libavcodec/ra144enc.c - manifest_repos/ffmpeg - Git at Google

 /*
  * Real Audio 1.0 (14.4K) encoder
  * Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
  *
  * 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
  * Real Audio 1.0 (14.4K) encoder
  * @author Francesco Lavra <francescolavra@interfree.it>
  */

 #include <float.h>

 #include "avcodec.h"
 #include "audio_frame_queue.h"
 #include "celp_filters.h"
 #include "internal.h"
 #include "mathops.h"
 #include "put_bits.h"
 #include "ra144.h"

 static av_cold int ra144_encode_close(AVCodecContext *avctx)
 {
     RA144Context *ractx = avctx->priv_data;
     ff_lpc_end(&ractx->lpc_ctx);
     ff_af_queue_close(&ractx->afq);
     return 0;
 }


 static av_cold int ra144_encode_init(AVCodecContext * avctx)
 {
     RA144Context *ractx;
     int ret;

     if (avctx->channels != 1) {
         av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
                avctx->channels);
         return -1;
     }
     avctx->frame_size = NBLOCKS * BLOCKSIZE;
     avctx->initial_padding = avctx->frame_size;
     avctx->bit_rate = 8000;
     ractx = avctx->priv_data;
     ractx->lpc_coef[0] = ractx->lpc_tables[0];
     ractx->lpc_coef[1] = ractx->lpc_tables[1];
     ractx->avctx = avctx;
     ff_audiodsp_init(&ractx->adsp);
     ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,
                       FF_LPC_TYPE_LEVINSON);
     if (ret < 0)
         return ret;

     ff_af_queue_init(avctx, &ractx->afq);

     return 0;
 }


 /**
  * Quantize a value by searching a sorted table for the element with the
  * nearest value
  *
  * @param value value to quantize
  * @param table array containing the quantization table
  * @param size size of the quantization table
  * @return index of the quantization table corresponding to the element with the
  *         nearest value
  */
 static int quantize(int value, const int16_t *table, unsigned int size)
 {
     unsigned int low = 0, high = size - 1;

     while (1) {
         int index = (low + high) >> 1;
         int error = table[index] - value;

         if (index == low)
             return table[high] + error > value ? low : high;
         if (error > 0) {
             high = index;
         } else {
             low = index;
         }
     }
 }


 /**
  * Orthogonalize a vector to another vector
  *
  * @param v vector to orthogonalize
  * @param u vector against which orthogonalization is performed
  */
 static void orthogonalize(float *v, const float *u)
 {
     int i;
     float num = 0, den = 0;

     for (i = 0; i < BLOCKSIZE; i++) {
         num += v[i] * u[i];
         den += u[i] * u[i];
     }
     num /= den;
     for (i = 0; i < BLOCKSIZE; i++)
         v[i] -= num * u[i];
 }


 /**
  * Calculate match score and gain of an LPC-filtered vector with respect to
  * input data, possibly orthogonalizing it to up to two other vectors.
  *
  * @param work array used to calculate the filtered vector
  * @param coefs coefficients of the LPC filter
  * @param vect original vector
  * @param ortho1 first vector against which orthogonalization is performed
  * @param ortho2 second vector against which orthogonalization is performed
  * @param data input data
  * @param score pointer to variable where match score is returned
  * @param gain pointer to variable where gain is returned
  */
 static void get_match_score(float *work, const float *coefs, float *vect,
                             const float *ortho1, const float *ortho2,
                             const float *data, float *score, float *gain)
 {
     float c, g;
     int i;

     ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
     if (ortho1)
         orthogonalize(work, ortho1);
     if (ortho2)
         orthogonalize(work, ortho2);
     c = g = 0;
     for (i = 0; i < BLOCKSIZE; i++) {
         g += work[i] * work[i];
         c += data[i] * work[i];
     }
     if (c <= 0) {
         *score = 0;
         return;
     }
     *gain = c / g;
     *score = *gain * c;
 }


