/*
  * AMR narrowband decoder
  * Copyright (c) 2006-2007 Robert Swain
  * Copyright (c) 2009 Colin McQuillan
  *
  * 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

  * AMR narrowband decoder
  *
  * This decoder uses floats for simplicity and so is not bit-exact. One
  * difference is that differences in phase can accumulate. The test sequences
  * in 3GPP TS 26.074 can still be useful.
  *
  * - Comparing this file's output to the output of the ref decoder gives a
  *   PSNR of 30 to 80. Plotting the output samples shows a difference in
  *   phase in some areas.
  *
  * - Comparing both decoders against their input, this decoder gives a similar
  *   PSNR. If the test sequence homing frames are removed (this decoder does
  *   not detect them), the PSNR is at least as good as the reference on 140
  *   out of 169 tests.
  */
 
 
 #include <string.h>
 #include <math.h>
 

 #include "libavutil/channel_layout.h"

 #include "libavutil/float_dsp.h"

 #include "avcodec.h"
 #include "libavutil/common.h"

 #include "libavutil/avassert.h"

 #include "celp_math.h"
 #include "celp_filters.h"
 #include "acelp_filters.h"
 #include "acelp_vectors.h"
 #include "acelp_pitch_delay.h"
 #include "lsp.h"

 #include "amr.h"

 #include "internal.h"

 
 #include "amrnbdata.h"
 
 #define AMR_BLOCK_SIZE              160   ///< samples per frame
 #define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow
 
 /**
  * Scale from constructed speech to [-1,1]
  *
  * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
  * upscales by two (section 6.2.2).
  *
  * Fundamentally, this scale is determined by energy_mean through
  * the fixed vector contribution to the excitation vector.
  */
 #define AMR_SAMPLE_SCALE  (2.0 / 32768.0)
 
 /** Prediction factor for 12.2kbit/s mode */
 #define PRED_FAC_MODE_12k2             0.65
 
 #define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
 #define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
 #define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode
 
 /** Initial energy in dB. Also used for bad frames (unimplemented). */
 #define MIN_ENERGY -14.0
 
 /** Maximum sharpening factor
  *
  * The specification says 0.8, which should be 13107, but the reference C code

  * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)

  */
 #define SHARP_MAX 0.79449462890625
 
 /** Number of impulse response coefficients used for tilt factor */
 #define AMR_TILT_RESPONSE   22
 /** Tilt factor = 1st reflection coefficient * gamma_t */
 #define AMR_TILT_GAMMA_T   0.8
 /** Adaptive gain control factor used in post-filter */
 #define AMR_AGC_ALPHA      0.9
 
 typedef struct AMRContext {
     AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
     uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
     enum Mode                cur_frame_mode;
 
     int16_t     prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
     double          lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
     double   prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
 
     float         lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
     float          lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
 
     float           lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
 
     uint8_t                   pitch_lag_int; ///< integer part of pitch lag from current subframe
 
     float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
     float                       *excitation; ///< pointer to the current excitation vector in excitation_buf
 
     float   pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
     float   fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
 
     float               prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes
     float                     pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes
     float                     fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
 
     float                              beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
     uint8_t                      diff_count; ///< the number of subframes for which diff has been above 0.65
     uint8_t                      hang_count; ///< the number of subframes since a hangover period started
 
     float            prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset"
     uint8_t               prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none
     uint8_t                 ir_filter_onset; ///< flag for impulse response filter strength
 
     float                postfilter_mem[10]; ///< previous intermediate values in the formant filter
     float                          tilt_mem; ///< previous input to tilt compensation filter
     float                    postfilter_agc; ///< previous factor used for adaptive gain control
     float                  high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
 
     float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
 

     ACELPFContext                     acelpf_ctx; ///< context for filters for ACELP-based codecs
     ACELPVContext                     acelpv_ctx; ///< context for vector operations for ACELP-based codecs
     CELPFContext                       celpf_ctx; ///< context for filters for CELP-based codecs
     CELPMContext                       celpm_ctx; ///< context for fixed point math operations
 

 } AMRContext;
 
