255 + av_clip(lpcs[64],               -255, 255);
      coeffs[0]  = coeffs[0]  * s->cos[idx];
      idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
      last_coeff = coeffs[64] * s->cos[idx];
      for (n = 63;; n--) {
          idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
          coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
          coeffs[n * 2]     = coeffs[n] * s->cos[idx];
  
          if (!--n) break;
  
          idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
          coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
          coeffs[n * 2]     = coeffs[n] * s->cos[idx];
      }
      coeffs[1] = last_coeff;
  
      /* move into real domain */
      s->irdft.rdft_calc(&s->irdft, coeffs);
  
      /* tilt correction and normalize scale */
      memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
      if (s->denoise_tilt_corr) {
          float tilt_mem = 0;
  
          coeffs[remainder - 1] = 0;
          ff_tilt_compensation(&tilt_mem,
                               -1.8 * tilt_factor(coeffs, remainder - 1),
                               coeffs, remainder);
      }
      sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
                                                                 remainder));
      for (n = 0; n < remainder; n++)
          coeffs[n] *= sq;
  }
  
  /**
   * This function applies a Wiener filter on the (noisy) speech signal as
   * a means to denoise it.
   *
   * - take RDFT of LPCs to get the power spectrum of the noise + speech;
   * - using this power spectrum, calculate (for each frequency) the Wiener
   *    filter gain, which depends on the frequency power and desired level
   *    of noise subtraction (when set too high, this leads to artifacts)
   *    We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
   *    of 4-8kHz);
   * - by doing a phase shift, calculate the Hilbert transform of this array
   *    of per-frequency filter-gains to get the filtering coefficients;
   * - smoothen/normalize/de-tilt these filter coefficients as desired;
   * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
   *    to get the denoised speech signal;
   * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
   *    the frame boundary) are saved and applied to subsequent frames by an
   *    overlap-add method (otherwise you get clicking-artifacts).
   *
   * @param s WMA Voice decoding context
   * @param fcb_type Frame (codebook) type
   * @param synth_pf input: the noisy speech signal, output: denoised speech
   *                 data; should be 16-byte aligned (for ASM purposes)
   * @param size size of the speech data
   * @param lpcs LPCs used to synthesize this frame's speech data
   */
  static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
                             float *synth_pf, int size,
                             const float *lpcs)
  {
      int remainder, lim, n;
  
      if (fcb_type != FCB_TYPE_SILENCE) {
          float *tilted_lpcs = s->tilted_lpcs_pf,
                *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
  
          tilted_lpcs[0]           = 1.0;
          memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
          memset(&tilted_lpcs[s->lsps + 1], 0,
                 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
          ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
                               tilted_lpcs, s->lsps + 2);
  
          /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
           * size is applied to the next frame. All input beyond this is zero,
           * and thus all output beyond this will go towards zero, hence we can
           * limit to min(size-1, 127-size) as a performance consideration. */
          remainder = FFMIN(127 - size, size - 1);
          calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
  
          /* apply coefficients (in frequency spectrum domain), i.e. complex
           * number multiplication */
          memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
          s->rdft.rdft_calc(&s->rdft, synth_pf);
          s->rdft.rdft_calc(&s->rdft, coeffs);
          synth_pf[0] *= coeffs[0];
          synth_pf[1] *= coeffs[1];
          for (n = 1; n < 64; n++) {
              float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
              synth_pf[n * 2]     = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
              synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
          }
          s->irdft.rdft_calc(&s->irdft, synth_pf);
      }
  
      /* merge filter output with the history of previous runs */
      if (s->denoise_filter_cache_size) {
          lim = FFMIN(s->denoise_filter_cache_size, size);
          for (n = 0; n < lim; n++)
              synth_pf[n] += s->denoise_filter_cache[n];
          s->denoise_filter_cache_size -= lim;
          memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
                  sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
      }
  
      /* move remainder of filter output into a cache for future runs */
      if (fcb_type != FCB_TYPE_SILENCE) {
          lim = FFMIN(remainder, s->denoise_filter_cache_size);
          for (n = 0; n < lim; n++)
              s->denoise_filter_cache[n] += synth_pf[size + n];
          if (lim < remainder) {
              memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
                     sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
              s->denoise_filter_cache_size = remainder;
          }
      }
  }
  
  /**
   * Averaging projection filter, the postfilter used in WMAVoice.
   *
   * This uses the following steps:
   * - A zero-synthesis filter (generate excitation from synth signal)
   * - Kalman smoothing on excitation, based on pitch
   * - Re-synthesized smoothened output
   * - Iterative Wiener denoise filter
   * - Adaptive gain filter
   * - DC filter
   *
   * @param s WMAVoice decoding context
   * @param synth Speech synthesis output (before postfilter)
   * @param samples Output buffer for filtered samples
   * @param size Buffer size of synth & samples
   * @param lpcs Generated LPCs used for speech synthesis
   * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
   * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
   * @param pitch Pitch of the input signal
   */
  static void postfilter(WMAVoiceContext *s, const float *synth,
                         float *samples,    int size,
                         const float *lpcs, float *zero_exc_pf,
                         int fcb_type,      int pitch)
  {
      float synth_filter_in_buf[MAX_FRAMESIZE / 2],
            *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
            *synth_filter_in = zero_exc_pf;
  
      av_assert0(size <= MAX_FRAMESIZE / 2);
  
      /* generate excitation from input signal */
      ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
  
      if (fcb_type >= FCB_TYPE_AW_PULSES &&
          !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
          synth_filter_in = synth_filter_in_buf;
  
      /* re-synthesize speech after smoothening, and keep history */
      ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
                                   synth_filter_in, size, s->lsps);
      memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
             sizeof(synth_pf[0]) * s->lsps);
  
      wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
  
      adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
                            &s->postfilter_agc);
  
      if (s->dc_level > 8) {
          /* remove ultra-low frequency DC noise / highpass filter;
           * coefficients are identical to those used in SIPR decoding,
           * and very closely resemble those used in AMR-NB decoding. */
          ff_acelp_apply_order_2_transfer_function(samples, samples,
              (const float[2]) { -1.99997,      1.0 },
              (const float[2]) { -1.9330735188, 0.93589198496 },
              0.93980580475, s->dcf_mem, size);
      }
  }
  /**
   * @}
   */
  