 /**
  * Create a vector from the adaptive codebook at a given lag value
  *
  * @param vect array where vector is stored
  * @param cb adaptive codebook
  * @param lag lag value
  */
 static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
 {
     int i;

     cb += BUFFERSIZE - lag;
     for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
         vect[i] = cb[i];
     if (lag < BLOCKSIZE)
         for (i = 0; i < BLOCKSIZE - lag; i++)
             vect[lag + i] = cb[i];
 }


 /**
  * Search the adaptive codebook for the best entry and gain and remove its
  * contribution from input data
  *
  * @param adapt_cb array from which the adaptive codebook is extracted
  * @param work array used to calculate LPC-filtered vectors
  * @param coefs coefficients of the LPC filter
  * @param data input data
  * @return index of the best entry of the adaptive codebook
  */
 static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
                               const float *coefs, float *data)
 {
     int i, av_uninit(best_vect);
     float score, gain, best_score, av_uninit(best_gain);
     float exc[BLOCKSIZE];

     gain = best_score = 0;
     for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
         create_adapt_vect(exc, adapt_cb, i);
         get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
         if (score > best_score) {
             best_score = score;
             best_vect = i;
             best_gain = gain;
         }
     }
     if (!best_score)
         return 0;

     /**
      * Re-calculate the filtered vector from the vector with maximum match score
      * and remove its contribution from input data.
      */
     create_adapt_vect(exc, adapt_cb, best_vect);
     ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
     for (i = 0; i < BLOCKSIZE; i++)
         data[i] -= best_gain * work[i];
     return best_vect - BLOCKSIZE / 2 + 1;
 }


 /**
  * Find the best vector of a fixed codebook by applying an LPC filter to
  * codebook entries, possibly orthogonalizing them to up to two other vectors
  * and matching the results with input data.
  *
  * @param work array used to calculate the filtered vectors
  * @param coefs coefficients of the LPC filter
  * @param cb fixed codebook
  * @param ortho1 first vector against which orthogonalization is performed
  * @param ortho2 second vector against which orthogonalization is performed
  * @param data input data
  * @param idx pointer to variable where the index of the best codebook entry is
  *        returned
  * @param gain pointer to variable where the gain of the best codebook entry is
  *        returned
  */
 static void find_best_vect(float *work, const float *coefs,
                            const int8_t cb[][BLOCKSIZE], const float *ortho1,
                            const float *ortho2, float *data, int *idx,
                            float *gain)
 {
     int i, j;
     float g, score, best_score;
     float vect[BLOCKSIZE];

     *idx = *gain = best_score = 0;
     for (i = 0; i < FIXED_CB_SIZE; i++) {
         for (j = 0; j < BLOCKSIZE; j++)
             vect[j] = cb[i][j];
         get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
         if (score > best_score) {
             best_score = score;
             *idx = i;
             *gain = g;
         }
     }
 }


 /**
  * Search the two fixed codebooks for the best entry and gain
  *
  * @param work array used to calculate LPC-filtered vectors
  * @param coefs coefficients of the LPC filter
  * @param data input data
  * @param cba_idx index of the best entry of the adaptive codebook
  * @param cb1_idx pointer to variable where the index of the best entry of the
  *        first fixed codebook is returned
  * @param cb2_idx pointer to variable where the index of the best entry of the
  *        second fixed codebook is returned
  */
 static void fixed_cb_search(float *work, const float *coefs, float *data,
                             int cba_idx, int *cb1_idx, int *cb2_idx)
 {
     int i, ortho_cb1;
     float gain;
     float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
     float vect[BLOCKSIZE];

     /**
      * The filtered vector from the adaptive codebook can be retrieved from
      * work, because this function is called just after adaptive_cb_search().
      */
     if (cba_idx)
         memcpy(cba_vect, work, sizeof(cba_vect));

     find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
                    data, cb1_idx, &gain);