 /** Double version of ff_weighted_vector_sumf() */
 static void weighted_vector_sumd(double *out, const double *in_a,
                                  const double *in_b, double weight_coeff_a,
                                  double weight_coeff_b, int length)
 {
     int i;
 
     for (i = 0; i < length; i++)
         out[i] = weight_coeff_a * in_a[i]
                + weight_coeff_b * in_b[i];
 }
 
 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
 {
     AMRContext *p = avctx->priv_data;
     int i;
 

     if (avctx->channels > 1) {

         avpriv_report_missing_feature(avctx, "multi-channel AMR");

         return AVERROR_PATCHWELCOME;
     }
 
     avctx->channels       = 1;
     avctx->channel_layout = AV_CH_LAYOUT_MONO;

     if (!avctx->sample_rate)
         avctx->sample_rate = 8000;

     avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;

 
     // p->excitation always points to the same position in p->excitation_buf
     p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
 
     for (i = 0; i < LP_FILTER_ORDER; i++) {
         p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
         p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
     }
 
     for (i = 0; i < 4; i++)
         p->prediction_error[i] = MIN_ENERGY;
 

     ff_acelp_filter_init(&p->acelpf_ctx);
     ff_acelp_vectors_init(&p->acelpv_ctx);
     ff_celp_filter_init(&p->celpf_ctx);
     ff_celp_math_init(&p->celpm_ctx);
 

     return 0;
 }
 
 
 /**
  * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
  *
  * The order of speech bits is specified by 3GPP TS 26.101.
  *
  * @param p the context
  * @param buf               pointer to the input buffer
  * @param buf_size          size of the input buffer
  *
  * @return the frame mode
  */
 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
                                   int buf_size)
 {
     enum Mode mode;
 
     // Decode the first octet.

     mode = buf[0] >> 3 & 0x0F;                      // frame type
     p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit

     if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) {
         return NO_DATA;
     }
 

     if (mode < MODE_DTX)
         ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
                            amr_unpacking_bitmaps_per_mode[mode]);

 
     return mode;
 }
 
 

 /// @name AMR pitch LPC coefficient decoding functions

 /// @{
 
 /**
  * Interpolate the LSF vector (used for fixed gain smoothing).
  * The interpolation is done over all four subframes even in MODE_12k2.
  *

  * @param[in]     ctx       The Context

  * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
  * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
  */

 static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)

 {
     int i;
 
     for (i = 0; i < 4; i++)

         ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,

                                 0.25 * (3 - i), 0.25 * (i + 1),
                                 LP_FILTER_ORDER);
 }
 
 /**
  * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
  *
  * @param p the context
  * @param lsp output LSP vector
  * @param lsf_no_r LSF vector without the residual vector added
  * @param lsf_quantizer pointers to LSF dictionary tables
  * @param quantizer_offset offset in tables
  * @param sign for the 3 dictionary table
  * @param update store data for computing the next frame's LSFs
  */
 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
                                  const float lsf_no_r[LP_FILTER_ORDER],
                                  const int16_t *lsf_quantizer[5],
                                  const int quantizer_offset,
                                  const int sign, const int update)
 {
     int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
     float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
     int i;
 
     for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
         memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
                2 * sizeof(*lsf_r));
 
     if (sign) {
         lsf_r[4] *= -1;
         lsf_r[5] *= -1;
     }
 
     if (update)

         memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));

 
     for (i = 0; i < LP_FILTER_ORDER; i++)
         lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
 
     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
 
     if (update)

         interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);

     ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);

 }
 
 /**
  * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
  *
  * @param p                 pointer to the AMRContext
  */
 static void lsf2lsp_5(AMRContext *p)
 {
     const uint16_t *lsf_param = p->frame.lsf;
     float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
     const int16_t *lsf_quantizer[5];
     int i;
 
     lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
     lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
     lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
     lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
     lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
 
     for (i = 0; i < LP_FILTER_ORDER; i++)
         lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
 
     lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
     lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
 
     // interpolate LSP vectors at subframes 1 and 3
     weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
     weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
 }
 
 /**
  * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
  *
  * @param p                 pointer to the AMRContext
  */
 static void lsf2lsp_3(AMRContext *p)
 {
     const uint16_t *lsf_param = p->frame.lsf;
     int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
     float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
     const int16_t *lsf_quantizer;
     int i, j;
 
     lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
     memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
 
     lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
     memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
 
     lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
     memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
 
