Subversion Repositories Kolibri OS

Rev

Go to most recent revision | Blame | Last modification | View Log | RSS feed

  1. /*
  2.  * QCELP decoder
  3.  * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
  4.  *
  5.  * This file is part of FFmpeg.
  6.  *
  7.  * FFmpeg is free software; you can redistribute it and/or
  8.  * modify it under the terms of the GNU Lesser General Public
  9.  * License as published by the Free Software Foundation; either
  10.  * version 2.1 of the License, or (at your option) any later version.
  11.  *
  12.  * FFmpeg is distributed in the hope that it will be useful,
  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15.  * Lesser General Public License for more details.
  16.  *
  17.  * You should have received a copy of the GNU Lesser General Public
  18.  * License along with FFmpeg; if not, write to the Free Software
  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20.  */
  21.  
  22. /**
  23.  * @file
  24.  * QCELP decoder
  25.  * @author Reynaldo H. Verdejo Pinochet
  26.  * @remark FFmpeg merging spearheaded by Kenan Gillet
  27.  * @remark Development mentored by Benjamin Larson
  28.  */
  29.  
  30. #include <stddef.h>
  31.  
  32. #include "libavutil/avassert.h"
  33. #include "libavutil/channel_layout.h"
  34. #include "libavutil/float_dsp.h"
  35. #include "avcodec.h"
  36. #include "internal.h"
  37. #include "get_bits.h"
  38. #include "qcelpdata.h"
  39. #include "celp_filters.h"
  40. #include "acelp_filters.h"
  41. #include "acelp_vectors.h"
  42. #include "lsp.h"
  43.  
  44. typedef enum {
  45.     I_F_Q = -1,    /**< insufficient frame quality */
  46.     SILENCE,
  47.     RATE_OCTAVE,
  48.     RATE_QUARTER,
  49.     RATE_HALF,
  50.     RATE_FULL
  51. } qcelp_packet_rate;
  52.  
  53. typedef struct {
  54.     GetBitContext     gb;
  55.     qcelp_packet_rate bitrate;
  56.     QCELPFrame        frame;    /**< unpacked data frame */
  57.  
  58.     uint8_t  erasure_count;
  59.     uint8_t  octave_count;      /**< count the consecutive RATE_OCTAVE frames */
  60.     float    prev_lspf[10];
  61.     float    predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
  62.     float    pitch_synthesis_filter_mem[303];
  63.     float    pitch_pre_filter_mem[303];
  64.     float    rnd_fir_filter_mem[180];
  65.     float    formant_mem[170];
  66.     float    last_codebook_gain;
  67.     int      prev_g1[2];
  68.     int      prev_bitrate;
  69.     float    pitch_gain[4];
  70.     uint8_t  pitch_lag[4];
  71.     uint16_t first16bits;
  72.     uint8_t  warned_buf_mismatch_bitrate;
  73.  
  74.     /* postfilter */
  75.     float    postfilter_synth_mem[10];
  76.     float    postfilter_agc_mem;
  77.     float    postfilter_tilt_mem;
  78. } QCELPContext;
  79.  
  80. /**
  81.  * Initialize the speech codec according to the specification.
  82.  *
  83.  * TIA/EIA/IS-733 2.4.9
  84.  */
  85. static av_cold int qcelp_decode_init(AVCodecContext *avctx)
  86. {
  87.     QCELPContext *q = avctx->priv_data;
  88.     int i;
  89.  
  90.     avctx->channels       = 1;
  91.     avctx->channel_layout = AV_CH_LAYOUT_MONO;
  92.     avctx->sample_fmt     = AV_SAMPLE_FMT_FLT;
  93.  
  94.     for (i = 0; i < 10; i++)
  95.         q->prev_lspf[i] = (i + 1) / 11.0;
  96.  
  97.     return 0;
  98. }
  99.  
  100. /**
  101.  * Decode the 10 quantized LSP frequencies from the LSPV/LSP
  102.  * transmission codes of any bitrate and check for badly received packets.
  103.  *
  104.  * @param q the context
  105.  * @param lspf line spectral pair frequencies
  106.  *
  107.  * @return 0 on success, -1 if the packet is badly received
  108.  *
  109.  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
  110.  */
  111. static int decode_lspf(QCELPContext *q, float *lspf)
  112. {
  113.     int i;
  114.     float tmp_lspf, smooth, erasure_coeff;
  115.     const float *predictors;
  116.  