     /**
      * Re-calculate the filtered vector from the vector with maximum match score
      * and remove its contribution from input data.
      */
     if (gain) {
         for (i = 0; i < BLOCKSIZE; i++)
             vect[i] = ff_cb1_vects[*cb1_idx][i];
         ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
         if (cba_idx)
             orthogonalize(work, cba_vect);
         for (i = 0; i < BLOCKSIZE; i++)
             data[i] -= gain * work[i];
         memcpy(cb1_vect, work, sizeof(cb1_vect));
         ortho_cb1 = 1;
     } else
         ortho_cb1 = 0;

     find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
                    ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
 }


 /**
  * Encode a subblock of the current frame
  *
  * @param ractx encoder context
  * @param sblock_data input data of the subblock
  * @param lpc_coefs coefficients of the LPC filter
  * @param rms RMS of the reflection coefficients
  * @param pb pointer to PutBitContext of the current frame
  */
 static void ra144_encode_subblock(RA144Context *ractx,
                                   const int16_t *sblock_data,
                                   const int16_t *lpc_coefs, unsigned int rms,
                                   PutBitContext *pb)
 {
     float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];
     float coefs[LPC_ORDER];
     float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
     int cba_idx, cb1_idx, cb2_idx, gain;
     int i, n;
     unsigned m[3];
     float g[3];
     float error, best_error;

     for (i = 0; i < LPC_ORDER; i++) {
         work[i] = ractx->curr_sblock[BLOCKSIZE + i];
         coefs[i] = lpc_coefs[i] * (1/4096.0);
     }

     /**
      * Calculate the zero-input response of the LPC filter and subtract it from
      * input data.
      */
     ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
                                  LPC_ORDER);
     for (i = 0; i < BLOCKSIZE; i++) {
         zero[i] = work[LPC_ORDER + i];
         data[i] = sblock_data[i] - zero[i];
     }

     /**
      * Codebook search is performed without taking into account the contribution
      * of the previous subblock, since it has been just subtracted from input
      * data.
      */
     memset(work, 0, LPC_ORDER * sizeof(*work));

     cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
                                  data);
     if (cba_idx) {
         /**
          * The filtered vector from the adaptive codebook can be retrieved from
          * work, see implementation of adaptive_cb_search().
          */
         memcpy(cba, work + LPC_ORDER, sizeof(cba));

         ff_copy_and_dup(ractx->buffer_a, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
         m[0] = (ff_irms(&ractx->adsp, ractx->buffer_a) * rms) >> 12;
     }
     fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
     for (i = 0; i < BLOCKSIZE; i++) {
         cb1[i] = ff_cb1_vects[cb1_idx][i];
         cb2[i] = ff_cb2_vects[cb2_idx][i];
     }
     ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,
                                  LPC_ORDER);
     memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
     m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
     ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,
                                  LPC_ORDER);
     memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
     m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
     best_error = FLT_MAX;
     gain = 0;
     for (n = 0; n < 256; n++) {
         g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
                (1/4096.0);
         g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
                (1/4096.0);
         error = 0;
         if (cba_idx) {
             g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
                    (1/4096.0);
             for (i = 0; i < BLOCKSIZE; i++) {
                 data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
                           g[2] * cb2[i];
                 error += (data[i] - sblock_data[i]) *
                          (data[i] - sblock_data[i]);
             }
         } else {
             for (i = 0; i < BLOCKSIZE; i++) {
                 data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
                 error += (data[i] - sblock_data[i]) *
                          (data[i] - sblock_data[i]);
             }
         }
         if (error < best_error) {
             best_error = error;
             gain = n;
         }
     }
     put_bits(pb, 7, cba_idx);
     put_bits(pb, 8, gain);
     put_bits(pb, 7, cb1_idx);
     put_bits(pb, 7, cb2_idx);
     ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
                           gain);
 }


