/* * Real Audio 1.0 (14.4K) encoder * Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it> * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * Real Audio 1.0 (14.4K) encoder * @author Francesco Lavra <francescolavra@interfree.it> */ #include <float.h> #include "avcodec.h" #include "audio_frame_queue.h" #include "celp_filters.h" #include "internal.h" #include "mathops.h" #include "put_bits.h" #include "ra144.h" static av_cold int ra144_encode_close(AVCodecContext *avctx) { RA144Context *ractx = avctx->priv_data; ff_lpc_end(&ractx->lpc_ctx); ff_af_queue_close(&ractx->afq); return 0; } static av_cold int ra144_encode_init(AVCodecContext * avctx) { RA144Context *ractx; int ret; if (avctx->channels != 1) { av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels); return -1; } avctx->frame_size = NBLOCKS * BLOCKSIZE; avctx->initial_padding = avctx->frame_size; avctx->bit_rate = 8000; ractx = avctx->priv_data; ractx->lpc_coef[0] = ractx->lpc_tables[0]; ractx->lpc_coef[1] = ractx->lpc_tables[1]; ractx->avctx = avctx; ff_audiodsp_init(&ractx->adsp); ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER, FF_LPC_TYPE_LEVINSON); if (ret < 0) goto error; ff_af_queue_init(avctx, &ractx->afq); return 0; error: ra144_encode_close(avctx); return ret; } /** * Quantize a value by searching a sorted table for the element with the * nearest value * * @param value value to quantize * @param table array containing the quantization table * @param size size of the quantization table * @return index of the quantization table corresponding to the element with the * nearest value */ static int quantize(int value, const int16_t *table, unsigned int size) { unsigned int low = 0, high = size - 1; while (1) { int index = (low + high) >> 1; int error = table[index] - value; if (index == low) return table[high] + error > value ? low : high; if (error > 0) { high = index; } else { low = index; } } } /** * Orthogonalize a vector to another vector * * @param v vector to orthogonalize * @param u vector against which orthogonalization is performed */ static void orthogonalize(float *v, const float *u) { int i; float num = 0, den = 0; for (i = 0; i < BLOCKSIZE; i++) { num += v[i] * u[i]; den += u[i] * u[i]; } num /= den; for (i = 0; i < BLOCKSIZE; i++) v[i] -= num * u[i]; } /** * Calculate match score and gain of an LPC-filtered vector with respect to * input data, possibly orthogonalizing it to up to two other vectors. * * @param work array used to calculate the filtered vector * @param coefs coefficients of the LPC filter * @param vect original vector * @param ortho1 first vector against which orthogonalization is performed * @param ortho2 second vector against which orthogonalization is performed * @param data input data * @param score pointer to variable where match score is returned * @param gain pointer to variable where gain is returned */ static void get_match_score(float *work, const float *coefs, float *vect, const float *ortho1, const float *ortho2, const float *data, float *score, float *gain) { float c, g; int i; ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER); if (ortho1) orthogonalize(work, ortho1); if (ortho2) orthogonalize(work, ortho2); c = g = 0; for (i = 0; i < BLOCKSIZE; i++) { g += work[i] * work[i]; c += data[i] * work[i]; } if (c <= 0) { *score = 0; return; } *gain = c / g; *score = *gain * c; } /** * Create a vector from the adaptive codebook at a given lag value * * @param vect array where vector is stored * @param cb adaptive codebook * @param lag lag value */ static void create_adapt_vect(float *vect, const int16_t *cb, int lag) { int i; cb += BUFFERSIZE - lag; for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++) vect[i] = cb[i]; if (lag < BLOCKSIZE) for (i = 0; i < BLOCKSIZE - lag; i++) vect[lag + i] = cb[i]; } /** * Search the adaptive codebook for the best entry and gain and remove its * contribution from input data * * @param adapt_cb array from which the adaptive codebook is extracted * @param work array used to calculate LPC-filtered vectors * @param coefs coefficients of the LPC filter * @param data input data * @return index of the best entry of the adaptive codebook */ static int adaptive_cb_search(const int16_t *adapt_cb, float *work, const float *coefs, float *data) { int i, av_uninit(best_vect); float score, gain, best_score, av_uninit(best_gain); float exc[BLOCKSIZE]; gain = best_score = 0; for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) { create_adapt_vect(exc, adapt_cb, i); get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain); if (score > best_score) { best_score = score; best_vect = i; best_gain = gain; } } if (!best_score) return 0; /** * Re-calculate the filtered vector from the vector with maximum match score * and remove its contribution from input data. */ create_adapt_vect(exc, adapt_cb, best_vect); ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER); for (i = 0; i < BLOCKSIZE; i++) data[i] -= best_gain * work[i]; return best_vect - BLOCKSIZE / 2 + 1; } /** * Find the best vector