  117.     if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
  118.         predictors = q->prev_bitrate != RATE_OCTAVE &&
  119.                      q->prev_bitrate != I_F_Q ? q->prev_lspf
  120.                                               : q->predictor_lspf;
  121.  
  122.         if (q->bitrate == RATE_OCTAVE) {
  123.             q->octave_count++;
  124.  
  125.             for (i = 0; i < 10; i++) {
  126.                 q->predictor_lspf[i] =
  127.                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
  128.                                                          : -QCELP_LSP_SPREAD_FACTOR) +
  129.                                         predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR   +
  130.                                         (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR) / 11);
  131.             }
  132.             smooth = q->octave_count < 10 ? .875 : 0.1;
  133.         } else {
  134.             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
  135.  
  136.             av_assert2(q->bitrate == I_F_Q);
  137.  
  138.             if (q->erasure_count > 1)
  139.                 erasure_coeff *= q->erasure_count < 4 ? 0.9 : 0.7;
  140.  
  141.             for (i = 0; i < 10; i++) {
  142.                 q->predictor_lspf[i] =
  143.                              lspf[i] = (i + 1) * (1 - erasure_coeff) / 11 +
  144.                                        erasure_coeff * predictors[i];
  145.             }
  146.             smooth = 0.125;
  147.         }
  148.  
  149.         // Check the stability of the LSP frequencies.
  150.         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
  151.         for (i = 1; i < 10; i++)
  152.             lspf[i] = FFMAX(lspf[i], lspf[i - 1] + QCELP_LSP_SPREAD_FACTOR);
  153.  
  154.         lspf[9] = FFMIN(lspf[9], 1.0 - QCELP_LSP_SPREAD_FACTOR);
  155.         for (i = 9; i > 0; i--)
  156.             lspf[i - 1] = FFMIN(lspf[i - 1], lspf[i] - QCELP_LSP_SPREAD_FACTOR);
  157.  
  158.         // Low-pass filter the LSP frequencies.
  159.         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0 - smooth, 10);
  160.     } else {
  161.         q->octave_count = 0;
  162.  
  163.         tmp_lspf = 0.0;
  164.         for (i = 0; i < 5; i++) {
  165.             lspf[2 * i + 0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
  166.             lspf[2 * i + 1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
  167.         }
  168.  
  169.         // Check for badly received packets.
  170.         if (q->bitrate == RATE_QUARTER) {
  171.             if (lspf[9] <= .70 || lspf[9] >= .97)
  172.                 return -1;
  173.             for (i = 3; i < 10; i++)
  174.                 if (fabs(lspf[i] - lspf[i - 2]) < .08)
  175.                     return -1;
  176.         } else {
  177.             if (lspf[9] <= .66 || lspf[9] >= .985)
  178.                 return -1;
  179.             for (i = 4; i < 10; i++)
  180.                 if (fabs(lspf[i] - lspf[i - 4]) < .0931)
  181.                     return -1;
  182.         }
  183.     }
  184.     return 0;
  185. }
  186.  
  187. /**
  188.  * Convert codebook transmission codes to GAIN and INDEX.
  189.  *
  190.  * @param q the context
  191.  * @param gain array holding the decoded gain
  192.  *
  193.  * TIA/EIA/IS-733 2.4.6.2
  194.  */
  195. static void decode_gain_and_index(QCELPContext *q, float *gain)
  196. {
  197.     int i, subframes_count, g1[16];
  198.     float slope;
  199.  
  200.     if (q->bitrate >= RATE_QUARTER) {
  201.         switch (q->bitrate) {
  202.         case RATE_FULL: subframes_count = 16; break;
  203.         case RATE_HALF: subframes_count =  4; break;
  204.         default:        subframes_count =  5;
  205.         }
  206.         for (i = 0; i < subframes_count; i++) {
  207.             g1[i] = 4 * q->frame.cbgain[i];
  208.             if (q->bitrate == RATE_FULL && !((i + 1) & 3)) {
  209.                 g1[i] += av_clip((g1[i - 1] + g1[i - 2] + g1[i - 3]) / 3 - 6, 0, 32);
  210.             }
  211.  
  212.             gain[i] = qcelp_g12ga[g1[i]];
  213.  