     // calculate mean-removed LSF vector and add mean
     for (i = 0; i < LP_FILTER_ORDER; i++)
         lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0);
 
     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
 
     // store data for computing the next frame's LSFs

     interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q);

     memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
 

     ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);

 
     // interpolate LSP vectors at subframes 1, 2 and 3
     for (i = 1; i <= 3; i++)
         for(j = 0; j < LP_FILTER_ORDER; j++)
             p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
                 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
 }
 
 /// @}
 
 

 /// @name AMR pitch vector decoding functions

 /// @{
 
 /**
  * Like ff_decode_pitch_lag(), but with 1/6 resolution
  */
 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
                                  const int prev_lag_int, const int subframe)
 {
     if (subframe == 0 || subframe == 2) {
         if (pitch_index < 463) {
             *lag_int  = (pitch_index + 107) * 10923 >> 16;
             *lag_frac = pitch_index - *lag_int * 6 + 105;
         } else {
             *lag_int  = pitch_index - 368;
             *lag_frac = 0;
         }
     } else {
         *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
         *lag_frac = pitch_index - *lag_int * 6 - 3;
         *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
                             PITCH_DELAY_MAX - 9);
     }
 }
 
 static void decode_pitch_vector(AMRContext *p,
                                 const AMRNBSubframe *amr_subframe,
                                 const int subframe)
 {
     int pitch_lag_int, pitch_lag_frac;
     enum Mode mode = p->cur_frame_mode;
 
     if (p->cur_frame_mode == MODE_12k2) {
         decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
                              amr_subframe->p_lag, p->pitch_lag_int,
                              subframe);

     } else {

         ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
                             amr_subframe->p_lag,
                             p->pitch_lag_int, subframe,
                             mode != MODE_4k75 && mode != MODE_5k15,
                             mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));

         pitch_lag_frac *= 2;
     }

 
     p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
 
     pitch_lag_int += pitch_lag_frac > 0;
 
     /* Calculate the pitch vector by interpolating the past excitation at the
        pitch lag using a b60 hamming windowed sinc function.   */

     p->acelpf_ctx.acelp_interpolatef(p->excitation,
                           p->excitation + 1 - pitch_lag_int,

                           ff_b60_sinc, 6,
                           pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
                           10, AMR_SUBFRAME_SIZE);
 
     memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
 }
 
 /// @}
 
 

 /// @name AMR algebraic code book (fixed) vector decoding functions

 /// @{
 
 /**
  * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
  */
 static void decode_10bit_pulse(int code, int pulse_position[8],
                                int i1, int i2, int i3)
 {
     // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
     // the 3 pulses and the upper 7 bits being coded in base 5
     const uint8_t *positions = base_five_table[code >> 3];
     pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
     pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
     pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
 }
 
 /**
  * Decode the algebraic codebook index to pulse positions and signs and
  * construct the algebraic codebook vector for MODE_10k2.
  *
  * @param fixed_index          positions of the eight pulses
  * @param fixed_sparse         pointer to the algebraic codebook vector
  */
 static void decode_8_pulses_31bits(const int16_t *fixed_index,
                                    AMRFixed *fixed_sparse)
 {
     int pulse_position[8];
     int i, temp;
 
     decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
     decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
 
     // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
     // the 2 pulses and the upper 5 bits being coded in base 5
     temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
     pulse_position[3] = temp % 5;
     pulse_position[7] = temp / 5;
     if (pulse_position[7] & 1)
         pulse_position[3] = 4 - pulse_position[3];
     pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
     pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
 
     fixed_sparse->n = 8;
     for (i = 0; i < 4; i++) {
         const int pos1   = (pulse_position[i]     << 2) + i;
         const int pos2   = (pulse_position[i + 4] << 2) + i;
         const float sign = fixed_index[i] ? -1.0 : 1.0;
         fixed_sparse->x[i    ] = pos1;
         fixed_sparse->x[i + 4] = pos2;
         fixed_sparse->y[i    ] = sign;
         fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
     }
 }
 