  /**
   * Dequantize LSPs
   * @param lsps output pointer to the array that will hold the LSPs
   * @param num number of LSPs to be dequantized
   * @param values quantized values, contains n_stages values
   * @param sizes range (i.e. max value) of each quantized value
   * @param n_stages number of dequantization runs
   * @param table dequantization table to be used
   * @param mul_q LSF multiplier
   * @param base_q base (lowest) LSF values
   */
  static void dequant_lsps(double *lsps, int num,
                           const uint16_t *values,
                           const uint16_t *sizes,
                           int n_stages, const uint8_t *table,
                           const double *mul_q,
                           const double *base_q)
  {
      int n, m;
  
      memset(lsps, 0, num * sizeof(*lsps));
      for (n = 0; n < n_stages; n++) {
          const uint8_t *t_off = &table[values[n] * num];
          double base = base_q[n], mul = mul_q[n];
  
          for (m = 0; m < num; m++)
              lsps[m] += base + mul * t_off[m];
  
          table += sizes[n] * num;
      }
  }
  
  /**
   * @name LSP dequantization routines
   * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
   * @note we assume enough bits are available, caller should check.
   * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
   * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
   * @{
   */
  /**
   * Parse 10 independently-coded LSPs.
   */
  static void dequant_lsp10i(GetBitContext *gb, double *lsps)
  {
      static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
      static const double mul_lsf[4] = {
          5.2187144800e-3,    1.4626986422e-3,
          9.6179549166e-4,    1.1325736225e-3
      };
      static const double base_lsf[4] = {
          M_PI * -2.15522e-1, M_PI * -6.1646e-2,
          M_PI * -3.3486e-2,  M_PI * -5.7408e-2
      };
      uint16_t v[4];
  
      v[0] = get_bits(gb, 8);
      v[1] = get_bits(gb, 6);
      v[2] = get_bits(gb, 5);
      v[3] = get_bits(gb, 5);
  
      dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
                   mul_lsf, base_lsf);
  }
  
  /**
   * Parse 10 independently-coded LSPs, and then derive the tables to
   * generate LSPs for the other frames from them (residual coding).
   */
  static void dequant_lsp10r(GetBitContext *gb,
                             double *i_lsps, const double *old,
                             double *a1, double *a2, int q_mode)
  {
      static const uint16_t vec_sizes[3] = { 128, 64, 64 };
      static const double mul_lsf[3] = {
          2.5807601174e-3,    1.2354460219e-3,   1.1763821673e-3
      };
      static const double base_lsf[3] = {
          M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
      };
      const float (*ipol_tab)[2][10] = q_mode ?
          wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
      uint16_t interpol, v[3];
      int n;
  
      dequant_lsp10i(gb, i_lsps);
  
      interpol = get_bits(gb, 5);
      v[0]     = get_bits(gb, 7);
      v[1]     = get_bits(gb, 6);
      v[2]     = get_bits(gb, 6);
  
      for (n = 0; n < 10; n++) {
          double delta = old[n] - i_lsps[n];
          a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
          a1[10 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
      }
  
      dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
                   mul_lsf, base_lsf);
  }
  
  /**
   * Parse 16 independently-coded LSPs.
   */
  static void dequant_lsp16i(GetBitContext *gb, double *lsps)
  {
      static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
      static const double mul_lsf[5] = {
          3.3439586280e-3,    6.9908173703e-4,
          3.3216608306e-3,    1.0334960326e-3,
          3.1899104283e-3
      };
      static const double base_lsf[5] = {
          M_PI * -1.27576e-1, M_PI * -2.4292e-2,
          M_PI * -1.28094e-1, M_PI * -3.2128e-2,
          M_PI * -1.29816e-1
      };
      uint16_t v[5];
  
      v[0] = get_bits(gb, 8);
      v[1] = get_bits(gb, 6);
      v[2] = get_bits(gb, 7);
      v[3] = get_bits(gb, 6);
      v[4] = get_bits(gb, 7);
  
      dequant_lsps( lsps,     5,  v,     vec_sizes,    2,
                   wmavoice_dq_lsp16i1,  mul_lsf,     base_lsf);
      dequant_lsps(&lsps[5],  5, &v[2], &vec_sizes[2], 2,
                   wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
      dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
                   wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
  }
  
  /**
   * Parse 16 independently-coded LSPs, and then derive the tables to
   * generate LSPs for the other frames from them (residual coding).
   */
  static void dequant_lsp16r(GetBitContext *gb,
                             double *i_lsps, const double *old,
                             double *a1, double *a2, int q_mode)
  {
      static const uint16_t vec_sizes[3] = { 128, 128, 128 };
      static const double mul_lsf[3] = {
          1.2232979501e-3,   1.4062241527e-3,   1.6114744851e-3
      };
      static const double base_lsf[3] = {
          M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
      };
      const float (*ipol_tab)[2][16] = q_mode ?
          wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
      uint16_t interpol, v[3];
      int n;
  
      dequant_lsp16i(gb, i_lsps);
  
      interpol = get_bits(gb, 5);
      v[0]     = get_bits(gb, 7);
      v[1]     = get_bits(gb, 7);
      v[2]     = get_bits(gb, 7);
  
      for (n = 0; n < 16; n++) {
          double delta = old[n] - i_lsps[n];
          a1[n]        = ipol_tab[interpol][0][n] * delta + i_lsps[n];
          a1[16 + n]   = ipol_tab[interpol][1][n] * delta + i_lsps[n];
      }
  
      dequant_lsps( a2,     10,  v,     vec_sizes,    1,
                   wmavoice_dq_lsp16r1,  mul_lsf,     base_lsf);
      dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
                   wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
      dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
                   wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
  }
  
