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568 lines
19 KiB
568 lines
19 KiB
/* |
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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 FFmpeg. |
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* |
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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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/** |
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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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|
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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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#if FF_API_OLD_ENCODE_AUDIO |
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av_freep(&avctx->coded_frame); |
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#endif |
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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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#if FF_API_OLD_ENCODE_AUDIO |
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avctx->coded_frame = avcodec_alloc_frame(); |
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if (!avctx->coded_frame) { |
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ret = AVERROR(ENOMEM); |
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goto error; |
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} |
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#endif |
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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], 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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memset(data, 0, sizeof(data)); |
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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_packet2(avctx, avpkt, FRAMESIZE))) |
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return ret; |
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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, |
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32)]; |
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ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER, |
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LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON, |
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0, ORDER_METHOD_EST, 12, 0); |
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for (i = 0; i < LPC_ORDER; i++) |
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block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] << |
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(12 - shift[LPC_ORDER - 1])); |
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|
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/** |
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* TODO: apply perceptual weighting of the input speech through bandwidth |
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* expansion of the LPC filter. |
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*/ |
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|
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if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { |
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/** |
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* The filter is unstable: use the coefficients of the previous frame. |
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*/ |
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ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]); |
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if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { |
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/* the filter is still unstable. set reflection coeffs to zero. */ |
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memset(lpc_refl, 0, sizeof(lpc_refl)); |
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} |
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} |
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init_put_bits(&pb, avpkt->data, avpkt->size); |
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for (i = 0; i < LPC_ORDER; i++) { |
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idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]); |
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put_bits(&pb, bit_sizes[i], idx); |
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lpc_refl[i] = ff_lpc_refl_cb[i][idx]; |
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} |
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ractx->lpc_refl_rms[0] = ff_rms(lpc_refl); |
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ff_eval_coefs(ractx->lpc_coef[0], lpc_refl); |
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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 = 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)"), |
|
};
|
|
|