 static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                               const AVFrame *frame, int *got_packet_ptr)
 {
     static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
     static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
     RA144Context *ractx = avctx->priv_data;
     PutBitContext pb;
     int32_t lpc_data[NBLOCKS * BLOCKSIZE];
     int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
     int shift[LPC_ORDER];
     int16_t block_coefs[NBLOCKS][LPC_ORDER];
     int lpc_refl[LPC_ORDER];    /**< reflection coefficients of the frame */
     unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
     const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
     int energy = 0;
     int i, idx, ret;

     if (ractx->last_frame)
         return 0;

     if ((ret = ff_alloc_packet2(avctx, avpkt, FRAME_SIZE, 0)) < 0)
         return ret;

     /**
      * Since the LPC coefficients are calculated on a frame centered over the
      * fourth subframe, to encode a given frame, data from the next frame is
      * needed. In each call to this function, the previous frame (whose data are
      * saved in the encoder context) is encoded, and data from the current frame
      * are saved in the encoder context to be used in the next function call.
      */
     for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
         lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
         energy += (lpc_data[i] * lpc_data[i]) >> 4;
     }
     if (frame) {
         int j;
         for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
             lpc_data[i] = samples[j] >> 2;
             energy += (lpc_data[i] * lpc_data[i]) >> 4;
         }
     }
     if (i < NBLOCKS * BLOCKSIZE)
         memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
     energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
                                     32)];

     ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
                       LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,
                       0, ORDER_METHOD_EST, 0, 12, 0);
     for (i = 0; i < LPC_ORDER; i++)
         block_coefs[NBLOCKS - 1][i] = -lpc_coefs[LPC_ORDER - 1][i]
                                        * (1 << (12 - shift[LPC_ORDER - 1]));

     /**
      * TODO: apply perceptual weighting of the input speech through bandwidth
      * expansion of the LPC filter.
      */

     if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
         /**
          * The filter is unstable: use the coefficients of the previous frame.
          */
         ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
         if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
             /* the filter is still unstable. set reflection coeffs to zero. */
             memset(lpc_refl, 0, sizeof(lpc_refl));
         }
     }
     init_put_bits(&pb, avpkt->data, avpkt->size);
     for (i = 0; i < LPC_ORDER; i++) {
         idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
         put_bits(&pb, bit_sizes[i], idx);
         lpc_refl[i] = ff_lpc_refl_cb[i][idx];
     }
     ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
     ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
     refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
     refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
                             energy <= ractx->old_energy,
                             ff_t_sqrt(energy * ractx->old_energy) >> 12);
     refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
     refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
     ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
     put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
     for (i = 0; i < NBLOCKS; i++)
         ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
                               block_coefs[i], refl_rms[i], &pb);
     flush_put_bits(&pb);
     ractx->old_energy = energy;
     ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
     FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);

     /* copy input samples to current block for processing in next call */
     i = 0;
     if (frame) {
         for (; i < frame->nb_samples; i++)
             ractx->curr_block[i] = samples[i] >> 2;

         if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)
             return ret;
     } else
         ractx->last_frame = 1;
     memset(&ractx->curr_block[i], 0,
            (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));

     /* Get the next frame pts/duration */
     ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
                        &avpkt->duration);

     avpkt->size = FRAME_SIZE;
     *got_packet_ptr = 1;
     return 0;
 }


 AVCodec ff_ra_144_encoder = {
     .name           = "real_144",
     .long_name      = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_RA_144,
     .priv_data_size = sizeof(RA144Context),
     .init           = ra144_encode_init,
     .encode2        = ra144_encode_frame,
     .close          = ra144_encode_close,
     .capabilities   = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                      AV_SAMPLE_FMT_NONE },
     .supported_samplerates = (const int[]){ 8000, 0 },
     .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 },
 };
	/*
	* Real Audio 1.0 (14.4K) encoder
	* Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
	*
	* 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
	* Real Audio 1.0 (14.4K) encoder
	* @author Francesco Lavra <francescolavra@interfree.it>
	*/