 /**
  * Decode the algebraic codebook index to pulse positions and signs,
  * then construct the algebraic codebook vector.
  *
  *                              nb of pulses | bits encoding pulses
  * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
  *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
  *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
  *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
  *
  * @param fixed_sparse pointer to the algebraic codebook vector
  * @param pulses       algebraic codebook indexes
  * @param mode         mode of the current frame
  * @param subframe     current subframe number
  */
 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
                                 const enum Mode mode, const int subframe)
 {

     av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2);

 
     if (mode == MODE_12k2) {
         ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
     } else if (mode == MODE_10k2) {
         decode_8_pulses_31bits(pulses, fixed_sparse);
     } else {
         int *pulse_position = fixed_sparse->x;
         int i, pulse_subset;
         const int fixed_index = pulses[0];
 
         if (mode <= MODE_5k15) {
             pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
             pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
             pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
             fixed_sparse->n = 2;
         } else if (mode == MODE_5k9) {
             pulse_subset      = ((fixed_index & 1) << 1) + 1;
             pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
             pulse_subset      = (fixed_index  >> 4) & 3;
             pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
             fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
         } else if (mode == MODE_6k7) {
             pulse_position[0] = (fixed_index        & 7) * 5;
             pulse_subset      = (fixed_index  >> 2) & 2;
             pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
             pulse_subset      = (fixed_index  >> 6) & 2;
             pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
             fixed_sparse->n = 3;
         } else { // mode <= MODE_7k95
             pulse_position[0] = gray_decode[ fixed_index        & 7];
             pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
             pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
             pulse_subset      = (fixed_index >> 9) & 1;
             pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
             fixed_sparse->n = 4;
         }
         for (i = 0; i < fixed_sparse->n; i++)
             fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
     }
 }
 
 /**
  * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
  *
  * @param p the context
  * @param subframe unpacked amr subframe
  * @param mode mode of the current frame

  * @param fixed_sparse sparse representation of the fixed vector

  */
 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
                              AMRFixed *fixed_sparse)
 {
     // The spec suggests the current pitch gain is always used, but in other

     // modes the pitch and codebook gains are jointly quantized (sec 5.8.2)

     // so the codebook gain cannot depend on the quantized pitch gain.
     if (mode == MODE_12k2)
         p->beta = FFMIN(p->pitch_gain[4], 1.0);
 
     fixed_sparse->pitch_lag  = p->pitch_lag_int;
     fixed_sparse->pitch_fac  = p->beta;
 
     // Save pitch sharpening factor for the next subframe
     // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
     // the fact that the gains for two subframes are jointly quantized.
     if (mode != MODE_4k75 || subframe & 1)
         p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
 }
 /// @}
 
 

 /// @name AMR gain decoding functions

 /// @{
 
 /**
  * fixed gain smoothing
  * Note that where the spec specifies the "spectrum in the q domain"
  * in section 6.1.4, in fact frequencies should be used.
  *
  * @param p the context
  * @param lsf LSFs for the current subframe, in the range [0,1]
  * @param lsf_avg averaged LSFs
  * @param mode mode of the current frame
  *
  * @return fixed gain smoothed
  */
 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
                                const float *lsf_avg, const enum Mode mode)
 {
     float diff = 0.0;
     int i;
 
     for (i = 0; i < LP_FILTER_ORDER; i++)
         diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
 
     // If diff is large for ten subframes, disable smoothing for a 40-subframe
     // hangover period.
     p->diff_count++;
     if (diff <= 0.65)
         p->diff_count = 0;
 
     if (p->diff_count > 10) {
         p->hang_count = 0;
         p->diff_count--; // don't let diff_count overflow
     }
 
     if (p->hang_count < 40) {
         p->hang_count++;
     } else if (mode < MODE_7k4 || mode == MODE_10k2) {
         const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
         const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
                                        p->fixed_gain[2] + p->fixed_gain[3] +
                                        p->fixed_gain[4]) * 0.2;
         return smoothing_factor * p->fixed_gain[4] +
                (1.0 - smoothing_factor) * fixed_gain_mean;
     }
     return p->fixed_gain[4];
 }
 
