559 lines
19 KiB
C
559 lines
19 KiB
C
/*
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* Real Audio 1.0 (14.4K) encoder
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* Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Real Audio 1.0 (14.4K) encoder
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* @author Francesco Lavra <francescolavra@interfree.it>
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*/
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#include <float.h>
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#include "avcodec.h"
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#include "audio_frame_queue.h"
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#include "internal.h"
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#include "put_bits.h"
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#include "celp_filters.h"
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#include "ra144.h"
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static av_cold int ra144_encode_close(AVCodecContext *avctx)
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{
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RA144Context *ractx = avctx->priv_data;
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ff_lpc_end(&ractx->lpc_ctx);
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ff_af_queue_close(&ractx->afq);
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return 0;
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}
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static av_cold int ra144_encode_init(AVCodecContext * avctx)
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{
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RA144Context *ractx;
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int ret;
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if (avctx->channels != 1) {
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av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
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avctx->channels);
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return -1;
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}
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avctx->frame_size = NBLOCKS * BLOCKSIZE;
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avctx->delay = avctx->frame_size;
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avctx->bit_rate = 8000;
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ractx = avctx->priv_data;
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ractx->lpc_coef[0] = ractx->lpc_tables[0];
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ractx->lpc_coef[1] = ractx->lpc_tables[1];
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ractx->avctx = avctx;
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ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,
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FF_LPC_TYPE_LEVINSON);
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if (ret < 0)
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goto error;
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ff_af_queue_init(avctx, &ractx->afq);
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return 0;
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error:
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ra144_encode_close(avctx);
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return ret;
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}
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/**
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* Quantize a value by searching a sorted table for the element with the
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* nearest value
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*
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* @param value value to quantize
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* @param table array containing the quantization table
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* @param size size of the quantization table
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* @return index of the quantization table corresponding to the element with the
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* nearest value
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*/
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static int quantize(int value, const int16_t *table, unsigned int size)
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{
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unsigned int low = 0, high = size - 1;
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while (1) {
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int index = (low + high) >> 1;
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int error = table[index] - value;
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if (index == low)
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return table[high] + error > value ? low : high;
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if (error > 0) {
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high = index;
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} else {
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low = index;
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}
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}
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}
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/**
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* Orthogonalize a vector to another vector
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*
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* @param v vector to orthogonalize
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* @param u vector against which orthogonalization is performed
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*/
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static void orthogonalize(float *v, const float *u)
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{
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int i;
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float num = 0, den = 0;
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for (i = 0; i < BLOCKSIZE; i++) {
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num += v[i] * u[i];
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den += u[i] * u[i];
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}
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num /= den;
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for (i = 0; i < BLOCKSIZE; i++)
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v[i] -= num * u[i];
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}
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/**
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* Calculate match score and gain of an LPC-filtered vector with respect to
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* input data, possibly othogonalizing it to up to 2 other vectors
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*
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* @param work array used to calculate the filtered vector
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* @param coefs coefficients of the LPC filter
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* @param vect original vector
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* @param ortho1 first vector against which orthogonalization is performed
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* @param ortho2 second vector against which orthogonalization is performed
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* @param data input data
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* @param score pointer to variable where match score is returned
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* @param gain pointer to variable where gain is returned
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*/
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static void get_match_score(float *work, const float *coefs, float *vect,
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const float *ortho1, const float *ortho2,
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const float *data, float *score, float *gain)
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{
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float c, g;
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int i;
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ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
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if (ortho1)
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orthogonalize(work, ortho1);
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if (ortho2)
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orthogonalize(work, ortho2);
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c = g = 0;
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for (i = 0; i < BLOCKSIZE; i++) {
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g += work[i] * work[i];
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c += data[i] * work[i];
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}
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if (c <= 0) {
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*score = 0;
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return;
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}
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*gain = c / g;
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*score = *gain * c;
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}
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/**
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* Create a vector from the adaptive codebook at a given lag value
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*
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* @param vect array where vector is stored
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* @param cb adaptive codebook
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* @param lag lag value
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*/
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static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
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{
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int i;
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cb += BUFFERSIZE - lag;
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for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
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vect[i] = cb[i];
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if (lag < BLOCKSIZE)
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for (i = 0; i < BLOCKSIZE - lag; i++)
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vect[lag + i] = cb[i];
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}
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/**
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* Search the adaptive codebook for the best entry and gain and remove its
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* contribution from input data
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*
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* @param adapt_cb array from which the adaptive codebook is extracted
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* @param work array used to calculate LPC-filtered vectors
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* @param coefs coefficients of the LPC filter
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* @param data input data
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* @return index of the best entry of the adaptive codebook
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*/
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static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
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const float *coefs, float *data)
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{
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int i, best_vect;
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float score, gain, best_score, best_gain;
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float exc[BLOCKSIZE];
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gain = best_score = 0;
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for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
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create_adapt_vect(exc, adapt_cb, i);
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get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
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if (score > best_score) {
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best_score = score;
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best_vect = i;
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best_gain = gain;
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}
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}
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if (!best_score)
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return 0;
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/**
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* Re-calculate the filtered vector from the vector with maximum match score
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* and remove its contribution from input data.