  /**
   * @}
   * @name Pitch-adaptive window coding functions
   * The next few functions are for pitch-adaptive window coding.
   * @{
   */
  /**
   * Parse the offset of the first pitch-adaptive window pulses, and
   * the distribution of pulses between the two blocks in this frame.
   * @param s WMA Voice decoding context private data
   * @param gb bit I/O context
   * @param pitch pitch for each block in this frame
   */
  static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
                              const int *pitch)
  {
      static const int16_t start_offset[94] = {
          -11,  -9,  -7,  -5,  -3,  -1,   1,   3,   5,   7,   9,  11,
           13,  15,  18,  17,  19,  20,  21,  22,  23,  24,  25,  26,
           27,  28,  29,  30,  31,  32,  33,  35,  37,  39,  41,  43,
           45,  47,  49,  51,  53,  55,  57,  59,  61,  63,  65,  67,
           69,  71,  73,  75,  77,  79,  81,  83,  85,  87,  89,  91,
           93,  95,  97,  99, 101, 103, 105, 107, 109, 111, 113, 115,
          117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
          141, 143, 145, 147, 149, 151, 153, 155, 157, 159
      };
      int bits, offset;
  
      /* position of pulse */
      s->aw_idx_is_ext = 0;
      if ((bits = get_bits(gb, 6)) >= 54) {
          s->aw_idx_is_ext = 1;
          bits += (bits - 54) * 3 + get_bits(gb, 2);
      }
  
      /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
       * the distribution of the pulses in each block contained in this frame. */
      s->aw_pulse_range        = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
      for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
      s->aw_n_pulses[0]        = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
      s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
      offset                  += s->aw_n_pulses[0] * pitch[0];
      s->aw_n_pulses[1]        = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
      s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
  
      /* if continuing from a position before the block, reset position to
       * start of block (when corrected for the range over which it can be
       * spread in aw_pulse_set1()). */
      if (start_offset[bits] < MAX_FRAMESIZE / 2) {
          while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
              s->aw_first_pulse_off[1] -= pitch[1];
          if (start_offset[bits] < 0)
              while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
                  s->aw_first_pulse_off[0] -= pitch[0];
      }
  }
  
  /**
   * Apply second set of pitch-adaptive window pulses.
   * @param s WMA Voice decoding context private data
   * @param gb bit I/O context
   * @param block_idx block index in frame [0, 1]
   * @param fcb structure containing fixed codebook vector info
   */
  static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
                            int block_idx, AMRFixed *fcb)
  {
      uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
      uint16_t *use_mask = use_mask_mem + 2;
      /* in this function, idx is the index in the 80-bit (+ padding) use_mask
       * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
       * of idx are the position of the bit within a particular item in the
       * array (0 being the most significant bit, and 15 being the least
       * significant bit), and the remainder (>> 4) is the index in the
       * use_mask[]-array. This is faster and uses less memory than using a
       * 80-byte/80-int array. */
      int pulse_off = s->aw_first_pulse_off[block_idx],
          pulse_start, n, idx, range, aidx, start_off = 0;
  
      /* set offset of first pulse to within this block */
      if (s->aw_n_pulses[block_idx] > 0)
          while (pulse_off + s->aw_pulse_range < 1)
              pulse_off += fcb->pitch_lag;
  
      /* find range per pulse */
      if (s->aw_n_pulses[0] > 0) {
          if (block_idx == 0) {
              range = 32;
          } else /* block_idx = 1 */ {
              range = 8;
              if (s->aw_n_pulses[block_idx] > 0)
                  pulse_off = s->aw_next_pulse_off_cache;
          }
      } else
          range = 16;
      pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
  
      /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
       * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
       * we exclude that range from being pulsed again in this function. */
      memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
      memset( use_mask,   -1, 5 * sizeof(use_mask[0]));
      memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
      if (s->aw_n_pulses[block_idx] > 0)
          for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
              int excl_range         = s->aw_pulse_range; // always 16 or 24
              uint16_t *use_mask_ptr = &use_mask[idx >> 4];
              int first_sh           = 16 - (idx & 15);
              *use_mask_ptr++       &= 0xFFFFu << first_sh;
              excl_range            -= first_sh;
              if (excl_range >= 16) {
                  *use_mask_ptr++    = 0;
                  *use_mask_ptr     &= 0xFFFF >> (excl_range - 16);
              } else
                  *use_mask_ptr     &= 0xFFFF >> excl_range;
          }
  
      /* find the 'aidx'th offset that is not excluded */
      aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
      for (n = 0; n <= aidx; pulse_start++) {
          for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
          if (idx >= MAX_FRAMESIZE / 2) { // find from zero
              if (use_mask[0])      idx = 0x0F;
              else if (use_mask[1]) idx = 0x1F;
              else if (use_mask[2]) idx = 0x2F;
              else if (use_mask[3]) idx = 0x3F;
              else if (use_mask[4]) idx = 0x4F;
              else                  return;
              idx -= av_log2_16bit(use_mask[idx >> 4]);
          }
          if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
              use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
              n++;
              start_off = idx;
          }
      }
  
      fcb->x[fcb->n] = start_off;
      fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
      fcb->n++;
  
      /* set offset for next block, relative to start of that block */
      n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
      s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
  }
  
  /**
   * Apply first set of pitch-adaptive window pulses.
   * @param s WMA Voice decoding context private data
   * @param gb bit I/O context
   * @param block_idx block index in frame [0, 1]
   * @param fcb storage location for fixed codebook pulse info
   */
  static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
                            int block_idx, AMRFixed *fcb)
  {
      int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
      float v;
  
      if (s->aw_n_pulses[block_idx] > 0) {
          int n, v_mask, i_mask, sh, n_pulses;
  
          if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
              n_pulses = 3;
              v_mask   = 8;
              i_mask   = 7;
              sh       = 4;
          } else { // 4 pulses, 1:sign + 2:index each
              n_pulses = 4;
              v_mask   = 4;
              i_mask   = 3;
              sh       = 3;
          }
  
          for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
              fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
              fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
                                   s->aw_first_pulse_off[block_idx];
              while (fcb->x[fcb->n] < 0)
                  fcb->x[fcb->n] += fcb->pitch_lag;
              if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
                  fcb->n++;
          }
      } else {
          int num2 = (val & 0x1FF) >> 1, delta, idx;
  
          if (num2 < 1 * 79)      { delta = 1; idx = num2 + 1; }
          else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
          else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
          else                    { delta = 7; idx = num2 + 1 - 3 * 75; }
          v = (val & 0x200) ? -1.0 : 1.0;
  
          fcb->no_repeat_mask |= 3 << fcb->n;
          fcb->x[fcb->n]       = idx - delta;
          fcb->y[fcb->n]       = v;
          fcb->x[fcb->n + 1]   = idx;
          fcb->y[fcb->n + 1]   = (val & 1) ? -v : v;
          fcb->n              += 2;
      }
  }
  