 /**
  * Decode pitch gain and fixed gain factor (part of section 6.1.3).
  *
  * @param p the context
  * @param amr_subframe unpacked amr subframe
  * @param mode mode of the current frame
  * @param subframe current subframe number
  * @param fixed_gain_factor decoded gain correction factor
  */
 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
                          const enum Mode mode, const int subframe,
                          float *fixed_gain_factor)
 {
     if (mode == MODE_12k2 || mode == MODE_7k95) {
         p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
             * (1.0 / 16384.0);
         *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
             * (1.0 /  2048.0);
     } else {
         const uint16_t *gains;
 
         if (mode >= MODE_6k7) {
             gains = gains_high[amr_subframe->p_gain];
         } else if (mode >= MODE_5k15) {
             gains = gains_low [amr_subframe->p_gain];
         } else {
             // gain index is only coded in subframes 0,2 for MODE_4k75
             gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
         }
 
         p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
         *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
     }
 }
 
 /// @}
 
 

 /// @name AMR preprocessing functions

 /// @{
 
 /**
  * Circularly convolve a sparse fixed vector with a phase dispersion impulse
  * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
  *
  * @param out vector with filter applied
  * @param in source vector
  * @param filter phase filter coefficients
  *
  *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
  */
 static void apply_ir_filter(float *out, const AMRFixed *in,
                             const float *filter)
 {

     float filter1[AMR_SUBFRAME_SIZE],     ///< filters at pitch lag*1 and *2

           filter2[AMR_SUBFRAME_SIZE];
     int   lag = in->pitch_lag;
     float fac = in->pitch_fac;
     int i;
 
     if (lag < AMR_SUBFRAME_SIZE) {
         ff_celp_circ_addf(filter1, filter, filter, lag, fac,
                           AMR_SUBFRAME_SIZE);
 
         if (lag < AMR_SUBFRAME_SIZE >> 1)
             ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
                               AMR_SUBFRAME_SIZE);
     }
 
     memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
     for (i = 0; i < in->n; i++) {
         int   x = in->x[i];
         float y = in->y[i];
         const float *filterp;
 
         if (x >= AMR_SUBFRAME_SIZE - lag) {
             filterp = filter;
         } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
             filterp = filter1;
         } else
             filterp = filter2;
 
         ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
     }
 }
 
 /**
  * Reduce fixed vector sparseness by smoothing with one of three IR filters.
  * Also know as "adaptive phase dispersion".
  *
  * This implements 3GPP TS 26.090 section 6.1(5).
  *
  * @param p the context
  * @param fixed_sparse algebraic codebook vector
  * @param fixed_vector unfiltered fixed vector
  * @param fixed_gain smoothed gain
  * @param out space for modified vector if necessary
  */
 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
                                     const float *fixed_vector,
                                     float fixed_gain, float *out)
 {
     int ir_filter_nr;
 
     if (p->pitch_gain[4] < 0.6) {
         ir_filter_nr = 0;      // strong filtering
     } else if (p->pitch_gain[4] < 0.9) {
         ir_filter_nr = 1;      // medium filtering
     } else
         ir_filter_nr = 2;      // no filtering
 
     // detect 'onset'
     if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
         p->ir_filter_onset = 2;
     } else if (p->ir_filter_onset)
         p->ir_filter_onset--;
 
     if (!p->ir_filter_onset) {
         int i, count = 0;
 
         for (i = 0; i < 5; i++)
             if (p->pitch_gain[i] < 0.6)
                 count++;
         if (count > 2)
             ir_filter_nr = 0;
 
         if (ir_filter_nr > p->prev_ir_filter_nr + 1)
             ir_filter_nr--;
     } else if (ir_filter_nr < 2)
         ir_filter_nr++;
 
     // Disable filtering for very low level of fixed_gain.
     // Note this step is not specified in the technical description but is in
     // the reference source in the function Ph_disp.
     if (fixed_gain < 5.0)
         ir_filter_nr = 2;
 
     if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
          && ir_filter_nr < 2) {
         apply_ir_filter(out, fixed_sparse,
                         (p->cur_frame_mode == MODE_7k95 ?
                              ir_filters_lookup_MODE_7k95 :
                              ir_filters_lookup)[ir_filter_nr]);
         fixed_vector = out;
     }
 
     // update ir filter strength history
     p->prev_ir_filter_nr       = ir_filter_nr;
     p->prev_sparse_fixed_gain  = fixed_gain;
 
     return fixed_vector;
 }
 
 /// @}
 
 