	#include <float.h>

	#include "avcodec.h"
	#include "audio_frame_queue.h"
	#include "celp_filters.h"
	#include "internal.h"
	#include "mathops.h"
	#include "put_bits.h"
	#include "ra144.h"

	static av_cold int ra144_encode_close(AVCodecContext *avctx)
	{
	RA144Context *ractx = avctx->priv_data;
	ff_lpc_end(&ractx->lpc_ctx);
	ff_af_queue_close(&ractx->afq);
	return 0;
	}


	static av_cold int ra144_encode_init(AVCodecContext * avctx)
	{
	RA144Context *ractx;
	int ret;

	if (avctx->channels != 1) {
	av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
	avctx->channels);
	return -1;
	}
	avctx->frame_size = NBLOCKS * BLOCKSIZE;
	avctx->initial_padding = avctx->frame_size;
	avctx->bit_rate = 8000;
	ractx = avctx->priv_data;
	ractx->lpc_coef[0] = ractx->lpc_tables[0];
	ractx->lpc_coef[1] = ractx->lpc_tables[1];
	ractx->avctx = avctx;
	ff_audiodsp_init(&ractx->adsp);
	ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,
	FF_LPC_TYPE_LEVINSON);
	if (ret < 0)
	return ret;

	ff_af_queue_init(avctx, &ractx->afq);

	return 0;
	}


	/**
	* Quantize a value by searching a sorted table for the element with the
	* nearest value
	*
	* @param value value to quantize
	* @param table array containing the quantization table
	* @param size size of the quantization table
	* @return index of the quantization table corresponding to the element with the
	* nearest value
	*/
	static int quantize(int value, const int16_t *table, unsigned int size)
	{
	unsigned int low = 0, high = size - 1;

	while (1) {
	int index = (low + high) >> 1;
	int error = table[index] - value;

	if (index == low)
	return table[high] + error > value ? low : high;
	if (error > 0) {
	high = index;
	} else {
	low = index;
	}
	}
	}


	/**
	* Orthogonalize a vector to another vector
	*
	* @param v vector to orthogonalize
	* @param u vector against which orthogonalization is performed
	*/
	static void orthogonalize(float v, const float u)
	{
	int i;
	float num = 0, den = 0;

	for (i = 0; i < BLOCKSIZE; i++) {
	num += v[i] * u[i];
	den += u[i] * u[i];
	}
	num /= den;
	for (i = 0; i < BLOCKSIZE; i++)
	v[i] -= num * u[i];
	}


	/**
	* Calculate match score and gain of an LPC-filtered vector with respect to
	* input data, possibly orthogonalizing it to up to two other vectors.
	*
	* @param work array used to calculate the filtered vector
	* @param coefs coefficients of the LPC filter
	* @param vect original vector
	* @param ortho1 first vector against which orthogonalization is performed
	* @param ortho2 second vector against which orthogonalization is performed
	* @param data input data
	* @param score pointer to variable where match score is returned
	* @param gain pointer to variable where gain is returned
	*/
	static void get_match_score(float work, const float coefs, float *vect,
	const float ortho1, const float ortho2,
	const float data, float score, float *gain)
	{
	float c, g;
	int i;

	ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
	if (ortho1)
	orthogonalize(work, ortho1);
	if (ortho2)
	orthogonalize(work, ortho2);
	c = g = 0;
	for (i = 0; i < BLOCKSIZE; i++) {
	g += work[i] * work[i];
	c += data[i] * work[i];
	}
	if (c <= 0) {
	*score = 0;
	return;
	}
	*gain = c / g;
	score = gain * c;
	}