  214.             if (q->frame.cbsign[i]) {
  215.                 gain[i] = -gain[i];
  216.                 q->frame.cindex[i] = (q->frame.cindex[i] - 89) & 127;
  217.             }
  218.         }
  219.  
  220.         q->prev_g1[0]         = g1[i - 2];
  221.         q->prev_g1[1]         = g1[i - 1];
  222.         q->last_codebook_gain = qcelp_g12ga[g1[i - 1]];
  223.  
  224.         if (q->bitrate == RATE_QUARTER) {
  225.             // Provide smoothing of the unvoiced excitation energy.
  226.             gain[7] =       gain[4];
  227.             gain[6] = 0.4 * gain[3] + 0.6 * gain[4];
  228.             gain[5] =       gain[3];
  229.             gain[4] = 0.8 * gain[2] + 0.2 * gain[3];
  230.             gain[3] = 0.2 * gain[1] + 0.8 * gain[2];
  231.             gain[2] =       gain[1];
  232.             gain[1] = 0.6 * gain[0] + 0.4 * gain[1];
  233.         }
  234.     } else if (q->bitrate != SILENCE) {
  235.         if (q->bitrate == RATE_OCTAVE) {
  236.             g1[0] = 2 * q->frame.cbgain[0] +
  237.                     av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
  238.             subframes_count = 8;
  239.         } else {
  240.             av_assert2(q->bitrate == I_F_Q);
  241.  
  242.             g1[0] = q->prev_g1[1];
  243.             switch (q->erasure_count) {
  244.             case 1 : break;
  245.             case 2 : g1[0] -= 1; break;
  246.             case 3 : g1[0] -= 2; break;
  247.             default: g1[0] -= 6;
  248.             }
  249.             if (g1[0] < 0)
  250.                 g1[0] = 0;
  251.             subframes_count = 4;
  252.         }
  253.         // This interpolation is done to produce smoother background noise.
  254.         slope = 0.5 * (qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
  255.         for (i = 1; i <= subframes_count; i++)
  256.                 gain[i - 1] = q->last_codebook_gain + slope * i;
  257.  
  258.         q->last_codebook_gain = gain[i - 2];
  259.         q->prev_g1[0]         = q->prev_g1[1];
  260.         q->prev_g1[1]         = g1[0];
  261.     }
  262. }
  263.  
  264. /**
  265.  * If the received packet is Rate 1/4 a further sanity check is made of the
  266.  * codebook gain.
  267.  *
  268.  * @param cbgain the unpacked cbgain array
  269.  * @return -1 if the sanity check fails, 0 otherwise
  270.  *
  271.  * TIA/EIA/IS-733 2.4.8.7.3
  272.  */
  273. static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
  274. {
  275.     int i, diff, prev_diff = 0;
  276.  
  277.     for (i = 1; i < 5; i++) {
  278.         diff = cbgain[i] - cbgain[i-1];
  279.         if (FFABS(diff) > 10)
  280.             return -1;
  281.         else if (FFABS(diff - prev_diff) > 12)
  282.             return -1;
  283.         prev_diff = diff;
  284.     }
  285.     return 0;
  286. }
  287.  
  288. /**
  289.  * Compute the scaled codebook vector Cdn From INDEX and GAIN
  290.  * for all rates.
  291.  *
  292.  * The specification lacks some information here.
  293.  *
  294.  * TIA/EIA/IS-733 has an omission on the codebook index determination
  295.  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
  296.  * you have to subtract the decoded index parameter from the given scaled
  297.  * codebook vector index 'n' to get the desired circular codebook index, but
  298.  * it does not mention that you have to clamp 'n' to [0-9] in order to get
  299.  * RI-compliant results.
  300.  *
  301.  * The reason for this mistake seems to be the fact they forgot to mention you
  302.  * have to do these calculations per codebook subframe and adjust given
  303.  * equation values accordingly.
  304.  *
  305.  * @param q the context
  306.  * @param gain array holding the 4 pitch subframe gain values
  307.  * @param cdn_vector array for the generated scaled codebook vector
  308.  */
  309. static void compute_svector(QCELPContext *q, const float *gain,
  310.                             float *cdn_vector)
  311. {
  312.     int i, j, k;
  313.     uint16_t cbseed, cindex;
  314.     float *rnd, tmp_gain, fir_filter_value;
  315.  
