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truespeech.c
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1 /*
2  * DSP Group TrueSpeech compatible decoder
3  * Copyright (c) 2005 Konstantin Shishkov
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 
23 #include "libavutil/intreadwrite.h"
24 #include "avcodec.h"
25 #include "bswapdsp.h"
26 #include "get_bits.h"
27 #include "internal.h"
28 
29 #include "truespeech_data.h"
30 /**
31  * @file
32  * TrueSpeech decoder.
33  */
34 
35 /**
36  * TrueSpeech decoder context
37  */
38 typedef struct {
40  /* input data */
42  int16_t vector[8]; ///< input vector: 5/5/4/4/4/3/3/3
43  int offset1[2]; ///< 8-bit value, used in one copying offset
44  int offset2[4]; ///< 7-bit value, encodes offsets for copying and for two-point filter
45  int pulseoff[4]; ///< 4-bit offset of pulse values block
46  int pulsepos[4]; ///< 27-bit variable, encodes 7 pulse positions
47  int pulseval[4]; ///< 7x2-bit pulse values
48  int flag; ///< 1-bit flag, shows how to choose filters
49  /* temporary data */
50  int filtbuf[146]; // some big vector used for storing filters
51  int prevfilt[8]; // filter from previous frame
52  int16_t tmp1[8]; // coefficients for adding to out
53  int16_t tmp2[8]; // coefficients for adding to out
54  int16_t tmp3[8]; // coefficients for adding to out
55  int16_t cvector[8]; // correlated input vector
56  int filtval; // gain value for one function
57  int16_t newvec[60]; // tmp vector
58  int16_t filters[32]; // filters for every subframe
59 } TSContext;
60 
62 {
63  TSContext *c = avctx->priv_data;
64 
65  if (avctx->channels != 1) {
66  avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
67  return AVERROR_PATCHWELCOME;
68  }
69 
72 
74 
75  return 0;
76 }
77 
78 static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
79 {
80  GetBitContext gb;
81 
82  dec->bdsp.bswap_buf((uint32_t *) dec->buffer, (const uint32_t *) input, 8);
83  init_get_bits(&gb, dec->buffer, 32 * 8);
84 
85  dec->vector[7] = ts_codebook[7][get_bits(&gb, 3)];
86  dec->vector[6] = ts_codebook[6][get_bits(&gb, 3)];
87  dec->vector[5] = ts_codebook[5][get_bits(&gb, 3)];
88  dec->vector[4] = ts_codebook[4][get_bits(&gb, 4)];
89  dec->vector[3] = ts_codebook[3][get_bits(&gb, 4)];
90  dec->vector[2] = ts_codebook[2][get_bits(&gb, 4)];
91  dec->vector[1] = ts_codebook[1][get_bits(&gb, 5)];
92  dec->vector[0] = ts_codebook[0][get_bits(&gb, 5)];
93  dec->flag = get_bits1(&gb);
94 
95  dec->offset1[0] = get_bits(&gb, 4) << 4;
96  dec->offset2[3] = get_bits(&gb, 7);
97  dec->offset2[2] = get_bits(&gb, 7);
98  dec->offset2[1] = get_bits(&gb, 7);
99  dec->offset2[0] = get_bits(&gb, 7);
100 
101  dec->offset1[1] = get_bits(&gb, 4);
102  dec->pulseval[1] = get_bits(&gb, 14);
103  dec->pulseval[0] = get_bits(&gb, 14);
104 
105  dec->offset1[1] |= get_bits(&gb, 4) << 4;
106  dec->pulseval[3] = get_bits(&gb, 14);
107  dec->pulseval[2] = get_bits(&gb, 14);
108 
109  dec->offset1[0] |= get_bits1(&gb);
110  dec->pulsepos[0] = get_bits_long(&gb, 27);
111  dec->pulseoff[0] = get_bits(&gb, 4);
112 
113  dec->offset1[0] |= get_bits1(&gb) << 1;
114  dec->pulsepos[1] = get_bits_long(&gb, 27);
115  dec->pulseoff[1] = get_bits(&gb, 4);
116 
117  dec->offset1[0] |= get_bits1(&gb) << 2;
118  dec->pulsepos[2] = get_bits_long(&gb, 27);
119  dec->pulseoff[2] = get_bits(&gb, 4);
120 
121  dec->offset1[0] |= get_bits1(&gb) << 3;
122  dec->pulsepos[3] = get_bits_long(&gb, 27);
123  dec->pulseoff[3] = get_bits(&gb, 4);
124 }
125 
127 {
128  int16_t tmp[8];
129  int i, j;
130 
131  for(i = 0; i < 8; i++){
132  if(i > 0){
133  memcpy(tmp, dec->cvector, i * sizeof(*tmp));
134  for(j = 0; j < i; j++)
135  dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
136  (dec->cvector[j] << 15) + 0x4000) >> 15;
137  }
138  dec->cvector[i] = (8 - dec->vector[i]) >> 3;
139  }
140  for(i = 0; i < 8; i++)
141  dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
142 
143  dec->filtval = dec->vector[0];
144 }
145 
147 {
148  int i;
149 
150  if(!dec->flag){
151  for(i = 0; i < 8; i++){
152  dec->filters[i + 0] = dec->prevfilt[i];
153  dec->filters[i + 8] = dec->prevfilt[i];
154  }
155  }else{
156  for(i = 0; i < 8; i++){
157  dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
158  dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
159  }
160  }
161  for(i = 0; i < 8; i++){
162  dec->filters[i + 16] = dec->cvector[i];
163  dec->filters[i + 24] = dec->cvector[i];
164  }
165 }
166 
167 static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
168 {
169  int16_t tmp[146 + 60], *ptr0, *ptr1;
170  const int16_t *filter;