  /**
   * @}
   *
   * Generate a random number from frame_cntr and block_idx, which will lief
   * in the range [0, 1000 - block_size] (so it can be used as an index in a
   * table of size 1000 of which you want to read block_size entries).
   *
   * @param frame_cntr current frame number
   * @param block_num current block index
   * @param block_size amount of entries we want to read from a table
   *                   that has 1000 entries
   * @return a (non-)random number in the [0, 1000 - block_size] range.
   */
  static int pRNG(int frame_cntr, int block_num, int block_size)
  {
      /* array to simplify the calculation of z:
       * y = (x % 9) * 5 + 6;
       * z = (49995 * x) / y;
       * Since y only has 9 values, we can remove the division by using a
       * LUT and using FASTDIV-style divisions. For each of the 9 values
       * of y, we can rewrite z as:
       * z = x * (49995 / y) + x * ((49995 % y) / y)
       * In this table, each col represents one possible value of y, the
       * first number is 49995 / y, and the second is the FASTDIV variant
       * of 49995 % y / y. */
      static const unsigned int div_tbl[9][2] = {
          { 8332,  3 * 715827883U }, // y =  6
          { 4545,  0 * 390451573U }, // y = 11
          { 3124, 11 * 268435456U }, // y = 16
          { 2380, 15 * 204522253U }, // y = 21
          { 1922, 23 * 165191050U }, // y = 26
          { 1612, 23 * 138547333U }, // y = 31
          { 1388, 27 * 119304648U }, // y = 36
          { 1219, 16 * 104755300U }, // y = 41
          { 1086, 39 *  93368855U }  // y = 46
      };
      unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
      if (x >= 0xFFFF) x -= 0xFFFF;   // max value of x is 8*1877+0xFFFE=0x13AA6,
                                      // so this is effectively a modulo (%)
      y = x - 9 * MULH(477218589, x); // x % 9
      z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
                                      // z = x * 49995 / (y * 5 + 6)
      return z % (1000 - block_size);
  }
  
  /**
   * Parse hardcoded signal for a single block.
   * @note see #synth_block().
   */
  static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
                                   int block_idx, int size,
                                   const struct frame_type_desc *frame_desc,
                                   float *excitation)
  {
      float gain;
      int n, r_idx;
  
      av_assert0(size <= MAX_FRAMESIZE);
  
      /* Set the offset from which we start reading wmavoice_std_codebook */
      if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
          r_idx = pRNG(s->frame_cntr, block_idx, size);
          gain  = s->silence_gain;
      } else /* FCB_TYPE_HARDCODED */ {
          r_idx = get_bits(gb, 8);
          gain  = wmavoice_gain_universal[get_bits(gb, 6)];
      }
  
      /* Clear gain prediction parameters */
      memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
  
      /* Apply gain to hardcoded codebook and use that as excitation signal */
      for (n = 0; n < size; n++)
          excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
  }
  
  /**
   * Parse FCB/ACB signal for a single block.
   * @note see #synth_block().
   */
  static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
                                  int block_idx, int size,
                                  int block_pitch_sh2,
                                  const struct frame_type_desc *frame_desc,
                                  float *excitation)
  {
      static const float gain_coeff[6] = {
          0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
      };
      float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
      int n, idx, gain_weight;
      AMRFixed fcb;
  
      av_assert0(size <= MAX_FRAMESIZE / 2);
      memset(pulses, 0, sizeof(*pulses) * size);
  
      fcb.pitch_lag      = block_pitch_sh2 >> 2;
      fcb.pitch_fac      = 1.0;
      fcb.no_repeat_mask = 0;
      fcb.n              = 0;
  
      /* For the other frame types, this is where we apply the innovation
       * (fixed) codebook pulses of the speech signal. */
      if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
          aw_pulse_set1(s, gb, block_idx, &fcb);
          aw_pulse_set2(s, gb, block_idx, &fcb);
      } else /* FCB_TYPE_EXC_PULSES */ {
          int offset_nbits = 5 - frame_desc->log_n_blocks;
  
          fcb.no_repeat_mask = -1;
          /* similar to ff_decode_10_pulses_35bits(), but with single pulses
           * (instead of double) for a subset of pulses */
          for (n = 0; n < 5; n++) {
              float sign;
              int pos1, pos2;
  
              sign           = get_bits1(gb) ? 1.0 : -1.0;
              pos1           = get_bits(gb, offset_nbits);
              fcb.x[fcb.n]   = n + 5 * pos1;
              fcb.y[fcb.n++] = sign;
              if (n < frame_desc->dbl_pulses) {
                  pos2           = get_bits(gb, offset_nbits);
                  fcb.x[fcb.n]   = n + 5 * pos2;
                  fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
              }
          }
      }
      ff_set_fixed_vector(pulses, &fcb, 1.0, size);
  
      /* Calculate gain for adaptive & fixed codebook signal.
       * see ff_amr_set_fixed_gain(). */
      idx = get_bits(gb, 7);
      fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
                                                   gain_coeff, 6) -
                      5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
      acb_gain = wmavoice_gain_codebook_acb[idx];
      pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
                          -2.9957322736 /* log(0.05) */,
                           1.6094379124 /* log(5.0)  */);
  
      gain_weight = 8 >> frame_desc->log_n_blocks;
      memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
              sizeof(*s->gain_pred_err) * (6 - gain_weight));
      for (n = 0; n < gain_weight; n++)
          s->gain_pred_err[n] = pred_err;
  