  316.     switch (q->bitrate) {
  317.     case RATE_FULL:
  318.         for (i = 0; i < 16; i++) {
  319.             tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
  320.             cindex   = -q->frame.cindex[i];
  321.             for (j = 0; j < 10; j++)
  322.                 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
  323.         }
  324.         break;
  325.     case RATE_HALF:
  326.         for (i = 0; i < 4; i++) {
  327.             tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
  328.             cindex   = -q->frame.cindex[i];
  329.             for (j = 0; j < 40; j++)
  330.                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
  331.         }
  332.         break;
  333.     case RATE_QUARTER:
  334.         cbseed = (0x0003 & q->frame.lspv[4]) << 14 |
  335.                  (0x003F & q->frame.lspv[3]) <<  8 |
  336.                  (0x0060 & q->frame.lspv[2]) <<  1 |
  337.                  (0x0007 & q->frame.lspv[1]) <<  3 |
  338.                  (0x0038 & q->frame.lspv[0]) >>  3;
  339.         rnd    = q->rnd_fir_filter_mem + 20;
  340.         for (i = 0; i < 8; i++) {
  341.             tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
  342.             for (k = 0; k < 20; k++) {
  343.                 cbseed = 521 * cbseed + 259;
  344.                 *rnd   = (int16_t) cbseed;
  345.  
  346.                     // FIR filter
  347.                 fir_filter_value = 0.0;
  348.                 for (j = 0; j < 10; j++)
  349.                     fir_filter_value += qcelp_rnd_fir_coefs[j] *
  350.                                         (rnd[-j] + rnd[-20+j]);
  351.  
  352.                 fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
  353.                 *cdn_vector++     = tmp_gain * fir_filter_value;
  354.                 rnd++;
  355.             }
  356.         }
  357.         memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160,
  358.                20 * sizeof(float));
  359.         break;
  360.     case RATE_OCTAVE:
  361.         cbseed = q->first16bits;
  362.         for (i = 0; i < 8; i++) {
  363.             tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
  364.             for (j = 0; j < 20; j++) {
  365.                 cbseed        = 521 * cbseed + 259;
  366.                 *cdn_vector++ = tmp_gain * (int16_t) cbseed;
  367.             }
  368.         }
  369.         break;
  370.     case I_F_Q:
  371.         cbseed = -44; // random codebook index
  372.         for (i = 0; i < 4; i++) {
  373.             tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
  374.             for (j = 0; j < 40; j++)
  375.                 *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
  376.         }
  377.         break;
  378.     case SILENCE:
  379.         memset(cdn_vector, 0, 160 * sizeof(float));
  380.         break;
  381.     }
  382. }
  383.  
  384. /**
  385.  * Apply generic gain control.
  386.  *
  387.  * @param v_out output vector
  388.  * @param v_in gain-controlled vector
  389.  * @param v_ref vector to control gain of
  390.  *
  391.  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
  392.  */
  393. static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
  394. {
  395.     int i;
  396.  
  397.     for (i = 0; i < 160; i += 40) {
  398.         float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
  399.         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
  400.     }
  401. }
  402.  
  403. /**
  404.  * Apply filter in pitch-subframe steps.
  405.  *
  406.  * @param memory buffer for the previous state of the filter
  407.  *        - must be able to contain 303 elements
  408.  *        - the 143 first elements are from the previous state
  409.  *        - the next 160 are for output
  410.  * @param v_in input filter vector
  411.  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
  412.  * @param lag per-subframe lag array, each element is
  413.  *        - between 16 and 143 if its corresponding pfrac is 0,
  414.  *        - between 16 and 139 otherwise
  415.  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
  416.  *        otherwise
  417.  *
  418.  * @return filter output vector
  419.  */
  420. static const float *do_pitchfilter(float memory[303], const float v_in[160],
  421.                                    const float gain[4], const uint8_t *lag,
  422.                                    const uint8_t pfrac[4])
  423. {
  424.     int i, j;
  425.     float *v_lag, *v_out;
  426.     const float *v_len;
  427.  
  428.     v_out = memory + 143; // Output vector starts at memory[143].
  429.  