	/**
	* Create a vector from the adaptive codebook at a given lag value
	*
	* @param vect array where vector is stored
	* @param cb adaptive codebook
	* @param lag lag value
	*/
	static void create_adapt_vect(float vect, const int16_t cb, int lag)
	{
	int i;

	cb += BUFFERSIZE - lag;
	for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
	vect[i] = cb[i];
	if (lag < BLOCKSIZE)
	for (i = 0; i < BLOCKSIZE - lag; i++)
	vect[lag + i] = cb[i];
	}


	/**
	* Search the adaptive codebook for the best entry and gain and remove its
	* contribution from input data
	*
	* @param adapt_cb array from which the adaptive codebook is extracted
	* @param work array used to calculate LPC-filtered vectors
	* @param coefs coefficients of the LPC filter
	* @param data input data
	* @return index of the best entry of the adaptive codebook
	*/
	static int adaptive_cb_search(const int16_t adapt_cb, float work,
	const float coefs, float data)
	{
	int i, av_uninit(best_vect);
	float score, gain, best_score, av_uninit(best_gain);
	float exc[BLOCKSIZE];

	gain = best_score = 0;
	for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
	create_adapt_vect(exc, adapt_cb, i);
	get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
	if (score > best_score) {
	best_score = score;
	best_vect = i;
	best_gain = gain;
	}
	}
	if (!best_score)
	return 0;

	/**
	* Re-calculate the filtered vector from the vector with maximum match score
	* and remove its contribution from input data.
	*/
	create_adapt_vect(exc, adapt_cb, best_vect);
	ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
	for (i = 0; i < BLOCKSIZE; i++)
	data[i] -= best_gain * work[i];
	return best_vect - BLOCKSIZE / 2 + 1;
	}


	/**
	* Find the best vector of a fixed codebook by applying an LPC filter to
	* codebook entries, possibly orthogonalizing them to up to two other vectors
	* and matching the results with input data.
	*
	* @param work array used to calculate the filtered vectors
	* @param coefs coefficients of the LPC filter
	* @param cb fixed codebook
	* @param ortho1 first vector against which orthogonalization is performed
	* @param ortho2 second vector against which orthogonalization is performed
	* @param data input data
	* @param idx pointer to variable where the index of the best codebook entry is
	* returned
	* @param gain pointer to variable where the gain of the best codebook entry is
	* returned
	*/
	static void find_best_vect(float work, const float coefs,
	const int8_t cb[][BLOCKSIZE], const float *ortho1,
	const float ortho2, float data, int *idx,
	float *gain)
	{
	int i, j;
	float g, score, best_score;
	float vect[BLOCKSIZE];

	idx = gain = best_score = 0;
	for (i = 0; i < FIXED_CB_SIZE; i++) {
	for (j = 0; j < BLOCKSIZE; j++)
	vect[j] = cb[i][j];
	get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
	if (score > best_score) {
	best_score = score;
	*idx = i;
	*gain = g;
	}
	}
	}


	/**
	* Search the two fixed codebooks for the best entry and gain
	*
	* @param work array used to calculate LPC-filtered vectors
	* @param coefs coefficients of the LPC filter
	* @param data input data
	* @param cba_idx index of the best entry of the adaptive codebook
	* @param cb1_idx pointer to variable where the index of the best entry of the
	* first fixed codebook is returned
	* @param cb2_idx pointer to variable where the index of the best entry of the
	* second fixed codebook is returned
	*/
	static void fixed_cb_search(float work, const float coefs, float *data,
	int cba_idx, int cb1_idx, int cb2_idx)
	{
	int i, ortho_cb1;
	float gain;
	float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
	float vect[BLOCKSIZE];

	/**
	* The filtered vector from the adaptive codebook can be retrieved from
	* work, because this function is called just after adaptive_cb_search().
	*/
	if (cba_idx)
	memcpy(cba_vect, work, sizeof(cba_vect));

	find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
	data, cb1_idx, &gain);