      /* Calculation of adaptive codebook */
      if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
          int len;
          for (n = 0; n < size; n += len) {
              int next_idx_sh16;
              int abs_idx    = block_idx * size + n;
              int pitch_sh16 = (s->last_pitch_val << 16) +
                               s->pitch_diff_sh16 * abs_idx;
              int pitch      = (pitch_sh16 + 0x6FFF) >> 16;
              int idx_sh16   = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
              idx            = idx_sh16 >> 16;
              if (s->pitch_diff_sh16) {
                  if (s->pitch_diff_sh16 > 0) {
                      next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
                  } else
                      next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
                  len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
                                1, size - n);
              } else
                  len = size;
  
              ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
                                    wmavoice_ipol1_coeffs, 17,
                                    idx, 9, len);
          }
      } else /* ACB_TYPE_HAMMING */ {
          int block_pitch = block_pitch_sh2 >> 2;
          idx             = block_pitch_sh2 & 3;
          if (idx) {
              ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
                                    wmavoice_ipol2_coeffs, 4,
                                    idx, 8, size);
          } else
              av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
                                sizeof(float) * size);
      }
  
      /* Interpolate ACB/FCB and use as excitation signal */
      ff_weighted_vector_sumf(excitation, excitation, pulses,
                              acb_gain, fcb_gain, size);
  }
  
  /**
   * Parse data in a single block.
   * @note we assume enough bits are available, caller should check.
   *
   * @param s WMA Voice decoding context private data
   * @param gb bit I/O context
   * @param block_idx index of the to-be-read block
   * @param size amount of samples to be read in this block
   * @param block_pitch_sh2 pitch for this block << 2
   * @param lsps LSPs for (the end of) this frame
   * @param prev_lsps LSPs for the last frame
   * @param frame_desc frame type descriptor
   * @param excitation target memory for the ACB+FCB interpolated signal
   * @param synth target memory for the speech synthesis filter output
   * @return 0 on success, <0 on error.
   */
  static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
                          int block_idx, int size,
                          int block_pitch_sh2,
                          const double *lsps, const double *prev_lsps,
                          const struct frame_type_desc *frame_desc,
                          float *excitation, float *synth)
  {
      double i_lsps[MAX_LSPS];
      float lpcs[MAX_LSPS];
      float fac;
      int n;
  
      if (frame_desc->acb_type == ACB_TYPE_NONE)
          synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
      else
          synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
                              frame_desc, excitation);
  
      /* convert interpolated LSPs to LPCs */
      fac = (block_idx + 0.5) / frame_desc->n_blocks;
      for (n = 0; n < s->lsps; n++) // LSF -> LSP
          i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
      ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
  
      /* Speech synthesis */
      ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
  }
  
  /**
   * Synthesize output samples for a single frame.
   * @note we assume enough bits are available, caller should check.
   *
   * @param ctx WMA Voice decoder context
   * @param gb bit I/O context (s->gb or one for cross-packet superframes)
   * @param frame_idx Frame number within superframe [0-2]
   * @param samples pointer to output sample buffer, has space for at least 160
   *                samples
   * @param lsps LSP array
   * @param prev_lsps array of previous frame's LSPs
   * @param excitation target buffer for excitation signal
   * @param synth target buffer for synthesized speech data
   * @return 0 on success, <0 on error.
   */
  static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
                         float *samples,
                         const double *lsps, const double *prev_lsps,
                         float *excitation, float *synth)
  {
      WMAVoiceContext *s = ctx->priv_data;
      int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val);
      int pitch[MAX_BLOCKS], av_uninit(last_block_pitch);
  
      /* Parse frame type ("frame header"), see frame_descs */
      int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
  
      if (bd_idx < 0) {
          av_log(ctx, AV_LOG_ERROR,
                 "Invalid frame type VLC code, skipping\n");
          return -1;
      }
  
      block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
  
      /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
      if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
          /* Pitch is provided per frame, which is interpreted as the pitch of
           * the last sample of the last block of this frame. We can interpolate
           * the pitch of other blocks (and even pitch-per-sample) by gradually
           * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
          n_blocks_x2      = frame_descs[bd_idx].n_blocks << 1;
          log_n_blocks_x2  = frame_descs[bd_idx].log_n_blocks + 1;
          cur_pitch_val    = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
          cur_pitch_val    = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
          if (s->last_acb_type == ACB_TYPE_NONE ||
              20 * abs(cur_pitch_val - s->last_pitch_val) >
                  (cur_pitch_val + s->last_pitch_val))
              s->last_pitch_val = cur_pitch_val;
  
          /* pitch per block */
          for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
              int fac = n * 2 + 1;
  
              pitch[n] = (MUL16(fac,                 cur_pitch_val) +
                          MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
                          frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
          }
  
          /* "pitch-diff-per-sample" for calculation of pitch per sample */
          s->pitch_diff_sh16 =
              ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
      }
  
      /* Global gain (if silence) and pitch-adaptive window coordinates */
      switch (frame_descs[bd_idx].fcb_type) {
      case FCB_TYPE_SILENCE:
          s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
          break;
      case FCB_TYPE_AW_PULSES:
          aw_parse_coords(s, gb, pitch);
          break;
      }
  
      for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
          int bl_pitch_sh2;
  
          /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
          switch (frame_descs[bd_idx].acb_type) {
          case ACB_TYPE_HAMMING: {
              /* Pitch is given per block. Per-block pitches are encoded as an
               * absolute value for the first block, and then delta values
               * relative to this value) for all subsequent blocks. The scale of
               * this pitch value is semi-logaritmic compared to its use in the
               * decoder, so we convert it to normal scale also. */
              int block_pitch,
                  t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
                  t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
                  t3 =  s->block_conv_table[3] - s->block_conv_table[2] + 1;
  
              if (n == 0) {
                  block_pitch = get_bits(gb, s->block_pitch_nbits);
              } else
                  block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
                                   get_bits(gb, s->block_delta_pitch_nbits);
              /* Convert last_ so that any next delta is within _range */
              last_block_pitch = av_clip(block_pitch,
                                         s->block_delta_pitch_hrange,
                                         s->block_pitch_range -
                                             s->block_delta_pitch_hrange);
  
