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_math.h"
29 #include "celp_filters.h"
31 #define MAX_BACKWARD_FILTER_ORDER 36
32 #define MAX_BACKWARD_FILTER_LEN 40
33 #define MAX_BACKWARD_FILTER_NONREC 35
35 #define RA288_BLOCK_SIZE 5
36 #define RA288_BLOCKS_PER_FRAME 32
41 AVFloatDSPContext fdsp;
42 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
43 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
45 /** speech data history (spec: SB).
46 * Its first 70 coefficients are updated only at backward filtering.
50 /// speech part of the gain autocorrelation (spec: REXP)
53 /** log-gain history (spec: SBLG).
54 * Its first 28 coefficients are updated only at backward filtering.
58 /// recursive part of the gain autocorrelation (spec: REXPLG)
62 static av_cold int ra288_decode_init(AVCodecContext *avctx)
64 RA288Context *ractx = avctx->priv_data;
65 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
66 avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
68 avcodec_get_frame_defaults(&ractx->frame);
69 avctx->coded_frame = &ractx->frame;
74 static void convolve(float *tgt, const float *src, int len, int n)
77 tgt[n] = ff_dot_productf(src, src - n, len);
81 static void decode(RA288Context *ractx, float gain, int cb_coef)
86 float *block = ractx->sp_hist + 70 + 36; // current block
87 float *gain_block = ractx->gain_hist + 28;
89 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
91 /* block 46 of G.728 spec */
93 for (i=0; i < 10; i++)
94 sum -= gain_block[9-i] * ractx->gain_lpc[i];
96 /* block 47 of G.728 spec */
97 sum = av_clipf(sum, 0, 60);
99 /* block 48 of G.728 spec */
100 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
101 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
103 for (i=0; i < 5; i++)
104 buffer[i] = codetable[cb_coef][i] * sumsum;
106 sum = ff_dot_productf(buffer, buffer, 5);
108 sum = FFMAX(sum, 5. / (1<<24));
110 /* shift and store */
111 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
113 gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32);
115 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
119 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
121 * @param order filter order
122 * @param n input length
123 * @param non_rec number of non-recursive samples
124 * @param out filter output
125 * @param hist pointer to the input history of the filter
126 * @param out pointer to the non-recursive part of the output
127 * @param out2 pointer to the recursive part of the output
128 * @param window pointer to the windowing function table
130 static void do_hybrid_window(RA288Context *ractx,
131 int order, int n, int non_rec, float *out,
132 float *hist, float *out2, const float *window)
135 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
136 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
137 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
138 MAX_BACKWARD_FILTER_LEN +
139 MAX_BACKWARD_FILTER_NONREC, 16)]);
141 ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
143 convolve(buffer1, work + order , n , order);
144 convolve(buffer2, work + order + n, non_rec, order);
146 for (i=0; i <= order; i++) {
147 out2[i] = out2[i] * 0.5625 + buffer1[i];
148 out [i] = out2[i] + buffer2[i];
151 /* Multiply by the white noise correcting factor (WNCF). */
156 * Backward synthesis filter, find the LPC coefficients from past speech data.
158 static void backward_filter(RA288Context *ractx,
159 float *hist, float *rec, const float *window,
160 float *lpc, const float *tab,
161 int order, int n, int non_rec, int move_size)
163 float temp[MAX_BACKWARD_FILTER_ORDER+1];
165 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
167 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
168 ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
170 memmove(hist, hist + n, move_size*sizeof(*hist));
173 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
174 int *got_frame_ptr, AVPacket *avpkt)
176 const uint8_t *buf = avpkt->data;
177 int buf_size = avpkt->size;
180 RA288Context *ractx = avctx->priv_data;
183 if (buf_size < avctx->block_align) {
184 av_log(avctx, AV_LOG_ERROR,
185 "Error! Input buffer is too small [%d<%d]\n",
186 buf_size, avctx->block_align);
187 return AVERROR_INVALIDDATA;
190 /* get output buffer */
191 ractx->frame.nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
192 if ((ret = avctx->get_buffer(avctx, &ractx->frame)) < 0) {
193 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
196 out = (float *)ractx->frame.data[0];
198 init_get_bits(&gb, buf, avctx->block_align * 8);
200 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
201 float gain = amptable[get_bits(&gb, 3)];
202 int cb_coef = get_bits(&gb, 6 + (i&1));
204 decode(ractx, gain, cb_coef);
206 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
207 out += RA288_BLOCK_SIZE;
210 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
211 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
213 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
214 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
219 *(AVFrame *)data = ractx->frame;
221 return avctx->block_align;
224 AVCodec ff_ra_288_decoder = {
226 .type = AVMEDIA_TYPE_AUDIO,
227 .id = AV_CODEC_ID_RA_288,
228 .priv_data_size = sizeof(RA288Context),
229 .init = ra288_decode_init,
230 .decode = ra288_decode_frame,
231 .capabilities = CODEC_CAP_DR1,
232 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),