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2ba45a60 DM |
1 | /* |
2 | * RealAudio 2.0 (28.8K) | |
3 | * Copyright (c) 2003 The FFmpeg Project | |
4 | * | |
5 | * This file is part of FFmpeg. | |
6 | * | |
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. | |
11 | * | |
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. | |
16 | * | |
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 | |
20 | */ | |
21 | ||
22 | #include "libavutil/channel_layout.h" | |
23 | #include "libavutil/float_dsp.h" | |
24 | #include "libavutil/internal.h" | |
25 | #include "avcodec.h" | |
26 | #include "internal.h" | |
27 | #define BITSTREAM_READER_LE | |
28 | #include "get_bits.h" | |
29 | #include "ra288.h" | |
30 | #include "lpc.h" | |
31 | #include "celp_filters.h" | |
32 | ||
33 | #define MAX_BACKWARD_FILTER_ORDER 36 | |
34 | #define MAX_BACKWARD_FILTER_LEN 40 | |
35 | #define MAX_BACKWARD_FILTER_NONREC 35 | |
36 | ||
37 | #define RA288_BLOCK_SIZE 5 | |
38 | #define RA288_BLOCKS_PER_FRAME 32 | |
39 | ||
40 | typedef struct { | |
f6fa7814 | 41 | AVFloatDSPContext *fdsp; |
2ba45a60 DM |
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) | |
44 | ||
45 | /** speech data history (spec: SB). | |
46 | * Its first 70 coefficients are updated only at backward filtering. | |
47 | */ | |
48 | float sp_hist[111]; | |
49 | ||
50 | /// speech part of the gain autocorrelation (spec: REXP) | |
51 | float sp_rec[37]; | |
52 | ||
53 | /** log-gain history (spec: SBLG). | |
54 | * Its first 28 coefficients are updated only at backward filtering. | |
55 | */ | |
56 | float gain_hist[38]; | |
57 | ||
58 | /// recursive part of the gain autocorrelation (spec: REXPLG) | |
59 | float gain_rec[11]; | |
60 | } RA288Context; | |
61 | ||
f6fa7814 DM |
62 | static av_cold int ra288_decode_close(AVCodecContext *avctx) |
63 | { | |
64 | RA288Context *ractx = avctx->priv_data; | |
65 | ||
66 | av_freep(&ractx->fdsp); | |
67 | ||
68 | return 0; | |
69 | } | |
70 | ||
2ba45a60 DM |
71 | static av_cold int ra288_decode_init(AVCodecContext *avctx) |
72 | { | |
73 | RA288Context *ractx = avctx->priv_data; | |
74 | ||
75 | avctx->channels = 1; | |
76 | avctx->channel_layout = AV_CH_LAYOUT_MONO; | |
77 | avctx->sample_fmt = AV_SAMPLE_FMT_FLT; | |
78 | ||
79 | if (avctx->block_align <= 0) { | |
80 | av_log(avctx, AV_LOG_ERROR, "unsupported block align\n"); | |
81 | return AVERROR_PATCHWELCOME; | |
82 | } | |
83 | ||
f6fa7814 DM |
84 | ractx->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); |
85 | if (!ractx->fdsp) | |
86 | return AVERROR(ENOMEM); | |
2ba45a60 DM |
87 | |
88 | return 0; | |
89 | } | |
90 | ||
91 | static void convolve(float *tgt, const float *src, int len, int n) | |
92 | { | |
93 | for (; n >= 0; n--) | |
94 | tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len); | |
95 | ||
96 | } | |
97 | ||
98 | static void decode(RA288Context *ractx, float gain, int cb_coef) | |
99 | { | |
100 | int i; | |
101 | double sumsum; | |
102 | float sum, buffer[5]; | |
103 | float *block = ractx->sp_hist + 70 + 36; // current block | |
104 | float *gain_block = ractx->gain_hist + 28; | |
105 | ||
106 | memmove(ractx->sp_hist + 70, ractx->sp_hist + 75, 36*sizeof(*block)); | |
107 | ||
108 | /* block 46 of G.728 spec */ | |
109 | sum = 32.0; | |
110 | for (i=0; i < 10; i++) | |
111 | sum -= gain_block[9-i] * ractx->gain_lpc[i]; | |
112 | ||
113 | /* block 47 of G.728 spec */ | |
114 | sum = av_clipf(sum, 0, 60); | |
115 | ||
116 | /* block 48 of G.728 spec */ | |
117 | /* exp(sum * 0.1151292546497) == pow(10.0,sum/20) */ | |
118 | sumsum = exp(sum * 0.1151292546497) * gain * (1.0/(1<<23)); | |
119 | ||
120 | for (i=0; i < 5; i++) | |
121 | buffer[i] = codetable[cb_coef][i] * sumsum; | |
122 | ||
123 | sum = avpriv_scalarproduct_float_c(buffer, buffer, 5); | |
124 | ||
125 | sum = FFMAX(sum, 5.0 / (1<<24)); | |
126 | ||
127 | /* shift and store */ | |
128 | memmove(gain_block, gain_block + 1, 9 * sizeof(*gain_block)); | |
129 | ||
130 | gain_block[9] = 10 * log10(sum) + (10*log10(((1<<24)/5.)) - 32); | |
131 | ||
132 | ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36); | |
133 | } | |
134 | ||
135 | /** | |
136 | * Hybrid window filtering, see blocks 36 and 49 of the G.728 specification. | |
137 | * | |
138 | * @param order filter order | |
139 | * @param n input length | |
