2 * RealAudio 2.0 (28.8K)
3 * Copyright (c) 2003 The FFmpeg project
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/float_dsp.h"
24 #include "libavutil/internal.h"
26 #define BITSTREAM_READER_LE
28 #include "bitstream.h"
29 #include "celp_filters.h"
34 #define MAX_BACKWARD_FILTER_ORDER 36
35 #define MAX_BACKWARD_FILTER_LEN 40
36 #define MAX_BACKWARD_FILTER_NONREC 35
38 #define RA288_BLOCK_SIZE 5
39 #define RA288_BLOCKS_PER_FRAME 32
41 typedef struct RA288Context {
42 AVFloatDSPContext fdsp;
43 DECLARE_ALIGNED(32, float, sp_lpc)[FFALIGN(36, 16)]; ///< LPC coefficients for speech data (spec: A)
44 DECLARE_ALIGNED(32, float, gain_lpc)[FFALIGN(10, 16)]; ///< LPC coefficients for gain (spec: GB)
46 /** speech data history (spec: SB).
47 * Its first 70 coefficients are updated only at backward filtering.
51 /// speech part of the gain autocorrelation (spec: REXP)
54 /** log-gain history (spec: SBLG).
55 * Its first 28 coefficients are updated only at backward filtering.
59 /// recursive part of the gain autocorrelation (spec: REXPLG)
63 static av_cold int ra288_decode_init(AVCodecContext *avctx)
65 RA288Context *ractx = avctx->priv_data;
68 avctx->channel_layout = AV_CH_LAYOUT_MONO;
69 avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
71 avpriv_float_dsp_init(&ractx->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT);
76 static void convolve(float *tgt, const float *src, int len, int n)
79 tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
83 static void decode(RA288Context *ractx, float gain, int cb_coef)
88 float *block = ractx->sp_hist + 70 + 36; // current block
89 float *gain_block = ractx->gain_hist + 28;
91 memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block));
93 /* block 46 of G.728 spec */
95 for (i=0; i < 10; i++)
96 sum -= gain_block[9-i] * ractx->gain_lpc[i];
98 /* block 47 of G.728 spec */
99 sum = av_clipf(sum, 0, 60);
101 /* block 48 of G.728 spec */
102 /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */
103 sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23));
105 for (i=0; i < 5; i++)
106 buffer[i] = codetable[cb_coef][i] * sumsum;
108 sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.0);
112 /* shift and store */
113 memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block));
115 gain_block[9] = 10 * log10(sum) - 32;
117 ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
121 * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification.
123 * @param order filter order
124 * @param n input length
125 * @param non_rec number of non-recursive samples
126 * @param out filter output
127 * @param hist pointer to the input history of the filter
128 * @param out pointer to the non-recursive part of the output
129 * @param out2 pointer to the recursive part of the output
130 * @param window pointer to the windowing function table
132 static void do_hybrid_window(RA288Context *ractx,
133 int order, int n, int non_rec, float *out,
134 float *hist, float *out2, const float *window)
137 float buffer1[MAX_BACKWARD_FILTER_ORDER + 1];
138 float buffer2[MAX_BACKWARD_FILTER_ORDER + 1];
139 LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER +
140 MAX_BACKWARD_FILTER_LEN +
141 MAX_BACKWARD_FILTER_NONREC, 16)]);
143 ractx->fdsp.vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16));
145 convolve(buffer1, work + order , n , order);
146 convolve(buffer2, work + order + n, non_rec, order);
148 for (i=0; i <= order; i++) {
149 out2[i] = out2[i] * 0.5625 + buffer1[i];
150 out [i] = out2[i] + buffer2[i];
153 /* Multiply by the white noise correcting factor (WNCF). */
154 *out *= 257.0 / 256.0;
158 * Backward synthesis filter, find the LPC coefficients from past speech data.
160 static void backward_filter(RA288Context *ractx,
161 float *hist, float *rec, const float *window,
162 float *lpc, const float *tab,
163 int order, int n, int non_rec, int move_size)
165 float temp[MAX_BACKWARD_FILTER_ORDER+1];
167 do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window);
169 if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1))
170 ractx->fdsp.vector_fmul(lpc, lpc, tab, FFALIGN(order, 16));
172 memmove(hist, hist + n, move_size*sizeof(*hist));
175 static int ra288_decode_frame(AVCodecContext * avctx, void *data,
176 int *got_frame_ptr, AVPacket *avpkt)
178 AVFrame *frame = data;
179 const uint8_t *buf = avpkt->data;
180 int buf_size = avpkt->size;
183 RA288Context *ractx = avctx->priv_data;
186 if (buf_size < avctx->block_align) {
187 av_log(avctx, AV_LOG_ERROR,
188 "Error! Input buffer is too small [%d<%d]\n",
189 buf_size, avctx->block_align);
190 return AVERROR_INVALIDDATA;
193 /* get output buffer */
194 frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME;
195 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
196 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
199 out = (float *)frame->data[0];
201 bitstream_init8(&bc, buf, avctx->block_align);
203 for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) {
204 float gain = amptable[bitstream_read(&bc, 3)];
205 int cb_coef = bitstream_read(&bc, 6 + (i & 1));
207 decode(ractx, gain, cb_coef);
209 memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out));
210 out += RA288_BLOCK_SIZE;
213 backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window,
214 ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70);
216 backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window,
217 ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28);
223 return avctx->block_align;
226 AVCodec ff_ra_288_decoder = {
228 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"),
229 .type = AVMEDIA_TYPE_AUDIO,
230 .id = AV_CODEC_ID_RA_288,
231 .priv_data_size = sizeof(RA288Context),
232 .init = ra288_decode_init,
233 .decode = ra288_decode_frame,
234 .capabilities = AV_CODEC_CAP_DR1,