	/**
	* Re-calculate the filtered vector from the vector with maximum match score
	* and remove its contribution from input data.
	*/
	if (gain) {
	for (i = 0; i < BLOCKSIZE; i++)
	vect[i] = ff_cb1_vects[*cb1_idx][i];
	ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
	if (cba_idx)
	orthogonalize(work, cba_vect);
	for (i = 0; i < BLOCKSIZE; i++)
	data[i] -= gain * work[i];
	memcpy(cb1_vect, work, sizeof(cb1_vect));
	ortho_cb1 = 1;
	} else
	ortho_cb1 = 0;

	find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
	ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
	}


	/**
	* Encode a subblock of the current frame
	*
	* @param ractx encoder context
	* @param sblock_data input data of the subblock
	* @param lpc_coefs coefficients of the LPC filter
	* @param rms RMS of the reflection coefficients
	* @param pb pointer to PutBitContext of the current frame
	*/
	static void ra144_encode_subblock(RA144Context *ractx,
	const int16_t *sblock_data,
	const int16_t *lpc_coefs, unsigned int rms,
	PutBitContext *pb)
	{
	float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];
	float coefs[LPC_ORDER];
	float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
	int cba_idx, cb1_idx, cb2_idx, gain;
	int i, n;
	unsigned m[3];
	float g[3];
	float error, best_error;

	for (i = 0; i < LPC_ORDER; i++) {
	work[i] = ractx->curr_sblock[BLOCKSIZE + i];
	coefs[i] = lpc_coefs[i] * (1/4096.0);
	}

	/**
	* Calculate the zero-input response of the LPC filter and subtract it from
	* input data.
	*/
	ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
	LPC_ORDER);
	for (i = 0; i < BLOCKSIZE; i++) {
	zero[i] = work[LPC_ORDER + i];
	data[i] = sblock_data[i] - zero[i];
	}

	/**
	* Codebook search is performed without taking into account the contribution
	* of the previous subblock, since it has been just subtracted from input
	* data.
	*/
	memset(work, 0, LPC_ORDER * sizeof(*work));

	cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
	data);
	if (cba_idx) {
	/**
	* The filtered vector from the adaptive codebook can be retrieved from
	* work, see implementation of adaptive_cb_search().
	*/
	memcpy(cba, work + LPC_ORDER, sizeof(cba));

	ff_copy_and_dup(ractx->buffer_a, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
	m[0] = (ff_irms(&ractx->adsp, ractx->buffer_a) * rms) >> 12;
	}
	fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
	for (i = 0; i < BLOCKSIZE; i++) {
	cb1[i] = ff_cb1_vects[cb1_idx][i];
	cb2[i] = ff_cb2_vects[cb2_idx][i];
	}
	ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,
	LPC_ORDER);
	memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
	m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
	ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,
	LPC_ORDER);
	memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
	m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
	best_error = FLT_MAX;
	gain = 0;
	for (n = 0; n < 256; n++) {
	g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
	(1/4096.0);
	g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
	(1/4096.0);
	error = 0;
	if (cba_idx) {
	g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
	(1/4096.0);
	for (i = 0; i < BLOCKSIZE; i++) {
	data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
	g[2] * cb2[i];
	error += (data[i] - sblock_data[i]) *
	(data[i] - sblock_data[i]);
	}
	} else {
	for (i = 0; i < BLOCKSIZE; i++) {
	data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
	error += (data[i] - sblock_data[i]) *
	(data[i] - sblock_data[i]);
	}
	}
	if (error < best_error) {
	best_error = error;
	gain = n;
	}
	}
	put_bits(pb, 7, cba_idx);
	put_bits(pb, 8, gain);
	put_bits(pb, 7, cb1_idx);
	put_bits(pb, 7, cb2_idx);
	ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
	gain);
	}