140 | * @param non_rec number of non-recursive samples | |
141 | * @param out filter output | |
142 | * @param hist pointer to the input history of the filter | |
143 | * @param out pointer to the non-recursive part of the output | |
144 | * @param out2 pointer to the recursive part of the output | |
145 | * @param window pointer to the windowing function table | |
146 | */ | |
147 | static void do_hybrid_window(RA288Context *ractx, | |
148 | int order, int n, int non_rec, float *out, | |
149 | float *hist, float *out2, const float *window) | |
150 | { | |
151 | int i; | |
152 | float buffer1[MAX_BACKWARD_FILTER_ORDER + 1]; | |
153 | float buffer2[MAX_BACKWARD_FILTER_ORDER + 1]; | |
154 | LOCAL_ALIGNED(32, float, work, [FFALIGN(MAX_BACKWARD_FILTER_ORDER + | |
155 | MAX_BACKWARD_FILTER_LEN + | |
156 | MAX_BACKWARD_FILTER_NONREC, 16)]); | |
157 | ||
158 | av_assert2(order>=0); | |
159 | ||
f6fa7814 | 160 | ractx->fdsp->vector_fmul(work, window, hist, FFALIGN(order + n + non_rec, 16)); |
2ba45a60 DM |
161 | |
162 | convolve(buffer1, work + order , n , order); | |
163 | convolve(buffer2, work + order + n, non_rec, order); | |
164 | ||
165 | for (i=0; i <= order; i++) { | |
166 | out2[i] = out2[i] * 0.5625 + buffer1[i]; | |
167 | out [i] = out2[i] + buffer2[i]; | |
168 | } | |
169 | ||
170 | /* Multiply by the white noise correcting factor (WNCF). */ | |
171 | *out *= 257.0 / 256.0; | |
172 | } | |
173 | ||
174 | /** | |
175 | * Backward synthesis filter, find the LPC coefficients from past speech data. | |
176 | */ | |
177 | static void backward_filter(RA288Context *ractx, | |
178 | float *hist, float *rec, const float *window, | |
179 | float *lpc, const float *tab, | |
180 | int order, int n, int non_rec, int move_size) | |
181 | { | |
182 | float temp[MAX_BACKWARD_FILTER_ORDER+1]; | |
183 | ||
184 | do_hybrid_window(ractx, order, n, non_rec, temp, hist, rec, window); | |
185 | ||
186 | if (!compute_lpc_coefs(temp, order, lpc, 0, 1, 1)) | |
f6fa7814 | 187 | ractx->fdsp->vector_fmul(lpc, lpc, tab, FFALIGN(order, 16)); |
2ba45a60 DM |
188 | |
189 | memmove(hist, hist + n, move_size*sizeof(*hist)); | |
190 | } | |
191 | ||
192 | static int ra288_decode_frame(AVCodecContext * avctx, void *data, | |
193 | int *got_frame_ptr, AVPacket *avpkt) | |
194 | { | |
195 | AVFrame *frame = data; | |
196 | const uint8_t *buf = avpkt->data; | |
197 | int buf_size = avpkt->size; | |
198 | float *out; | |
199 | int i, ret; | |
200 | RA288Context *ractx = avctx->priv_data; | |
201 | GetBitContext gb; | |
202 | ||
203 | if (buf_size < avctx->block_align) { | |
204 | av_log(avctx, AV_LOG_ERROR, | |
205 | "Error! Input buffer is too small [%d<%d]\n", | |
206 | buf_size, avctx->block_align); | |
207 | return AVERROR_INVALIDDATA; | |
208 | } | |
209 | ||
210 | /* get output buffer */ | |
211 | frame->nb_samples = RA288_BLOCK_SIZE * RA288_BLOCKS_PER_FRAME; | |
212 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) | |
213 | return ret; | |
214 | out = (float *)frame->data[0]; | |
215 | ||
216 | init_get_bits8(&gb, buf, avctx->block_align); | |
217 | ||
218 | for (i=0; i < RA288_BLOCKS_PER_FRAME; i++) { | |
219 | float gain = amptable[get_bits(&gb, 3)]; | |
220 | int cb_coef = get_bits(&gb, 6 + (i&1)); | |
221 | ||
222 | decode(ractx, gain, cb_coef); | |
223 | ||
224 | memcpy(out, &ractx->sp_hist[70 + 36], RA288_BLOCK_SIZE * sizeof(*out)); | |
225 | out += RA288_BLOCK_SIZE; | |
226 | ||
227 | if ((i & 7) == 3) { | |
228 | backward_filter(ractx, ractx->sp_hist, ractx->sp_rec, syn_window, | |
229 | ractx->sp_lpc, syn_bw_tab, 36, 40, 35, 70); | |
230 | ||
231 | backward_filter(ractx, ractx->gain_hist, ractx->gain_rec, gain_window, | |
232 | ractx->gain_lpc, gain_bw_tab, 10, 8, 20, 28); | |
233 | } | |
234 | } | |
235 | ||
236 | *got_frame_ptr = 1; | |
237 | ||
238 | return avctx->block_align; | |
239 | } | |
240 | ||
241 | AVCodec ff_ra_288_decoder = { | |
242 | .name = "real_288", | |
243 | .long_name = NULL_IF_CONFIG_SMALL("RealAudio 2.0 (28.8K)"), | |
244 | .type = AVMEDIA_TYPE_AUDIO, | |
245 | .id = AV_CODEC_ID_RA_288, | |
246 | .priv_data_size = sizeof(RA288Context), | |
247 | .init = ra288_decode_init, | |
248 | .decode = ra288_decode_frame, | |
f6fa7814 | 249 | .close = ra288_decode_close, |
2ba45a60 DM |
250 | .capabilities = CODEC_CAP_DR1, |
251 | }; |