171  int i, t, off;
172 
173  t = dec->offset2[quart];
174  if(t == 127){
175  memset(dec->newvec, 0, 60 * sizeof(*dec->newvec));
176  return;
177  }
178  for(i = 0; i < 146; i++)
179  tmp[i] = dec->filtbuf[i];
180  off = (t / 25) + dec->offset1[quart >> 1] + 18;
181  off = av_clip(off, 0, 145);
182  ptr0 = tmp + 145 - off;
183  ptr1 = tmp + 146;
184  filter = ts_order2_coeffs + (t % 25) * 2;
185  for(i = 0; i < 60; i++){
186  t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
187  ptr0++;
188  dec->newvec[i] = t;
189  ptr1[i] = t;
190  }
191 }
192 
193 static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
194 {
195  int16_t tmp[7];
196  int i, j, t;
197  const int16_t *ptr1;
198  int16_t *ptr2;
199  int coef;
200 
201  memset(out, 0, 60 * sizeof(*out));
202  for(i = 0; i < 7; i++) {
203  t = dec->pulseval[quart] & 3;
204  dec->pulseval[quart] >>= 2;
205  tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
206  }
207 
208  coef = dec->pulsepos[quart] >> 15;
209  ptr1 = ts_pulse_values + 30;
210  ptr2 = tmp;
211  for(i = 0, j = 3; (i < 30) && (j > 0); i++){
212  t = *ptr1++;
213  if(coef >= t)
214  coef -= t;
215  else{
216  out[i] = *ptr2++;
217  ptr1 += 30;
218  j--;
219  }
220  }
221  coef = dec->pulsepos[quart] & 0x7FFF;
222  ptr1 = ts_pulse_values;
223  for(i = 30, j = 4; (i < 60) && (j > 0); i++){
224  t = *ptr1++;
225  if(coef >= t)
226  coef -= t;
227  else{
228  out[i] = *ptr2++;
229  ptr1 += 30;
230  j--;
231  }
232  }
233 
234 }
235 
236 static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
237 {
238  int i;
239 
240  memmove(dec->filtbuf, &dec->filtbuf[60], 86 * sizeof(*dec->filtbuf));
241  for(i = 0; i < 60; i++){
242  dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
243  out[i] += dec->newvec[i];
244  }
245 }
246 
247 static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
248 {
249  int i,k;
250  int t[8];
251  int16_t *ptr0, *ptr1;
252 
253  ptr0 = dec->tmp1;
254  ptr1 = dec->filters + quart * 8;
255  for(i = 0; i < 60; i++){
256  int sum = 0;
257  for(k = 0; k < 8; k++)
258  sum += ptr0[k] * ptr1[k];
259  sum = (sum + (out[i] << 12) + 0x800) >> 12;
260  out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
261  for(k = 7; k > 0; k--)
262  ptr0[k] = ptr0[k - 1];
263  ptr0[0] = out[i];
264  }
265 
266  for(i = 0; i < 8; i++)
267  t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
268 
269  ptr0 = dec->tmp2;
270  for(i = 0; i < 60; i++){
271  int sum = 0;
272  for(k = 0; k < 8; k++)
273  sum += ptr0[k] * t[k];
274  for(k = 7; k > 0; k--)
275  ptr0[k] = ptr0[k - 1];
276  ptr0[0] = out[i];
277  out[i] = ((out[i] << 12) - sum) >> 12;
278  }
279 
280  for(i = 0; i < 8; i++)
281  t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
282 
283  ptr0 = dec->tmp3;
284  for(i = 0; i < 60; i++){
285  int sum = out[i] << 12;
286  for(k = 0; k < 8; k++)
287  sum += ptr0[k] * t[k];
288  for(k = 7; k > 0; k--)
289  ptr0[k] = ptr0[k - 1];
290  ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
291 
292  sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
293  sum = sum - (sum >> 3);
294  out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
295  }
296 }
297 
299 {
300  int i;
301 
302  for(i = 0; i < 8; i++)
303  c->prevfilt[i] = c->cvector[i];
304 }
305 
306 static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
307  int *got_frame_ptr, AVPacket *avpkt)
308 {
309  AVFrame *frame = data;
310  const uint8_t *buf = avpkt->data;
311  int buf_size = avpkt->size;
312  TSContext *c = avctx->priv_data;
313 
314  int i, j;
315  int16_t *samples;
316  int iterations, ret;
317 
318  iterations = buf_size / 32;
319 
320  if (!iterations) {
321  av_log(avctx, AV_LOG_ERROR,
322  "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
323  return -1;
324  }
325 
326  /* get output buffer */
327  frame->nb_samples = iterations * 240;
328  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
329  return ret;
330  samples = (int16_t *)frame->data[0];
331 
332  memset(samples, 0, iterations * 240 * sizeof(*samples));
333 
334  for(j = 0; j < iterations; j++) {
335  truespeech_read_frame(c, buf);
336  buf += 32;
337 
340 
341  for(i = 0; i < 4; i++) {
343  truespeech_place_pulses (c, samples, i);
344  truespeech_update_filters(c, samples, i);
345  truespeech_synth (c, samples, i);
346  samples += 60;
347  }
348 
350  }
351 
352  *got_frame_ptr = 1;
353 
354  return buf_size;
355 }
356 
358  .name = "truespeech",
359  .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
360  .type = AVMEDIA_TYPE_AUDIO,
362  .priv_data_size = sizeof(TSContext),
365  .capabilities = CODEC_CAP_DR1,
366 };