  430.     for (i = 0; i < 4; i++) {
  431.         if (gain[i]) {
  432.             v_lag = memory + 143 + 40 * i - lag[i];
  433.             for (v_len = v_in + 40; v_in < v_len; v_in++) {
  434.                 if (pfrac[i]) { // If it is a fractional lag...
  435.                     for (j = 0, *v_out = 0.0; j < 4; j++)
  436.                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j - 4] + v_lag[3 - j]);
  437.                 } else
  438.                     *v_out = *v_lag;
  439.  
  440.                 *v_out = *v_in + gain[i] * *v_out;
  441.  
  442.                 v_lag++;
  443.                 v_out++;
  444.             }
  445.         } else {
  446.             memcpy(v_out, v_in, 40 * sizeof(float));
  447.             v_in  += 40;
  448.             v_out += 40;
  449.         }
  450.     }
  451.  
  452.     memmove(memory, memory + 160, 143 * sizeof(float));
  453.     return memory + 143;
  454. }
  455.  
  456. /**
  457.  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
  458.  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
  459.  *
  460.  * @param q the context
  461.  * @param cdn_vector the scaled codebook vector
  462.  */
  463. static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
  464. {
  465.     int i;
  466.     const float *v_synthesis_filtered, *v_pre_filtered;
  467.  
  468.     if (q->bitrate >= RATE_HALF || q->bitrate == SILENCE ||
  469.         (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
  470.  
  471.         if (q->bitrate >= RATE_HALF) {
  472.             // Compute gain & lag for the whole frame.
  473.             for (i = 0; i < 4; i++) {
  474.                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
  475.  
  476.                 q->pitch_lag[i] = q->frame.plag[i] + 16;
  477.             }
  478.         } else {
  479.             float max_pitch_gain;
  480.  
  481.             if (q->bitrate == I_F_Q) {
  482.                   if (q->erasure_count < 3)
  483.                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
  484.                   else
  485.                       max_pitch_gain = 0.0;
  486.             } else {
  487.                 av_assert2(q->bitrate == SILENCE);
  488.                 max_pitch_gain = 1.0;
  489.             }
  490.             for (i = 0; i < 4; i++)
  491.                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
  492.  
  493.             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
  494.         }
  495.  
  496.         // pitch synthesis filter
  497.         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
  498.                                               cdn_vector, q->pitch_gain,
  499.                                               q->pitch_lag, q->frame.pfrac);
  500.  
  501.         // pitch prefilter update
  502.         for (i = 0; i < 4; i++)
  503.             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
  504.  
  505.         v_pre_filtered       = do_pitchfilter(q->pitch_pre_filter_mem,
  506.                                               v_synthesis_filtered,
  507.                                               q->pitch_gain, q->pitch_lag,
  508.                                               q->frame.pfrac);
  509.  
  510.         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
  511.     } else {
  512.         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17, 143 * sizeof(float));
  513.         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
  514.         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
  515.         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
  516.     }
  517. }
  518.  
  519. /**
  520.  * Reconstruct LPC coefficients from the line spectral pair frequencies
  521.  * and perform bandwidth expansion.
  522.  *
  523.  * @param lspf line spectral pair frequencies
  524.  * @param lpc linear predictive coding coefficients
  525.  *
  526.  * @note: bandwidth_expansion_coeff could be precalculated into a table
  527.  *        but it seems to be slower on x86
  528.  *
  529.  * TIA/EIA/IS-733 2.4.3.3.5
  530.  */
  531. static void lspf2lpc(const float *lspf, float *lpc)
  532. {
  533.     double lsp[10];
  534.     double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
  535.     int i;
  536.  
  537.     for (i = 0; i < 10; i++)
  538.         lsp[i] = cos(M_PI * lspf[i]);
  539.  
  540.     ff_acelp_lspd2lpc(lsp, lpc, 5);
  541.  
  542.     for (i = 0; i < 10; i++) {
  543.         lpc[i]                    *= bandwidth_expansion_coeff;
  544.         bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
  545.     }
  546. }
  547.  
  548. /**
  549.  * Interpolate LSP frequencies and compute LPC coefficients
  550.  * for a given bitrate & pitch subframe.
  551.  *
  552.  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
  553.  *
  554.  * @param q the context
  555.  * @param curr_lspf LSP frequencies vector of the current frame
  556.  * @param lpc float vector for the resulting LPC
  557.  * @param subframe_num frame number in decoded stream
  558.  */
  559. static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
  560.                             float *lpc, const int subframe_num)
  561. {
  562.     float interpolated_lspf[10];
  563.     float weight;
  564.  