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*/
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create_adapt_vect(exc, adapt_cb, best_vect);
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ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
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for (i = 0; i < BLOCKSIZE; i++)
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data[i] -= best_gain * work[i];
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return best_vect - BLOCKSIZE / 2 + 1;
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}
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/**
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* Find the best vector of a fixed codebook by applying an LPC filter to
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* codebook entries, possibly othogonalizing them to up to 2 other vectors and
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* matching the results with input data
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*
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* @param work array used to calculate the filtered vectors
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* @param coefs coefficients of the LPC filter
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* @param cb fixed codebook
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* @param ortho1 first vector against which orthogonalization is performed
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* @param ortho2 second vector against which orthogonalization is performed
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* @param data input data
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* @param idx pointer to variable where the index of the best codebook entry is
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* returned
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* @param gain pointer to variable where the gain of the best codebook entry is
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* returned
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*/
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static void find_best_vect(float *work, const float *coefs,
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const int8_t cb[][BLOCKSIZE], const float *ortho1,
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const float *ortho2, float *data, int *idx,
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float *gain)
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{
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int i, j;
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float g, score, best_score;
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float vect[BLOCKSIZE];
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*idx = *gain = best_score = 0;
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for (i = 0; i < FIXED_CB_SIZE; i++) {
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for (j = 0; j < BLOCKSIZE; j++)
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vect[j] = cb[i][j];
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get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
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if (score > best_score) {
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best_score = score;
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*idx = i;
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*gain = g;
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}
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}
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}
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/**
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* Search the two fixed codebooks for the best entry and gain
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*
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* @param work array used to calculate LPC-filtered vectors
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* @param coefs coefficients of the LPC filter
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* @param data input data
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* @param cba_idx index of the best entry of the adaptive codebook
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* @param cb1_idx pointer to variable where the index of the best entry of the
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* first fixed codebook is returned
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* @param cb2_idx pointer to variable where the index of the best entry of the
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* second fixed codebook is returned
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*/
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static void fixed_cb_search(float *work, const float *coefs, float *data,
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int cba_idx, int *cb1_idx, int *cb2_idx)
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{
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int i, ortho_cb1;
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float gain;
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float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
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float vect[BLOCKSIZE];
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/**
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* The filtered vector from the adaptive codebook can be retrieved from
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* work, because this function is called just after adaptive_cb_search().
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*/
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if (cba_idx)
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memcpy(cba_vect, work, sizeof(cba_vect));
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find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
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data, cb1_idx, &gain);
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/**
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* Re-calculate the filtered vector from the vector with maximum match score
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* and remove its contribution from input data.
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*/
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if (gain) {
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for (i = 0; i < BLOCKSIZE; i++)
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vect[i] = ff_cb1_vects[*cb1_idx][i];
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ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
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if (cba_idx)
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orthogonalize(work, cba_vect);
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for (i = 0; i < BLOCKSIZE; i++)
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data[i] -= gain * work[i];
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memcpy(cb1_vect, work, sizeof(cb1_vect));
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ortho_cb1 = 1;
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} else
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ortho_cb1 = 0;
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find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
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ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
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}
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/**
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* Encode a subblock of the current frame
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*
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* @param ractx encoder context
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* @param sblock_data input data of the subblock
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* @param lpc_coefs coefficients of the LPC filter
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* @param rms RMS of the reflection coefficients
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* @param pb pointer to PutBitContext of the current frame
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*/
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static void ra144_encode_subblock(RA144Context *ractx,
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const int16_t *sblock_data,
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const int16_t *lpc_coefs, unsigned int rms,
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PutBitContext *pb)
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{
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float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];
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float coefs[LPC_ORDER];
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float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
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int16_t cba_vect[BLOCKSIZE];
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int cba_idx, cb1_idx, cb2_idx, gain;
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int i, n, m[3];
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float g[3];
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float error, best_error;
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for (i = 0; i < LPC_ORDER; i++) {
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work[i] = ractx->curr_sblock[BLOCKSIZE + i];
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coefs[i] = lpc_coefs[i] * (1/4096.0);
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}
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/**
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* Calculate the zero-input response of the LPC filter and subtract it from
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* input data.
