/* * 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" #include "lpc.h" #include "celp_math.h" #include "celp_filters.h" typedef struct { float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A) float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB) /** speech data history (spec: SB). * Its first 70 coefficients are updated only at backward filtering. */ float sp_hist[111]; /// speech part of the gain autocorrelation (spec: REXP) float sp_rec[37]; /** log-gain history (spec: SBLG). * Its first 28 coefficients are updated only at backward filtering. */ float gain_hist[38]; /// recursive part of the gain autocorrelation (spec: REXPLG) float gain_rec[11]; } RA288Context; static av_cold int ra288_decode_init(AVCodecContext *avctx) { avctx->sample_fmt = SAMPLE_FMT_FLT; return 0; } static void apply_window(float *tgt, const float *m1, const float *m2, int n) { while (n--) *tgt++ = *m1++ * *m2++; } static void convolve(float *tgt, const float *src, int len, int n) { for (; n >= 0; n--) tgt[n] = ff_dot_productf(src, src - n, len); } static void decode(RA288Context *ractx, float gain, int cb_coef) { int i; double sumsum; float sum, buffer[5]; float *block = ractx->sp_hist + 70 + 36; // current block float *gain_block = ractx->gain_hist + 28; memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); /* block 46 of G.728 spec */ sum = 32.; for (i=0; i < 10; i++) sum -= gain_block[9-i] * ractx->gain_lpc[i]; /* block 47 of G.728 spec */ sum = av_clipf(sum, 0, 60); /* block 48 of G.728 spec */ /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); for (i=0; i < 5; i++) buffer[i] = codetable[cb_coef][i] * sumsum; sum = ff_dot_productf(buffer, buffer, 5) * ((1<<24)/5.); sum = FFMAX(sum, 1); /* shift and store */ memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); gain_block[9] = 10 * log10(sum) - 32; ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); /* output */ for (i=0; i < 5; i++) block[i] = av_clipf(block[i], -4095./4096., 4095./4096.); } /** * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. * * @param order filter order * @param n input length * @param non_rec number of non-recursive samples * @param out filter output * @param hist pointer to the input history of the filter * @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, float *out, float *hist, float *out2, const float *window) { int i; float buffer1[order + 1]; float buffer2[order + 1]; float work[order + n + non_rec]; apply_window(work, window, hist, order + n + non_rec); convolve(buffer1, work + order , n , order); convolve(buffer2, work + order + n, non_rec, order); for (i=0; i <= order; i++) { out2[i] = out2[i] * 0.5625 + buffer1[i]; out [i] = out2[i] + buffer2[i]; } /* 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(float *hist, float *rec, const float *window, float *lpc, const float *tab, int order, int n, int non_rec, int move_size) { float temp[order+1]; do_hybrid_window(order, n, non_rec, temp, hist, rec, window); if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) apply_window(lpc, lpc, tab, order); memmove(hist, hist + n, move_size*sizeof(*hist)); } static int ra288_decode_frame(AVCodecContext * avctx, void *data, int *data_size, const uint8_t * buf, int buf_size) { float *out = data; int i, j; 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; } if (*data_size < 32*5*4) return -1; init_get_bits(&gb, buf, avctx->block_align * 8); for (i=0; i < 32; i++) { float gain = amptable[get_bits(&gb, 3)]; int cb_coef = get_bits(&gb, 6 + (i&1)); decode(ractx, gain, cb_coef); for (j=0; j < 5; j++) *(out++) = ractx->sp_hist[70 + 36 + j]; if ((i & 7) == 3) { backward_filter(ractx->sp_hist, ractx->sp_rec, syn_window, ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); backward_filter(ractx->gain_hist, ractx->gain_rec, gain_window, ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); } } *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), ra288_decode_init, NULL, NULL, ra288_decode_frame, .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), };