--- /dev/null
+/*
+ * 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 libavcodec/ra144enc.c
+ * Real Audio 1.0 (14.4K) encoder
+ * @author Francesco Lavra <francescolavra@interfree.it>
+ */
+
+#include <values.h>
+
+#include "avcodec.h"
+#include "put_bits.h"
+#include "lpc.h"
+#include "celp_filters.h"
+#include "ra144.h"
+
+
+static av_cold int ra144_encode_init(AVCodecContext * avctx)
+{
+ RA144Context *ractx;
+
+ if (avctx->sample_fmt != SAMPLE_FMT_S16) {
+ av_log(avctx, AV_LOG_ERROR, "invalid sample format\n");
+ return -1;
+ }
+ 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->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;
+ dsputil_init(&ractx->dsp, avctx);
+ return 0;
+}
+
+
+/**
+ * Quantizes 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;
+ }
+ }
+}
+
+
+/**
+ * Orthogonalizes 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];
+}
+
+
+/**
+ * Calculates match score and gain of an LPC-filtered vector with respect to
+ * input data, possibly othogonalizing it to up to 2 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;
+}
+
+
+/**
+ * Creates 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];
+}
+
+
+/**
+ * Searches the adaptive codebook for the best entry and gain and removes 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, best_vect;
+ float score, gain, best_score, 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);
+}
+
+
+/**
+ * Finds the best vector of a fixed codebook by applying an LPC filter to
+ * codebook entries, possibly othogonalizing them to up to 2 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;
+ }
+ }
+}
+
+
+/**
+ * Searches 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);
+}
+
+
+/**
+ * Encodes 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], work[LPC_ORDER + BLOCKSIZE];
+ float coefs[LPC_ORDER];
+ float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
+ int16_t cba_vect[BLOCKSIZE];
+ int cba_idx, cb1_idx, cb2_idx, gain;
+ int i, n, 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.
+ */
+ memset(data, 0, sizeof(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(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
+ m[0] = (ff_irms(cba_vect) * 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, uint8_t *frame,
+ int buf_size, void *data)
+{
+ 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;
+ 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 */
+ int energy = 0;
+ int i, idx;
+
+ if (buf_size < FRAMESIZE) {
+ av_log(avctx, AV_LOG_ERROR, "output buffer too small\n");
+ return 0;
+ }
+ ractx = avctx->priv_data;
+
+ /**
+ * 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;
+ }
+ for (i = 2 * BLOCKSIZE + BLOCKSIZE / 2; i < NBLOCKS * BLOCKSIZE; i++) {
+ lpc_data[i] = *((int16_t *)data + i - 2 * BLOCKSIZE - BLOCKSIZE / 2) >>
+ 2;
+ energy += (lpc_data[i] * lpc_data[i]) >> 4;
+ }
+ energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
+ 32)];
+
+ ff_lpc_calc_coefs(&ractx->dsp, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
+ LPC_ORDER, 16, lpc_coefs, shift, 1, ORDER_METHOD_EST, 12,
+ 0);
+ for (i = 0; i < LPC_ORDER; i++)
+ block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
+ (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]);
+ ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx);
+ }
+ init_put_bits(&pb, frame, buf_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]);
+ for (i = 0; i < NBLOCKS * BLOCKSIZE; i++)
+ ractx->curr_block[i] = *((int16_t *)data + i) >> 2;
+ return FRAMESIZE;
+}
+
+
+AVCodec ra_144_encoder =
+{
+ "real_144",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_RA_144,
+ sizeof(RA144Context),
+ ra144_encode_init,
+ ra144_encode_frame,
+ .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K) encoder"),
+};
projects / platform / upstream / libav.git / commitdiff