 /// @name AMR synthesis functions

 /// @{
 
 /**
  * Conduct 10th order linear predictive coding synthesis.
  *
  * @param p             pointer to the AMRContext
  * @param lpc           pointer to the LPC coefficients
  * @param fixed_gain    fixed codebook gain for synthesis
  * @param fixed_vector  algebraic codebook vector
  * @param samples       pointer to the output speech samples
  * @param overflow      16-bit overflow flag
  */
 static int synthesis(AMRContext *p, float *lpc,
                      float fixed_gain, const float *fixed_vector,
                      float *samples, uint8_t overflow)
 {

     int i;

     float excitation[AMR_SUBFRAME_SIZE];
 
     // if an overflow has been detected, the pitch vector is scaled down by a
     // factor of 4
     if (overflow)
         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
             p->pitch_vector[i] *= 0.25;
 

     p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,

                             p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
 
     // emphasize pitch vector contribution
     if (p->pitch_gain[4] > 0.5 && !overflow) {

         float energy = p->celpm_ctx.dot_productf(excitation, excitation,

                                                     AMR_SUBFRAME_SIZE);

         float pitch_factor =
             p->pitch_gain[4] *
             (p->cur_frame_mode == MODE_12k2 ?
                 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
                 0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
 
         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
             excitation[i] += pitch_factor * p->pitch_vector[i];
 
         ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
                                                 AMR_SUBFRAME_SIZE);
     }
 

     p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation,
                                  AMR_SUBFRAME_SIZE,

                                  LP_FILTER_ORDER);
 
     // detect overflow
     for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
         if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {

             return 1;

         }
 

     return 0;

 }
 
 /// @}
 
 

 /// @name AMR update functions

 /// @{
 
 /**
  * Update buffers and history at the end of decoding a subframe.
  *
  * @param p             pointer to the AMRContext
  */
 static void update_state(AMRContext *p)
 {
     memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
 
     memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
             (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
 
     memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
     memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
 
     memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
             LP_FILTER_ORDER * sizeof(float));
 }
 
 /// @}
 
 

 /// @name AMR Postprocessing functions

 /// @{
 
 /**
  * Get the tilt factor of a formant filter from its transfer function
  *

  * @param p     The Context

  * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
  * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
  */

 static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d)

 {
     float rh0, rh1; // autocorrelation at lag 0 and 1
 
     // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
     float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
     float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
 
     hf[0] = 1.0;
     memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);

     p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf,
                                  AMR_TILT_RESPONSE,

                                  LP_FILTER_ORDER);
 

     rh0 = p->celpm_ctx.dot_productf(hf, hf,     AMR_TILT_RESPONSE);
     rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);

 
     // The spec only specifies this check for 12.2 and 10.2 kbit/s
     // modes. But in the ref source the tilt is always non-negative.
     return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
 }
 
 /**
  * Perform adaptive post-filtering to enhance the quality of the speech.
  * See section 6.2.1.
  *
  * @param p             pointer to the AMRContext
  * @param lpc           interpolated LP coefficients for this subframe
  * @param buf_out       output of the filter
  */
 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
 {
     int i;
     float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
 

     float speech_gain       = p->celpm_ctx.dot_productf(samples, samples,

                                                            AMR_SUBFRAME_SIZE);

 
     float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
     const float *gamma_n, *gamma_d;                       // Formant filter factor table
     float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
 
     if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
         gamma_n = ff_pow_0_7;
         gamma_d = ff_pow_0_75;
     } else {
         gamma_n = ff_pow_0_55;
         gamma_d = ff_pow_0_7;
     }
 
     for (i = 0; i < LP_FILTER_ORDER; i++) {
          lpc_n[i] = lpc[i] * gamma_n[i];
          lpc_d[i] = lpc[i] * gamma_d[i];
     }
 
     memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);

     p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,

                                  AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
     memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
            sizeof(float) * LP_FILTER_ORDER);
 

     p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n,

                                       pole_out + LP_FILTER_ORDER,
                                       AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
 

     ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out,

                          AMR_SUBFRAME_SIZE);
 

     ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,

                              AMR_AGC_ALPHA, &p->postfilter_agc);

 }
 
 /// @}
 

 static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
                               int *got_frame_ptr, AVPacket *avpkt)

 {
 
     AMRContext *p = avctx->priv_data;        // pointer to private data

     AVFrame *frame     = data;

     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;

     float *buf_out;                          // pointer to the output data buffer
     int i, subframe, ret;

     float fixed_gain_factor;
     AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
     float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
     float synth_fixed_gain;                  // the fixed gain that synthesis should use
     const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
 