              /* Convert semi-log-style scale back to normal scale */
              if (block_pitch < t1) {
                  bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
              } else {
                  block_pitch -= t1;
                  if (block_pitch < t2) {
                      bl_pitch_sh2 =
                          (s->block_conv_table[1] << 2) + (block_pitch << 1);
                  } else {
                      block_pitch -= t2;
                      if (block_pitch < t3) {
                          bl_pitch_sh2 =
                              (s->block_conv_table[2] + block_pitch) << 2;
                      } else
                          bl_pitch_sh2 = s->block_conv_table[3] << 2;
                  }
              }
              pitch[n] = bl_pitch_sh2 >> 2;
              break;
          }
  
          case ACB_TYPE_ASYMMETRIC: {
              bl_pitch_sh2 = pitch[n] << 2;
              break;
          }
  
          default: // ACB_TYPE_NONE has no pitch
              bl_pitch_sh2 = 0;
              break;
          }
  
          synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
                      lsps, prev_lsps, &frame_descs[bd_idx],
                      &excitation[n * block_nsamples],
                      &synth[n * block_nsamples]);
      }
  
      /* Averaging projection filter, if applicable. Else, just copy samples
       * from synthesis buffer */
      if (s->do_apf) {
          double i_lsps[MAX_LSPS];
          float lpcs[MAX_LSPS];
  
          for (n = 0; n < s->lsps; n++) // LSF -> LSP
              i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
          ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
          postfilter(s, synth, samples, 80, lpcs,
                     &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
                     frame_descs[bd_idx].fcb_type, pitch[0]);
  
          for (n = 0; n < s->lsps; n++) // LSF -> LSP
              i_lsps[n] = cos(lsps[n]);
          ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
          postfilter(s, &synth[80], &samples[80], 80, lpcs,
                     &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
                     frame_descs[bd_idx].fcb_type, pitch[0]);
      } else
          memcpy(samples, synth, 160 * sizeof(synth[0]));
  
      /* Cache values for next frame */
      s->frame_cntr++;
      if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
      s->last_acb_type = frame_descs[bd_idx].acb_type;
      switch (frame_descs[bd_idx].acb_type) {
      case ACB_TYPE_NONE:
          s->last_pitch_val = 0;
          break;
      case ACB_TYPE_ASYMMETRIC:
          s->last_pitch_val = cur_pitch_val;
          break;
      case ACB_TYPE_HAMMING:
          s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
          break;
      }
  
      return 0;
  }
  
  /**
   * Ensure minimum value for first item, maximum value for last value,
   * proper spacing between each value and proper ordering.
   *
   * @param lsps array of LSPs
   * @param num size of LSP array
   *
   * @note basically a double version of #ff_acelp_reorder_lsf(), might be
   *       useful to put in a generic location later on. Parts are also
   *       present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
   *       which is in float.
   */
  static void stabilize_lsps(double *lsps, int num)
  {
      int n, m, l;
  
      /* set minimum value for first, maximum value for last and minimum
       * spacing between LSF values.
       * Very similar to ff_set_min_dist_lsf(), but in double. */
      lsps[0]       = FFMAX(lsps[0],       0.0015 * M_PI);
      for (n = 1; n < num; n++)
          lsps[n]   = FFMAX(lsps[n],       lsps[n - 1] + 0.0125 * M_PI);
      lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
  
      /* reorder (looks like one-time / non-recursed bubblesort).
       * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
      for (n = 1; n < num; n++) {
          if (lsps[n] < lsps[n - 1]) {
              for (m = 1; m < num; m++) {
                  double tmp = lsps[m];
                  for (l = m - 1; l >= 0; l--) {
                      if (lsps[l] <= tmp) break;
                      lsps[l + 1] = lsps[l];
                  }
                  lsps[l + 1] = tmp;
              }
              break;
          }
      }
  }
  
  /**
   * Test if there's enough bits to read 1 superframe.
   *
   * @param orig_gb bit I/O context used for reading. This function
   *                does not modify the state of the bitreader; it
   *                only uses it to copy the current stream position
   * @param s WMA Voice decoding context private data
   * @return -1 if unsupported, 1 on not enough bits or 0 if OK.
   */
  static int check_bits_for_superframe(GetBitContext *orig_gb,
                                       WMAVoiceContext *s)
  {
      GetBitContext s_gb, *gb = &s_gb;
      int n, need_bits, bd_idx;
      const struct frame_type_desc *frame_desc;
  
      /* initialize a copy */
      init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
      skip_bits_long(gb, get_bits_count(orig_gb));
      av_assert1(get_bits_left(gb) == get_bits_left(orig_gb));
  
      /* superframe header */
      if (get_bits_left(gb) < 14)
          return 1;
      if (!get_bits1(gb))
          return -1;                        // WMAPro-in-WMAVoice superframe
      if (get_bits1(gb)) skip_bits(gb, 12); // number of  samples in superframe
      if (s->has_residual_lsps) {           // residual LSPs (for all frames)
          if (get_bits_left(gb) < s->sframe_lsp_bitsize)
              return 1;
          skip_bits_long(gb, s->sframe_lsp_bitsize);
      }
  
      /* frames */
      for (n = 0; n < MAX_FRAMES; n++) {
          int aw_idx_is_ext = 0;
  
          if (!s->has_residual_lsps) {     // independent LSPs (per-frame)
             if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
             skip_bits_long(gb, s->frame_lsp_bitsize);
          }
          bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
          if (bd_idx < 0)
              return -1;                   // invalid frame type VLC code
          frame_desc = &frame_descs[bd_idx];
          if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
              if (get_bits_left(gb) < s->pitch_nbits)
                  return 1;
              skip_bits_long(gb, s->pitch_nbits);
          }
          if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
              skip_bits(gb, 8);
          } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
              int tmp = get_bits(gb, 6);
              if (tmp >= 0x36) {
                  skip_bits(gb, 2);
                  aw_idx_is_ext = 1;
              }
          }
  
