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235 lines
6.7 KiB
C
235 lines
6.7 KiB
C
/*
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* RealAudio 2.0 (28.8K)
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* Copyright (c) 2003 the ffmpeg project
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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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#include "avcodec.h"
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#define ALT_BITSTREAM_READER_LE
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#include "bitstream.h"
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#include "ra288.h"
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#include "lpc.h"
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typedef struct {
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float sp_lpc[36]; ///< LPC coefficients for speech data (spec: A)
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float gain_lpc[10]; ///< LPC coefficients for gain (spec: GB)
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/** speech data history (spec: SB).
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* Its first 70 coefficients are updated only at backward filtering.
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*/
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float sp_hist[111];
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/// speech part of the gain autocorrelation (spec: REXP)
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float sp_rec[37];
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/** log-gain history (spec: SBLG).
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* Its first 28 coefficients are updated only at backward filtering.
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*/
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float gain_hist[38];
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/// recursive part of the gain autocorrelation (spec: REXPLG)
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float gain_rec[11];
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} RA288Context;
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static av_cold int ra288_decode_init(AVCodecContext *avctx)
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{
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avctx->sample_fmt = SAMPLE_FMT_FLT;
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return 0;
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}
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static inline float scalar_product_float(const float * v1, const float * v2,
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int size)
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{
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float res = 0.;
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while (size--)
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res += *v1++ * *v2++;
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return res;
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}
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static void apply_window(float *tgt, const float *m1, const float *m2, int n)
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{
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while (n--)
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*tgt++ = *m1++ * *m2++;
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}
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static void decode(RA288Context *ractx, float gain, int cb_coef)
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{
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int i, j;
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double sumsum;
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float sum, buffer[5];
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float *block = ractx->sp_hist + 70 + 36; // current block
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float *gain_block = ractx->gain_hist + 28;
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memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
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/* block 46 of G.728 spec */
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sum = 32.;
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for (i=0; i < 10; i++)
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sum -= gain_block[9-i] * ractx->gain_lpc[i];
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/* block 47 of G.728 spec */
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sum = av_clipf(sum, 0, 60);
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/* block 48 of G.728 spec */
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/* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
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sumsum = exp(sum * 0.1151292546497) * gain / 2048.;
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for (i=0; i < 5; i++)
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buffer[i] = codetable[cb_coef][i] * sumsum;
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sum = scalar_product_float(buffer, buffer, 5) / 5;
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sum = FFMAX(sum, 1);
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/* shift and store */
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memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
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gain_block[9] = 10 * log10(sum) - 32;
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for (i=0; i < 5; i++) {
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block[i] = buffer[i];
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for (j=0; j < 36; j++)
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block[i] -= block[i-1-j]*ractx->sp_lpc[j];
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}
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/* output */
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for (i=0; i < 5; i++)
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block[i] = av_clipf(block[i], -4095, 4095);
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}
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static void convolve(float *tgt, const float *src, int len, int n)
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{
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for (; n >= 0; n--)
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tgt[n] = scalar_product_float(src, src - n, len);
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}
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/**
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* Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
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*
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* @param order filter order
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* @param n input length
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* @param non_rec number of non-recursive samples
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* @param out filter output
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* @param hist pointer to the input history of the filter
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* @param out pointer to the non-recursive part of the output
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* @param out2 pointer to the recursive part of the output
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* @param window pointer to the windowing function table
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*/
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static void do_hybrid_window(int order, int n, int non_rec, float *out,
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float *hist, float *out2, const float *window)
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{
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int i;
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float buffer1[order + 1];
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float buffer2[order + 1];
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float work[order + n + non_rec];
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apply_window(work, window, hist, order + n + non_rec);
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convolve(buffer1, work + order , n , order);
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convolve(buffer2, work + order + n, non_rec, order);
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for (i=0; i <= order; i++) {
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out2[i] = out2[i] * 0.5625 + buffer1[i];
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out [i] = out2[i] + buffer2[i];
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}
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/* Multiply by the white noise correcting factor (WNCF). */
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*out *= 257./256.;
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}
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/**
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* Backward synthesis filter, find the LPC coefficients from past speech data.
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*/
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static void backward_filter(RA288Context *ractx)
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{
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float temp1[37]; // RTMP in the spec
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float temp2[11]; // GPTPMP in the spec
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do_hybrid_window(36, 40, 35, temp1, ractx->sp_hist,
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ractx->sp_rec, syn_window);
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if (!compute_lpc_coefs(temp1, 36, ractx->sp_lpc, 0, 1, 1))
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apply_window(ractx->sp_lpc, ractx->sp_lpc, syn_bw_tab, 36);
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do_hybrid_window(10, 8, 20, temp2, ractx->gain_hist,
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ractx->gain_rec, gain_window);
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if (!compute_lpc_coefs(temp2, 10, ractx->gain_lpc, 0, 1, 1))
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apply_window(ractx->gain_lpc, ractx->gain_lpc, gain_bw_tab, 10);
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memmove(ractx->gain_hist, ractx->gain_hist + 8,
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28*sizeof(*ractx->gain_hist));
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memmove(ractx->sp_hist , ractx->sp_hist + 40,
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70*sizeof(*ractx->sp_hist ));
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}
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static int ra288_decode_frame(AVCodecContext * avctx, void *data,
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int *data_size, const uint8_t * buf,
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int buf_size)
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{
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float *out = data;
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int i, j;
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RA288Context *ractx = avctx->priv_data;
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GetBitContext gb;
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if (buf_size < avctx->block_align) {
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av_log(avctx, AV_LOG_ERROR,
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"Error! Input buffer is too small [%d<%d]\n",
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buf_size, avctx->block_align);
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return 0;
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}
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if (*data_size < 32*5*4)
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return -1;
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init_get_bits(&gb, buf, avctx->block_align * 8);
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for (i=0; i < 32; i++) {
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float gain = amptable[get_bits(&gb, 3)];
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int cb_coef = get_bits(&gb, 6 + (i&1));
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decode(ractx, gain, cb_coef);
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for (j=0; j < 5; j++)
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*(out++) = (1/4096.) * ractx->sp_hist[70 + 36 + j];
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if ((i & 7) == 3)
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backward_filter(ractx);
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}
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*data_size = (char *)out - (char *)data;
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return avctx->block_align;
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}
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AVCodec ra_288_decoder =
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{
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"real_288",
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CODEC_TYPE_AUDIO,
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CODEC_ID_RA_288,
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sizeof(RA288Context),
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ra288_decode_init,
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NULL,
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NULL,
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ra288_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
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};
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