     /* get output buffer */

     frame->nb_samples = AMR_BLOCK_SIZE;

     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

         return ret;

     buf_out = (float *)frame->data[0];

     p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);

     if (p->cur_frame_mode == NO_DATA) {
         av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n");
         return AVERROR_INVALIDDATA;
     }

     if (p->cur_frame_mode == MODE_DTX) {

         avpriv_report_missing_feature(avctx, "dtx mode");

         av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n");

         return AVERROR_PATCHWELCOME;

     }
 
     if (p->cur_frame_mode == MODE_12k2) {
         lsf2lsp_5(p);
     } else
         lsf2lsp_3(p);
 
     for (i = 0; i < 4; i++)
         ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
 
     for (subframe = 0; subframe < 4; subframe++) {
         const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
 
         decode_pitch_vector(p, amr_subframe, subframe);
 
         decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
                             p->cur_frame_mode, subframe);
 
         // The fixed gain (section 6.1.3) depends on the fixed vector
         // (section 6.1.2), but the fixed vector calculation uses
         // pitch sharpening based on the on the pitch gain (section 6.1.3).
         // So the correct order is: pitch gain, pitch sharpening, fixed gain.
         decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
                      &fixed_gain_factor);
 
         pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
 

         if (fixed_sparse.pitch_lag == 0) {
             av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n");
             return AVERROR_INVALIDDATA;
         }

         ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
                             AMR_SUBFRAME_SIZE);
 
         p->fixed_gain[4] =
             ff_amr_set_fixed_gain(fixed_gain_factor,

                        p->celpm_ctx.dot_productf(p->fixed_vector,

                                                                p->fixed_vector,
                                                                AMR_SUBFRAME_SIZE) /

                                   AMR_SUBFRAME_SIZE,

                        p->prediction_error,
                        energy_mean[p->cur_frame_mode], energy_pred_fac);
 
         // The excitation feedback is calculated without any processing such
         // as fixed gain smoothing. This isn't mentioned in the specification.
         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
             p->excitation[i] *= p->pitch_gain[4];
         ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
                             AMR_SUBFRAME_SIZE);
 
         // In the ref decoder, excitation is stored with no fractional bits.
         // This step prevents buzz in silent periods. The ref encoder can
         // emit long sequences with pitch factor greater than one. This
         // creates unwanted feedback if the excitation vector is nonzero.
         // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
             p->excitation[i] = truncf(p->excitation[i]);
 
         // Smooth fixed gain.
         // The specification is ambiguous, but in the reference source, the
         // smoothed value is NOT fed back into later fixed gain smoothing.
         synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
                                              p->lsf_avg, p->cur_frame_mode);
 
         synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
                                              synth_fixed_gain, spare_vector);
 
         if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
                       synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
             // overflow detected -> rerun synthesis scaling pitch vector down
             // by a factor of 4, skipping pitch vector contribution emphasis
             // and adaptive gain control
             synthesis(p, p->lpc[subframe], synth_fixed_gain,
                       synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
 
         postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
 
         // update buffers and history
         ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
         update_state(p);
     }
 

     p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out,
                                              buf_out, highpass_zeros,

                                              highpass_poles,
                                              highpass_gain * AMR_SAMPLE_SCALE,

                                              p->high_pass_mem, AMR_BLOCK_SIZE);
 
     /* Update averaged lsf vector (used for fixed gain smoothing).
      *
      * Note that lsf_avg should not incorporate the current frame's LSFs
      * for fixed_gain_smooth.
      * The specification has an incorrect formula: the reference decoder uses
      * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */

     p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],

                             0.84, 0.16, LP_FILTER_ORDER);
 

     *got_frame_ptr = 1;

 
     /* return the amount of bytes consumed if everything was OK */
     return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
 }
 
 

 AVCodec ff_amrnb_decoder = {

     .name           = "amrnb",

     .long_name      = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"),

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_AMR_NB,

     .priv_data_size = sizeof(AMRContext),
     .init           = amrnb_decode_init,
     .decode         = amrnb_decode_frame,

     .capabilities   = AV_CODEC_CAP_DR1,

     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT,
                                                      AV_SAMPLE_FMT_NONE },

 };