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*/
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ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
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LPC_ORDER);
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for (i = 0; i < BLOCKSIZE; i++) {
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zero[i] = work[LPC_ORDER + i];
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data[i] = sblock_data[i] - zero[i];
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}
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/**
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* Codebook search is performed without taking into account the contribution
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* of the previous subblock, since it has been just subtracted from input
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* data.
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*/
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memset(work, 0, LPC_ORDER * sizeof(*work));
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cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
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data);
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if (cba_idx) {
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/**
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* The filtered vector from the adaptive codebook can be retrieved from
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* work, see implementation of adaptive_cb_search().
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*/
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memcpy(cba, work + LPC_ORDER, sizeof(cba));
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ff_copy_and_dup(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
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m[0] = (ff_irms(cba_vect) * rms) >> 12;
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}
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fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
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for (i = 0; i < BLOCKSIZE; i++) {
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cb1[i] = ff_cb1_vects[cb1_idx][i];
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cb2[i] = ff_cb2_vects[cb2_idx][i];
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}
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ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,
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LPC_ORDER);
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memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
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m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
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ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,
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LPC_ORDER);
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memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
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m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
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best_error = FLT_MAX;
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gain = 0;
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for (n = 0; n < 256; n++) {
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g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
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(1/4096.0);
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g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
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(1/4096.0);
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error = 0;
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if (cba_idx) {
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g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
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(1/4096.0);
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for (i = 0; i < BLOCKSIZE; i++) {
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data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
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g[2] * cb2[i];
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error += (data[i] - sblock_data[i]) *
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(data[i] - sblock_data[i]);
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}
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} else {
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for (i = 0; i < BLOCKSIZE; i++) {
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data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
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error += (data[i] - sblock_data[i]) *
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(data[i] - sblock_data[i]);
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}
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}
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if (error < best_error) {
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best_error = error;
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gain = n;
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}
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}
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put_bits(pb, 7, cba_idx);
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put_bits(pb, 8, gain);
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put_bits(pb, 7, cb1_idx);
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put_bits(pb, 7, cb2_idx);
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ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
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gain);
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}
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static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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const AVFrame *frame, int *got_packet_ptr)
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{
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static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
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static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
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RA144Context *ractx = avctx->priv_data;
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PutBitContext pb;
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int32_t lpc_data[NBLOCKS * BLOCKSIZE];
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int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
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int shift[LPC_ORDER];
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int16_t block_coefs[NBLOCKS][LPC_ORDER];
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int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */
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unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
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const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
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int energy = 0;
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int i, idx, ret;
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if (ractx->last_frame)
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return 0;
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if ((ret = ff_alloc_packet(avpkt, FRAMESIZE))) {
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av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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return ret;
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}
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/**
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* Since the LPC coefficients are calculated on a frame centered over the
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* fourth subframe, to encode a given frame, data from the next frame is
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* needed. In each call to this function, the previous frame (whose data are
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* saved in the encoder context) is encoded, and data from the current frame
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* are saved in the encoder context to be used in the next function call.
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*/
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for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
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lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
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energy += (lpc_data[i] * lpc_data[i]) >> 4;
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}
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if (frame) {
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int j;
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for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
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lpc_data[i] = samples[j] >> 2;
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energy += (lpc_data[i] * lpc_data[i]) >> 4;
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}
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}
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if (i < NBLOCKS * BLOCKSIZE)
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memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
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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, 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]);
|
|
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 = FRAMESIZE;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
|
|
AVCodec ff_ra_144_encoder = {
|
|
.name = "real_144",
|
|
.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 = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
|
|
};
|