	static int ra144_encode_frame(AVCodecContext avctx, AVPacket avpkt,
	const AVFrame frame, int got_packet_ptr)
	{
	static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
	static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
	RA144Context *ractx = avctx->priv_data;
	PutBitContext pb;
	int32_t lpc_data[NBLOCKS * BLOCKSIZE];
	int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
	int shift[LPC_ORDER];
	int16_t block_coefs[NBLOCKS][LPC_ORDER];
	int lpc_refl[LPC_ORDER]; /*< reflection coefficients of the frame /
	unsigned int refl_rms[NBLOCKS]; /*< RMS of the reflection coefficients /
	const int16_t samples = frame ? (const int16_t )frame->data[0] : NULL;
	int energy = 0;
	int i, idx, ret;

	if (ractx->last_frame)
	return 0;

	if ((ret = ff_alloc_packet2(avctx, avpkt, FRAME_SIZE, 0)) < 0)
	return ret;

	/**
	* Since the LPC coefficients are calculated on a frame centered over the
	* fourth subframe, to encode a given frame, data from the next frame is
	* needed. In each call to this function, the previous frame (whose data are
	* saved in the encoder context) is encoded, and data from the current frame
	* are saved in the encoder context to be used in the next function call.
	*/
	for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
	lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
	energy += (lpc_data[i] * lpc_data[i]) >> 4;
	}
	if (frame) {
	int j;
	for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
	lpc_data[i] = samples[j] >> 2;
	energy += (lpc_data[i] * lpc_data[i]) >> 4;
	}
	}
	if (i < NBLOCKS * BLOCKSIZE)
	memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
	energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
	32)];

	ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
	LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,
	0, ORDER_METHOD_EST, 0, 12, 0);
	for (i = 0; i < LPC_ORDER; i++)
	block_coefs[NBLOCKS - 1][i] = -lpc_coefs[LPC_ORDER - 1][i]
	* (1 << (12 - shift[LPC_ORDER - 1]));

	/**
	* TODO: apply perceptual weighting of the input speech through bandwidth
	* expansion of the LPC filter.
	*/

	if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
	/**
	* The filter is unstable: use the coefficients of the previous frame.
	*/
	ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
	if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
	/* the filter is still unstable. set reflection coeffs to zero. */
	memset(lpc_refl, 0, sizeof(lpc_refl));
	}
	}
	init_put_bits(&pb, avpkt->data, avpkt->size);
	for (i = 0; i < LPC_ORDER; i++) {
	idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
	put_bits(&pb, bit_sizes[i], idx);
	lpc_refl[i] = ff_lpc_refl_cb[i][idx];
	}
	ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
	ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
	refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
	refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
	energy <= ractx->old_energy,
	ff_t_sqrt(energy * ractx->old_energy) >> 12);
	refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
	refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
	ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
	put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
	for (i = 0; i < NBLOCKS; i++)
	ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
	block_coefs[i], refl_rms[i], &pb);
	flush_put_bits(&pb);
	ractx->old_energy = energy;
	ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
	FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);

	/* copy input samples to current block for processing in next call */
	i = 0;
	if (frame) {
	for (; i < frame->nb_samples; i++)
	ractx->curr_block[i] = samples[i] >> 2;

	if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)
	return ret;
	} else
	ractx->last_frame = 1;
	memset(&ractx->curr_block[i], 0,
	(NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));

	/* Get the next frame pts/duration */
	ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
	&avpkt->duration);

	avpkt->size = FRAME_SIZE;
	*got_packet_ptr = 1;
	return 0;
	}


	AVCodec ff_ra_144_encoder = {
	.name = "real_144",
	.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
	.type = AVMEDIA_TYPE_AUDIO,
	.id = AV_CODEC_ID_RA_144,
	.priv_data_size = sizeof(RA144Context),
	.init = ra144_encode_init,
	.encode2 = ra144_encode_frame,
	.close = ra144_encode_close,
	.capabilities = AV_CODEC_CAP_DELAY \| AV_CODEC_CAP_SMALL_LAST_FRAME,
	.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
	AV_SAMPLE_FMT_NONE },
	.supported_samplerates = (const int[]){ 8000, 0 },
	.channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 },
	};