of a fixed codebook by applying an LPC filter to * codebook entries, possibly orthogonalizing them to up to two other vectors * and matching the results with input data. * * @param work array used to calculate the filtered vectors * @param coefs coefficients of the LPC filter * @param cb fixed codebook * @param ortho1 first vector against which orthogonalization is performed * @param ortho2 second vector against which orthogonalization is performed * @param data input data * @param idx pointer to variable where the index of the best codebook entry is * returned * @param gain pointer to variable where the gain of the best codebook entry is * returned */ static void find_best_vect(float *work, const float *coefs, const int8_t cb[][BLOCKSIZE], const float *ortho1, const float *ortho2, float *data, int *idx, float *gain) { int i, j; float g, score, best_score; float vect[BLOCKSIZE]; *idx = *gain = best_score = 0; for (i = 0; i < FIXED_CB_SIZE; i++) { for (j = 0; j < BLOCKSIZE; j++) vect[j] = cb[i][j]; get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g); if (score > best_score) { best_score = score; *idx = i; *gain = g; } } } /** * Search the two fixed codebooks for the best entry and gain * * @param work array used to calculate LPC-filtered vectors * @param coefs coefficients of the LPC filter * @param data input data * @param cba_idx index of the best entry of the adaptive codebook * @param cb1_idx pointer to variable where the index of the best entry of the * first fixed codebook is returned * @param cb2_idx pointer to variable where the index of the best entry of the * second fixed codebook is returned */ static void fixed_cb_search(float *work, const float *coefs, float *data, int cba_idx, int *cb1_idx, int *cb2_idx) { int i, ortho_cb1; float gain; float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE]; float vect[BLOCKSIZE]; /** * The filtered vector from the adaptive codebook can be retrieved from * work, because this function is called just after adaptive_cb_search(). */ if (cba_idx) memcpy(cba_vect, work, sizeof(cba_vect)); find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL, data, cb1_idx, &gain); /** * Re-calculate the filtered vector from the vector with maximum match score * and remove its contribution from input data. */ if (gain) { for (i = 0; i < BLOCKSIZE; i++) vect[i] = ff_cb1_vects[*cb1_idx][i]; ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER); if (cba_idx) orthogonalize(work, cba_vect); for (i = 0; i < BLOCKSIZE; i++) data[i] -= gain * work[i]; memcpy(cb1_vect, work, sizeof(cb1_vect)); ortho_cb1 = 1; } else ortho_cb1 = 0; find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL, ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain); } /** * Encode a subblock of the current frame * * @param ractx encoder context * @param sblock_data input data of the subblock * @param lpc_coefs coefficients of the LPC filter * @param rms RMS of the reflection coefficients * @param pb pointer to PutBitContext of the current frame */ static void ra144_encode_subblock(RA144Context *ractx, const int16_t *sblock_data, const int16_t *lpc_coefs, unsigned int rms, PutBitContext *pb) { float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE]; float coefs[LPC_ORDER]; float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE]; int cba_idx, cb1_idx, cb2_idx, gain; int i, n; unsigned m[3]; float g[3]; float error, best_error; for (i = 0; i < LPC_ORDER; i++) { work[i] = ractx->curr_sblock[BLOCKSIZE + i]; coefs[i] = lpc_coefs[i] * (1/4096.0); } /** * Calculate the zero-input response of the LPC filter and subtract it from * input data. */ ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE, LPC_ORDER); for (i = 0; i < BLOCKSIZE; i++) { zero[i] = work[LPC_ORDER + i]; data[i] = sblock_data[i] - zero[i]; } /** * Codebook search is performed without taking into account the contribution * of the previous subblock, since it has been just subtracted from input * data. */ memset(work, 0, LPC_ORDER * sizeof(*work)); cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs, data); if (cba_idx) { /** * The filtered vector from the adaptive codebook can be retrieved from * work, see implementation of adaptive_cb_search(). */ memcpy(cba, work + LPC_ORDER, sizeof(cba)); ff_copy_and_dup(ractx->buffer_a, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1); m[0] = (ff_irms(&ractx->adsp, ractx->buffer_a) * rms) >> 12; } fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx); for (i = 0; i < BLOCKSIZE; i++) { cb1[i] = ff_cb1_vects[cb1_idx][i]; cb2[i] = ff_cb2_vects[cb2_idx][i]; } ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE, LPC_ORDER); memcpy(cb1, work + LPC_ORDER, sizeof(cb1)); m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8; ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE, LPC_ORDER); memcpy(cb2, work + LPC_ORDER, sizeof(cb2)); m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8; best_error = FLT_MAX; gain = 0; for (n = 0; n < 