  565.     if (q->bitrate >= RATE_QUARTER)
  566.         weight = 0.25 * (subframe_num + 1);
  567.     else if (q->bitrate == RATE_OCTAVE && !subframe_num)
  568.         weight = 0.625;
  569.     else
  570.         weight = 1.0;
  571.  
  572.     if (weight != 1.0) {
  573.         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
  574.                                 weight, 1.0 - weight, 10);
  575.         lspf2lpc(interpolated_lspf, lpc);
  576.     } else if (q->bitrate >= RATE_QUARTER ||
  577.                (q->bitrate == I_F_Q && !subframe_num))
  578.         lspf2lpc(curr_lspf, lpc);
  579.     else if (q->bitrate == SILENCE && !subframe_num)
  580.         lspf2lpc(q->prev_lspf, lpc);
  581. }
  582.  
  583. static qcelp_packet_rate buf_size2bitrate(const int buf_size)
  584. {
  585.     switch (buf_size) {
  586.     case 35: return RATE_FULL;
  587.     case 17: return RATE_HALF;
  588.     case  8: return RATE_QUARTER;
  589.     case  4: return RATE_OCTAVE;
  590.     case  1: return SILENCE;
  591.     }
  592.  
  593.     return I_F_Q;
  594. }
  595.  
  596. /**
  597.  * Determine the bitrate from the frame size and/or the first byte of the frame.
  598.  *
  599.  * @param avctx the AV codec context
  600.  * @param buf_size length of the buffer
  601.  * @param buf the bufffer
  602.  *
  603.  * @return the bitrate on success,
  604.  *         I_F_Q  if the bitrate cannot be satisfactorily determined
  605.  *
  606.  * TIA/EIA/IS-733 2.4.8.7.1
  607.  */
  608. static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx,
  609.                                            const int buf_size,
  610.                                            const uint8_t **buf)
  611. {
  612.     qcelp_packet_rate bitrate;
  613.  
  614.     if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
  615.         if (bitrate > **buf) {
  616.             QCELPContext *q = avctx->priv_data;
  617.             if (!q->warned_buf_mismatch_bitrate) {
  618.             av_log(avctx, AV_LOG_WARNING,
  619.                    "Claimed bitrate and buffer size mismatch.\n");
  620.                 q->warned_buf_mismatch_bitrate = 1;
  621.             }
  622.             bitrate = **buf;
  623.         } else if (bitrate < **buf) {
  624.             av_log(avctx, AV_LOG_ERROR,
  625.                    "Buffer is too small for the claimed bitrate.\n");
  626.             return I_F_Q;
  627.         }
  628.         (*buf)++;
  629.     } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
  630.         av_log(avctx, AV_LOG_WARNING,
  631.                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
  632.     } else
  633.         return I_F_Q;
  634.  
  635.     if (bitrate == SILENCE) {
  636.         // FIXME: Remove this warning when tested with samples.
  637.         avpriv_request_sample(avctx, "Blank frame handling");
  638.     }
  639.     return bitrate;
  640. }
  641.  
  642. static void warn_insufficient_frame_quality(AVCodecContext *avctx,
  643.                                             const char *message)
  644. {
  645.     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n",
  646.            avctx->frame_number, message);
  647. }
  648.  
  649. static void postfilter(QCELPContext *q, float *samples, float *lpc)
  650. {
  651.     static const float pow_0_775[10] = {
  652.         0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
  653.         0.216676, 0.167924, 0.130141, 0.100859, 0.078166
  654.     }, pow_0_625[10] = {
  655.         0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
  656.         0.059605, 0.037253, 0.023283, 0.014552, 0.009095
  657.     };
  658.     float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
  659.     int n;
  660.  
  661.     for (n = 0; n < 10; n++) {
  662.         lpc_s[n] = lpc[n] * pow_0_625[n];
  663.         lpc_p[n] = lpc[n] * pow_0_775[n];
  664.     }
  665.  
  666.     ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
  667.                                       q->formant_mem + 10, 160, 10);
  668.     memcpy(pole_out, q->postfilter_synth_mem, sizeof(float) * 10);
  669.     ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
  670.     memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
  671.  