          /* blocks */
          if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
              need_bits = s->block_pitch_nbits +
                  (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
          } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
              need_bits = 2 * !aw_idx_is_ext;
          } else
              need_bits = 0;
          need_bits += frame_desc->frame_size;
          if (get_bits_left(gb) < need_bits)
              return 1;
          skip_bits_long(gb, need_bits);
      }
  
      return 0;
  }
  
  /**
   * Synthesize output samples for a single superframe. If we have any data
   * cached in s->sframe_cache, that will be used instead of whatever is loaded
   * in s->gb.
   *
   * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
   * to give a total of 480 samples per frame. See #synth_frame() for frame
   * parsing. In addition to 3 frames, superframes can also contain the LSPs
   * (if these are globally specified for all frames (residually); they can
   * also be specified individually per-frame. See the s->has_residual_lsps
   * option), and can specify the number of samples encoded in this superframe
   * (if less than 480), usually used to prevent blanks at track boundaries.
   *
   * @param ctx WMA Voice decoder context
   * @return 0 on success, <0 on error or 1 if there was not enough data to
   *         fully parse the superframe
   */
  static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
                              int *got_frame_ptr)
  {
      WMAVoiceContext *s = ctx->priv_data;
      GetBitContext *gb = &s->gb, s_gb;
      int n, res, n_samples = 480;
      double lsps[MAX_FRAMES][MAX_LSPS];
      const double *mean_lsf = s->lsps == 16 ?
          wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
      float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
      float synth[MAX_LSPS + MAX_SFRAMESIZE];
      float *samples;
  
      memcpy(synth,      s->synth_history,
             s->lsps             * sizeof(*synth));
      memcpy(excitation, s->excitation_history,
             s->history_nsamples * sizeof(*excitation));
  
      if (s->sframe_cache_size > 0) {
          gb = &s_gb;
          init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
          s->sframe_cache_size = 0;
      }
  
      if ((res = check_bits_for_superframe(gb, s)) == 1) {
          *got_frame_ptr = 0;
          return 1;
      }
  
      /* First bit is speech/music bit, it differentiates between WMAVoice
       * speech samples (the actual codec) and WMAVoice music samples, which
       * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
       * the wild yet. */
      if (!get_bits1(gb)) {
          av_log_missing_feature(ctx, "WMAPro-in-WMAVoice", 1);
          return AVERROR_PATCHWELCOME;
      }
  
      /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
      if (get_bits1(gb)) {
          if ((n_samples = get_bits(gb, 12)) > 480) {
              av_log(ctx, AV_LOG_ERROR,
                     "Superframe encodes >480 samples (%d), not allowed\n",
                     n_samples);
              return -1;
          }
      }
      /* Parse LSPs, if global for the superframe (can also be per-frame). */
      if (s->has_residual_lsps) {
          double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
  
          for (n = 0; n < s->lsps; n++)
              prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
  
          if (s->lsps == 10) {
              dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
          } else /* s->lsps == 16 */
              dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
  
          for (n = 0; n < s->lsps; n++) {
              lsps[0][n]  = mean_lsf[n] + (a1[n]           - a2[n * 2]);
              lsps[1][n]  = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
              lsps[2][n] += mean_lsf[n];
          }
          for (n = 0; n < 3; n++)
              stabilize_lsps(lsps[n], s->lsps);
      }
  
      /* get output buffer */
      frame->nb_samples = 480;
      if ((res = ff_get_buffer(ctx, frame)) < 0) {
          av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n");
          return res;
      }
      frame->nb_samples = n_samples;
      samples = (float *)frame->data[0];
  
      /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
      for (n = 0; n < 3; n++) {
          if (!s->has_residual_lsps) {
              int m;
  
              if (s->lsps == 10) {
                  dequant_lsp10i(gb, lsps[n]);
              } else /* s->lsps == 16 */
                  dequant_lsp16i(gb, lsps[n]);
  
              for (m = 0; m < s->lsps; m++)
                  lsps[n][m] += mean_lsf[m];
              stabilize_lsps(lsps[n], s->lsps);
          }
  
          if ((res = synth_frame(ctx, gb, n,
                                 &samples[n * MAX_FRAMESIZE],
                                 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
                                 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
                                 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
              *got_frame_ptr = 0;
              return res;
          }
      }
  
      /* Statistics? FIXME - we don't check for length, a slight overrun
       * will be caught by internal buffer padding, and anything else
       * will be skipped, not read. */
      if (get_bits1(gb)) {
          res = get_bits(gb, 4);
          skip_bits(gb, 10 * (res + 1));
      }
  
      *got_frame_ptr = 1;
  
      /* Update history */
      memcpy(s->prev_lsps,           lsps[2],
             s->lsps             * sizeof(*s->prev_lsps));
      memcpy(s->synth_history,      &synth[MAX_SFRAMESIZE],
             s->lsps             * sizeof(*synth));
      memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
             s->history_nsamples * sizeof(*excitation));
      if (s->do_apf)
          memmove(s->zero_exc_pf,       &s->zero_exc_pf[MAX_SFRAMESIZE],
                  s->history_nsamples * sizeof(*s->zero_exc_pf));
  
      return 0;
  }
  
  /**
   * Parse the packet header at the start of each packet (input data to this
   * decoder).
   *
   * @param s WMA Voice decoding context private data
   * @return 1 if not enough bits were available, or 0 on success.
   */
  static int parse_packet_header(WMAVoiceContext *s)
  {
      GetBitContext *gb = &s->gb;
      unsigned int res;
  
      if (get_bits_left(gb) < 11)
          return 1;
      skip_bits(gb, 4);          // packet sequence number
      s->has_residual_lsps = get_bits1(gb);
      do {
          res = get_bits(gb, 6); // number of superframes per packet
                                 // (minus first one if there is spillover)
          if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
              return 1;
      } while (res == 0x3F);
      s->spillover_nbits   = get_bits(gb, s->spillover_bitsize);
  
