2 * RealAudio 2.0 (28.8K)
3 * Copyright (c) 2003 the ffmpeg project
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/float_dsp.h"
24 #define BITSTREAM_READER_LE
28 #include "celp_filters.h"
30 #define MAX_BACKWARD_FILTER_ORDER 36
31 #define MAX_BACKWARD_FILTER_LEN 40
32 #define MAX_BACKWARD_FILTER_NONREC 35
34 #define RA288_BLOCK_SIZE 5
35 #define RA288_BLOCKS_PER_FRAME 32
40 AVFloatDSPContext fdsp;
41 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
42 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
44 /** speech data history (spec: SB).
45 * Its first 70 coefficients are updated only at backward filtering.
49 /// speech part of the gain autocorrelation (spec: REXP)
52 /** log-gain history (spec: SBLG).
53 * Its first 28 coefficients are updated only at backward filtering.
57 /// recursive part of the gain autocorrelation (spec: REXPLG)
61 static av_cold int ra288_decode_init(AVCodecContext *avctx)
63 RA288Context *ractx = avctx->priv_data;
64 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
65 avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
67 avcodec_get_frame_defaults(&ractx->frame);
68 avctx->coded_frame = &ractx->frame;
73 static void convolve(float *tgt, const float *src, int len, int n)
76 tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
80 static void decode(RA288Context *ractx, float gain, int cb_coef)
85 float *block = ractx->sp_hist + 70 + 36; // current block
86 float *gain_block = ractx->gain_hist + 28;
88 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
90 /* block 46 of G.728 spec */
92 for (i=0; i < 10; i++)
93 sum -= gain_block[9-i] * ractx->gain_lpc[i];
95 /* block 47 of G.728 spec */
96 sum = av_clipf(sum, 0, 60);
98 /* block 48 of G.728 spec */
99 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
100 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
102 for (i=0; i < 5; i++)
103 buffer[i] = codetable[cb_coef][i] * sumsum;
105 sum = ff_scalarproduct_float_c(buffer, buffer, 5);
107 sum = FFMAX(sum, 5. / (1<<24));
109 /* shift and store */
110 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
112 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
114 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
118 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
120 * @param order filter order
121 * @param n input length
122 * @param non_rec number of non-recursive samples
123 * @param out filter output
124 * @param hist pointer to the input history of the filter
125 * @param out pointer to the non-recursive part of the output
126 * @param out2 pointer to the recursive part of the output
127 * @param window pointer to the windowing function table
129 static void do_hybrid_window(RA288Context *ractx,
130 int order, int n, int non_rec, float *out,
131 float *hist, float *out2, const float *window)
134 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
135 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
136 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
137 MAX_BACKWARD_FILTER_LEN +
138 MAX_BACKWARD_FILTER_NONREC, 16)]);
140 ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
142 convolve(buffer1, work + order , n , order);
143 convolve(buffer2, work + order + n, non_rec, order);
145 for (i=0; i <= order; i++) {
146 out2[i] = out2[i] * 0.5625 + buffer1[i];
147 out [i] = out2[i] + buffer2[i];
150 /* Multiply by the white noise correcting factor (WNCF). */
155 * Backward synthesis filter, find the LPC coefficients from past speech data.
157 static void backward_filter(RA288Context *ractx,
158 float *hist, float *rec, const float *window,
159 float *lpc, const float *tab,
160 int order, int n, int non_rec, int move_size)
162 float temp[MAX_BACKWARD_FILTER_ORDER+1];
164 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
166 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
167 ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
169 memmove(hist, hist + n, move_size*sizeof(*hist));
172 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
173 int *got_frame_ptr, AVPacket *avpkt)
175 const uint8_t *buf = avpkt->data;
176 int buf_size = avpkt->size;
179 RA288Context *ractx = avctx->priv_data;
182 if (buf_size < avctx->block_align) {
183 av_log(avctx, AV_LOG_ERROR,
184 "Error! Input buffer is too small [%d<%d]\n",
185 buf_size, avctx->block_align);
186 return AVERROR_INVALIDDATA;
189 /* get output buffer */
190 ractx->frame.nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
191 if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) {
192 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
195 out = (float *)ractx->frame.data[0];
197 init_get_bits(&gb, buf, avctx->block_align * 8);
199 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
200 float gain = amptable[get_bits(&gb, 3)];
201 int cb_coef = get_bits(&gb, 6 + (i&1));
203 decode(ractx, gain, cb_coef);
205 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
206 out += RA288_BLOCK_SIZE;
209 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
210 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
212 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
213 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
218 *(AVFrame *)data = ractx->frame;
220 return avctx->block_align;
223 AVCodec ff_ra_288_decoder = {
225 .type = AVMEDIA_TYPE_AUDIO,
226 .id = AV_CODEC_ID_RA_288,
227 .priv_data_size = sizeof(RA288Context),
228 .init = ra288_decode_init,
229 .decode = ra288_decode_frame,
230 .capabilities = CODEC_CAP_DR1,
231 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),