  672.     ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
  673.  
  674.     ff_adaptive_gain_control(samples, pole_out + 10,
  675.                              avpriv_scalarproduct_float_c(q->formant_mem + 10,
  676.                                                           q->formant_mem + 10,
  677.                                                           160),
  678.                              160, 0.9375, &q->postfilter_agc_mem);
  679. }
  680.  
  681. static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
  682.                               int *got_frame_ptr, AVPacket *avpkt)
  683. {
  684.     const uint8_t *buf = avpkt->data;
  685.     int buf_size       = avpkt->size;
  686.     QCELPContext *q    = avctx->priv_data;
  687.     AVFrame *frame     = data;
  688.     float *outbuffer;
  689.     int   i, ret;
  690.     float quantized_lspf[10], lpc[10];
  691.     float gain[16];
  692.     float *formant_mem;
  693.  
  694.     /* get output buffer */
  695.     frame->nb_samples = 160;
  696.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  697.         return ret;
  698.     outbuffer = (float *)frame->data[0];
  699.  
  700.     if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
  701.         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
  702.         goto erasure;
  703.     }
  704.  
  705.     if (q->bitrate == RATE_OCTAVE &&
  706.         (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
  707.         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
  708.         goto erasure;
  709.     }
  710.  
  711.     if (q->bitrate > SILENCE) {
  712.         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
  713.         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate] +
  714.                                          qcelp_unpacking_bitmaps_lengths[q->bitrate];
  715.         uint8_t *unpacked_data         = (uint8_t *)&q->frame;
  716.  
  717.         init_get_bits(&q->gb, buf, 8 * buf_size);
  718.  
  719.         memset(&q->frame, 0, sizeof(QCELPFrame));
  720.  
  721.         for (; bitmaps < bitmaps_end; bitmaps++)
  722.             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
  723.  
  724.         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
  725.         if (q->frame.reserved) {
  726.             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
  727.             goto erasure;
  728.         }
  729.         if (q->bitrate == RATE_QUARTER &&
  730.             codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
  731.             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
  732.             goto erasure;
  733.         }
  734.  
  735.         if (q->bitrate >= RATE_HALF) {
  736.             for (i = 0; i < 4; i++) {
  737.                 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
  738.                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
  739.                     goto erasure;
  740.                 }
  741.             }
  742.         }
  743.     }
  744.  
  745.     decode_gain_and_index(q, gain);
  746.     compute_svector(q, gain, outbuffer);
  747.  
  748.     if (decode_lspf(q, quantized_lspf) < 0) {
  749.         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
  750.         goto erasure;
  751.     }
  752.  
  753.     apply_pitch_filters(q, outbuffer);
  754.  
  755.     if (q->bitrate == I_F_Q) {
  756. erasure:
  757.         q->bitrate = I_F_Q;
  758.         q->erasure_count++;
  759.         decode_gain_and_index(q, gain);
  760.         compute_svector(q, gain, outbuffer);
  761.         decode_lspf(q, quantized_lspf);
  762.         apply_pitch_filters(q, outbuffer);
  763.     } else
  764.         q->erasure_count = 0;
  765.  
  766.     formant_mem = q->formant_mem + 10;
  767.     for (i = 0; i < 4; i++) {
  768.         interpolate_lpc(q, quantized_lspf, lpc, i);
  769.         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40, 10);
  770.         formant_mem += 40;
  771.     }
  772.  
  773.     // postfilter, as per TIA/EIA/IS-733 2.4.8.6
  774.     postfilter(q, outbuffer, lpc);
  775.  
  776.     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
  777.  
  778.     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
  779.     q->prev_bitrate  = q->bitrate;
  780.  
  781.     *got_frame_ptr = 1;
  782.  
  783.     return buf_size;
  784. }
  785.  
  786. AVCodec ff_qcelp_decoder = {
  787.     .name           = "qcelp",
  788.     .long_name      = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
  789.     .type           = AVMEDIA_TYPE_AUDIO,
  790.     .id             = AV_CODEC_ID_QCELP,
  791.     .init           = qcelp_decode_init,
  792.     .decode         = qcelp_decode_frame,
  793.     .capabilities   = CODEC_CAP_DR1,
  794.     .priv_data_size = sizeof(QCELPContext),
  795. };
  796.