      return 0;
  }
  
  /**
   * Copy (unaligned) bits from gb/data/size to pb.
   *
   * @param pb target buffer to copy bits into
   * @param data source buffer to copy bits from
   * @param size size of the source data, in bytes
   * @param gb bit I/O context specifying the current position in the source.
   *           data. This function might use this to align the bit position to
   *           a whole-byte boundary before calling #avpriv_copy_bits() on aligned
   *           source data
   * @param nbits the amount of bits to copy from source to target
   *
   * @note after calling this function, the current position in the input bit
   *       I/O context is undefined.
   */
  static void copy_bits(PutBitContext *pb,
                        const uint8_t *data, int size,
                        GetBitContext *gb, int nbits)
  {
      int rmn_bytes, rmn_bits;
  
      rmn_bits = rmn_bytes = get_bits_left(gb);
      if (rmn_bits < nbits)
          return;
      if (nbits > pb->size_in_bits - put_bits_count(pb))
          return;
      rmn_bits &= 7; rmn_bytes >>= 3;
      if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
          put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
      avpriv_copy_bits(pb, data + size - rmn_bytes,
                   FFMIN(nbits - rmn_bits, rmn_bytes << 3));
  }
  
  /**
   * Packet decoding: a packet is anything that the (ASF) demuxer contains,
   * and we expect that the demuxer / application provides it to us as such
   * (else you'll probably get garbage as output). Every packet has a size of
   * ctx->block_align bytes, starts with a packet header (see
   * #parse_packet_header()), and then a series of superframes. Superframe
   * boundaries may exceed packets, i.e. superframes can split data over
   * multiple (two) packets.
   *
   * For more information about frames, see #synth_superframe().
   */
  static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
                                    int *got_frame_ptr, AVPacket *avpkt)
  {
      WMAVoiceContext *s = ctx->priv_data;
      GetBitContext *gb = &s->gb;
      int size, res, pos;
  
      /* Packets are sometimes a multiple of ctx->block_align, with a packet
       * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
       * feeds us ASF packets, which may concatenate multiple "codec" packets
       * in a single "muxer" packet, so we artificially emulate that by
       * capping the packet size at ctx->block_align. */
      for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
      if (!size) {
          *got_frame_ptr = 0;
          return 0;
      }
      init_get_bits(&s->gb, avpkt->data, size << 3);
  
      /* size == ctx->block_align is used to indicate whether we are dealing with
       * a new packet or a packet of which we already read the packet header
       * previously. */
      if (size == ctx->block_align) { // new packet header
          if ((res = parse_packet_header(s)) < 0)
              return res;
  
          /* If the packet header specifies a s->spillover_nbits, then we want
           * to push out all data of the previous packet (+ spillover) before
           * continuing to parse new superframes in the current packet. */
          if (s->spillover_nbits > 0) {
              if (s->sframe_cache_size > 0) {
                  int cnt = get_bits_count(gb);
                  copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
                  flush_put_bits(&s->pb);
                  s->sframe_cache_size += s->spillover_nbits;
                  if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 &&
                      *got_frame_ptr) {
                      cnt += s->spillover_nbits;
                      s->skip_bits_next = cnt & 7;
                      return cnt >> 3;
                  } else
                      skip_bits_long (gb, s->spillover_nbits - cnt +
                                      get_bits_count(gb)); // resync
              } else
                  skip_bits_long(gb, s->spillover_nbits);  // resync
          }
      } else if (s->skip_bits_next)
          skip_bits(gb, s->skip_bits_next);
  
      /* Try parsing superframes in current packet */
      s->sframe_cache_size = 0;
      s->skip_bits_next = 0;
      pos = get_bits_left(gb);
      if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
          return res;
      } else if (*got_frame_ptr) {
          int cnt = get_bits_count(gb);
          s->skip_bits_next = cnt & 7;
          return cnt >> 3;
      } else if ((s->sframe_cache_size = pos) > 0) {
          /* rewind bit reader to start of last (incomplete) superframe... */
          init_get_bits(gb, avpkt->data, size << 3);
          skip_bits_long(gb, (size << 3) - pos);
          av_assert1(get_bits_left(gb) == pos);
  
          /* ...and cache it for spillover in next packet */
          init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
          copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
          // FIXME bad - just copy bytes as whole and add use the
          // skip_bits_next field
      }
  
      return size;
  }
  
  static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
  {
      WMAVoiceContext *s = ctx->priv_data;
  
      if (s->do_apf) {
          ff_rdft_end(&s->rdft);
          ff_rdft_end(&s->irdft);
          ff_dct_end(&s->dct);
          ff_dct_end(&s->dst);
      }
  
      return 0;
  }
  
  static av_cold void wmavoice_flush(AVCodecContext *ctx)
  {
      WMAVoiceContext *s = ctx->priv_data;
      int n;
  
      s->postfilter_agc    = 0;
      s->sframe_cache_size = 0;
      s->skip_bits_next    = 0;
      for (n = 0; n < s->lsps; n++)
          s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
      memset(s->excitation_history, 0,
             sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
      memset(s->synth_history,      0,
             sizeof(*s->synth_history)      * MAX_LSPS);
      memset(s->gain_pred_err,      0,
             sizeof(s->gain_pred_err));
  
      if (s->do_apf) {
          memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
                 sizeof(*s->synth_filter_out_buf) * s->lsps);
          memset(s->dcf_mem,              0,
                 sizeof(*s->dcf_mem)              * 2);
          memset(s->zero_exc_pf,          0,
                 sizeof(*s->zero_exc_pf)          * s->history_nsamples);
          memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
      }
  }
  
  AVCodec ff_wmavoice_decoder = {
      .name           = "wmavoice",
      .type           = AVMEDIA_TYPE_AUDIO,
      .id             = AV_CODEC_ID_WMAVOICE,
      .priv_data_size = sizeof(WMAVoiceContext),
      .init           = wmavoice_decode_init,
      .close          = wmavoice_decode_end,
      .decode         = wmavoice_decode_packet,
      .capabilities   = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
      .flush          = wmavoice_flush,
      .long_name      = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
  };
  