Mercurial > libavcodec.hg
view ra288.c @ 7418:83d005a098b5 libavcodec
Rename st{1,2}{a,b} context variables to more meaningful names: {sp,gain}_{rec,hist}
author | vitor |
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date | Sat, 26 Jul 2008 15:10:17 +0000 |
parents | a65363f8ab4b |
children | 175c2c4d6887 |
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/* * RealAudio 2.0 (28.8K) * Copyright (c) 2003 the ffmpeg project * * 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 */ #include "avcodec.h" #define ALT_BITSTREAM_READER_LE #include "bitstream.h" #include "ra288.h" typedef struct { float history[8]; float output[40]; float pr1[36]; float pr2[10]; int phase; float sp_hist[111]; ///< Speech data history (spec: SB) /** Speech part of the gain autocorrelation (spec: REXP) */ float sp_rec[37]; float st1[37]; float gain_hist[38]; ///< Log-gain history (spec: SBLG) /** Recursive part of the gain autocorrelation (spec: REXPLG) */ float gain_rec[11]; float st2[11]; float sb[41]; float lhist[10]; } RA288Context; static inline float scalar_product_float(const float * v1, const float * v2, int size) { float res = 0.; while (size--) res += *v1++ * *v2++; return res; } static void colmult(float *tgt, const float *m1, const float *m2, int n) { while (n--) *tgt++ = *m1++ * *m2++; } /* Decode and produce output */ static void decode(RA288Context *ractx, float gain, int cb_coef) { int x, y; double sumsum; float sum, buffer[5]; memmove(ractx->sb + 5, ractx->sb, 36 * sizeof(*ractx->sb)); for (x=4; x >= 0; x--) ractx->sb[x] = -scalar_product_float(ractx->sb + x + 1, ractx->pr1, 36); /* convert log and do rms */ sum = 32. - scalar_product_float(ractx->pr2, ractx->lhist, 10); sum = av_clipf(sum, 0, 60); sumsum = exp(sum * 0.1151292546497) * gain; /* pow(10.0,sum/20)*f */ for (x=0; x < 5; x++) buffer[x] = codetable[cb_coef][x] * sumsum; sum = scalar_product_float(buffer, buffer, 5) / 5; sum = FFMAX(sum, 1); /* shift and store */ memmove(ractx->lhist, ractx->lhist - 1, 10 * sizeof(*ractx->lhist)); *ractx->lhist = ractx->history[ractx->phase] = 10 * log10(sum) - 32; for (x=1; x < 5; x++) for (y=x-1; y >= 0; y--) buffer[x] -= ractx->pr1[x-y-1] * buffer[y]; /* output */ for (x=0; x < 5; x++) { ractx->output[ractx->phase*5+x] = ractx->sb[4-x] = av_clipf(ractx->sb[4-x] + buffer[x], -4095, 4095); } } /** * Converts autocorrelation coefficients to LPC coefficients using the * Levinson-Durbin algorithm. See blocks 37 and 50 of the G.728 specification. * * @return 0 if success, -1 if fail */ static int eval_lpc_coeffs(const float *in, float *tgt, int n) { int x, y; double f0, f1, f2; if (in[n] == 0) return -1; if ((f0 = *in) <= 0) return -1; in--; // To avoid a -1 subtraction in the inner loop for (x=1; x <= n; x++) { f1 = in[x+1]; for (y=0; y < x - 1; y++) f1 += in[x-y]*tgt[y]; tgt[x-1] = f2 = -f1/f0; for (y=0; y < x >> 1; y++) { float temp = tgt[y] + tgt[x-y-2]*f2; tgt[x-y-2] += tgt[y]*f2; tgt[y] = temp; } if ((f0 += f1*f2) < 0) return -1; } return 0; } static void prodsum(float *tgt, const float *src, int len, int n) { for (; n >= 0; n--) tgt[n] = scalar_product_float(src, src - n, len); } /** * Hybrid window filtering. See blocks 36 and 49 of the G.728 specification. * * @param order the order of the filter * @param n the length of the input * @param non_rec the number of non-recursive samples * @param out the filter output * @param in pointer to the input of the filter * @param hist pointer to the input history of the filter. It is updated by * this function. * @param out pointer to the non-recursive part of the output * @param out2 pointer to the recursive part of the output * @param window pointer to the windowing function table */ static void do_hybrid_window(int order, int n, int non_rec, const float *in, float *out, float *hist, float *out2, const float *window) { unsigned int x; float buffer1[37]; float buffer2[37]; float work[111]; /* update history */ memmove(hist , hist + n, (order + non_rec)*sizeof(*hist)); memcpy (hist + order + non_rec, in , n *sizeof(*hist)); colmult(work, window, hist, order + n + non_rec); prodsum(buffer1, work + order , n , order); prodsum(buffer2, work + order + n, non_rec, order); for (x=0; x <= order; x++) { out2[x] = out2[x] * 0.5625 + buffer1[x]; out [x] = out2[x] + buffer2[x]; } /* Multiply by the white noise correcting factor (WNCF) */ *out *= 257./256.; } /** * Backward synthesis filter. Find the LPC coefficients from past speech data. */ static void backward_filter(RA288Context *ractx) { float buffer1[40], temp1[37]; float buffer2[8], temp2[11]; memcpy(buffer1 , ractx->output + 20, 20*sizeof(*buffer1)); memcpy(buffer1 + 20, ractx->output , 20*sizeof(*buffer1)); do_hybrid_window(36, 40, 35, buffer1, temp1, ractx->sp_hist, ractx->sp_rec, syn_window); if (!eval_lpc_coeffs(temp1, ractx->st1, 36)) colmult(ractx->pr1, ractx->st1, syn_bw_tab, 36); memcpy(buffer2 , ractx->history + 4, 4*sizeof(*buffer2)); memcpy(buffer2 + 4, ractx->history , 4*sizeof(*buffer2)); do_hybrid_window(10, 8, 20, buffer2, temp2, ractx->gain_hist, ractx->gain_rec, gain_window); if (!eval_lpc_coeffs(temp2, ractx->st2, 10)) colmult(ractx->pr2, ractx->st2, gain_bw_tab, 10); } /* Decode a block (celp) */ static int ra288_decode_frame(AVCodecContext * avctx, void *data, int *data_size, const uint8_t * buf, int buf_size) { int16_t *out = data; int x, y; RA288Context *ractx = avctx->priv_data; GetBitContext gb; if (buf_size < avctx->block_align) { av_log(avctx, AV_LOG_ERROR, "Error! Input buffer is too small [%d<%d]\n", buf_size, avctx->block_align); return 0; } init_get_bits(&gb, buf, avctx->block_align * 8); for (x=0; x < 32; x++) { float gain = amptable[get_bits(&gb, 3)]; int cb_coef = get_bits(&gb, 6 + (x&1)); ractx->phase = x & 7; decode(ractx, gain, cb_coef); for (y=0; y < 5; y++) *(out++) = 8 * ractx->output[ractx->phase*5 + y]; if (ractx->phase == 3) backward_filter(ractx); } *data_size = (char *)out - (char *)data; return avctx->block_align; } AVCodec ra_288_decoder = { "real_288", CODEC_TYPE_AUDIO, CODEC_ID_RA_288, sizeof(RA288Context), NULL, NULL, NULL, ra288_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), };