256; n++) { g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) * (1/4096.0); g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) * (1/4096.0); error = 0; if (cba_idx) { g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) * (1/4096.0); for (i = 0; i < BLOCKSIZE; i++) { data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] + g[2] * cb2[i]; error += (data[i] - sblock_data[i]) * (data[i] - sblock_data[i]); } } else { for (i = 0; i < BLOCKSIZE; i++) { data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i]; error += (data[i] - sblock_data[i]) * (data[i] - sblock_data[i]); } } if (error < best_error) { best_error = error; gain = n; } } put_bits(pb, 7, cba_idx); put_bits(pb, 8, gain); put_bits(pb, 7, cb1_idx); put_bits(pb, 7, cb2_idx); ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms, gain); } static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4}; static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; RA144Context *ractx = avctx->priv_data; PutBitContext pb; int32_t lpc_data[NBLOCKS * BLOCKSIZE]; int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER]; int shift[LPC_ORDER]; int16_t block_coefs[NBLOCKS][LPC_ORDER]; int lpc_refl[LPC_ORDER]; /**< reflection coefficients of the frame */ unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */ const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL; int energy = 0; int i, idx, ret; if (ractx->last_frame) return 0; if ((ret = ff_alloc_packet2(avctx, avpkt, FRAME_SIZE, 0)) < 0) return ret; /** * Since the LPC coefficients are calculated on a frame centered over the * fourth subframe, to encode a given frame, data from the next frame is * needed. In each call to this function, the previous frame (whose data are * saved in the encoder context) is encoded, and data from the current frame * are saved in the encoder context to be used in the next function call. */ for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) { lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i]; energy += (lpc_data[i] * lpc_data[i]) >> 4; } if (frame) { int j; for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) { lpc_data[i] = samples[j] >> 2; energy += (lpc_data[i] * lpc_data[i]) >> 4; } } if (i < NBLOCKS * BLOCKSIZE) memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data)); energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab, 32)]; ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER, LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON, 0, ORDER_METHOD_EST, 0, 12, 0); for (i = 0; i < LPC_ORDER; i++) block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] << (12 - shift[LPC_ORDER - 1])); /** * TODO: apply perceptual weighting of the input speech through bandwidth * expansion of the LPC filter. */ if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { /** * The filter is unstable: use the coefficients of the previous frame. */ ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]); if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) { /* the filter is still unstable. set reflection coeffs to zero. */ memset(lpc_refl, 0, sizeof(lpc_refl)); } } init_put_bits(&pb, avpkt->data, avpkt->size); for (i = 0; i < LPC_ORDER; i++) { idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]); put_bits(&pb, bit_sizes[i], idx); lpc_refl[i] = ff_lpc_refl_cb[i][idx]; } ractx->lpc_refl_rms[0] = ff_rms(lpc_refl); ff_eval_coefs(ractx->lpc_coef[0], lpc_refl); refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); refl_rms[1] = ff_interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy, ff_t_sqrt(energy * ractx->old_energy) >> 12); refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy); refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy); ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]); put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32)); for (i = 0; i < NBLOCKS; i++) ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE, block_coefs[i], refl_rms[i], &pb); flush_put_bits(&pb); ractx->old_energy = energy; ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); /* copy input samples to current block for processing in next call */ i = 0; if (frame) { for (; i < frame->nb_samples; i++) ractx->curr_block[i] = samples[i] >> 2; if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0) return ret; } else ractx->last_frame = 1; memset(&ractx->curr_block[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block)); /* Get the next frame pts/duration */ ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts, &avpkt->duration); avpkt->size = FRAME_SIZE; *got_packet_ptr = 1; return 0; } AVCodec ff_ra_144_encoder = { .name = "real_144", .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), .type = AVMEDIA_TYPE_AUDIO, .id = AV_CODEC_ID_RA_144, .priv_data_size = sizeof(RA144Context), .init = ra144_encode_init, .encode2 = ra144_encode_frame, .close = ra144_encode_close, .capabilities = AV_CODEC_CAP_DELAY | AV_CODEC_CAP_SMALL_LAST_FRAME, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, .supported_samplerates = (const int[]){ 8000, 0 }, .channel_layouts = (const uint64_t[]) { AV_CH_LAYOUT_MONO, 0 }, };