avprobe: also output dar/par if only defined in stream
[FFMpeg-mirror/mplayer-patches.git] / libavcodec / truespeech.c
blob486e41f895d8cd6587c67cc40c828c4e1d404229
1 /*
2 * DSP Group TrueSpeech compatible decoder
3 * Copyright (c) 2005 Konstantin Shishkov
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/intreadwrite.h"
24 #include "avcodec.h"
25 #include "dsputil.h"
26 #include "get_bits.h"
27 #include "internal.h"
29 #include "truespeech_data.h"
30 /**
31 * @file
32 * TrueSpeech decoder.
35 /**
36 * TrueSpeech decoder context
38 typedef struct {
39 AVFrame frame;
40 DSPContext dsp;
41 /* input data */
42 DECLARE_ALIGNED(16, uint8_t, buffer)[32];
43 int16_t vector[8]; ///< input vector: 5/5/4/4/4/3/3/3
44 int offset1[2]; ///< 8-bit value, used in one copying offset
45 int offset2[4]; ///< 7-bit value, encodes offsets for copying and for two-point filter
46 int pulseoff[4]; ///< 4-bit offset of pulse values block
47 int pulsepos[4]; ///< 27-bit variable, encodes 7 pulse positions
48 int pulseval[4]; ///< 7x2-bit pulse values
49 int flag; ///< 1-bit flag, shows how to choose filters
50 /* temporary data */
51 int filtbuf[146]; // some big vector used for storing filters
52 int prevfilt[8]; // filter from previous frame
53 int16_t tmp1[8]; // coefficients for adding to out
54 int16_t tmp2[8]; // coefficients for adding to out
55 int16_t tmp3[8]; // coefficients for adding to out
56 int16_t cvector[8]; // correlated input vector
57 int filtval; // gain value for one function
58 int16_t newvec[60]; // tmp vector
59 int16_t filters[32]; // filters for every subframe
60 } TSContext;
62 static av_cold int truespeech_decode_init(AVCodecContext * avctx)
64 TSContext *c = avctx->priv_data;
66 if (avctx->channels != 1) {
67 av_log_ask_for_sample(avctx, "Unsupported channel count: %d\n", avctx->channels);
68 return AVERROR_PATCHWELCOME;
71 avctx->channel_layout = AV_CH_LAYOUT_MONO;
72 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
74 ff_dsputil_init(&c->dsp, avctx);
76 avcodec_get_frame_defaults(&c->frame);
77 avctx->coded_frame = &c->frame;
79 return 0;
82 static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
84 GetBitContext gb;
86 dec->dsp.bswap_buf((uint32_t *)dec->buffer, (const uint32_t *)input, 8);
87 init_get_bits(&gb, dec->buffer, 32 * 8);
89 dec->vector[7] = ts_codebook[7][get_bits(&gb, 3)];
90 dec->vector[6] = ts_codebook[6][get_bits(&gb, 3)];
91 dec->vector[5] = ts_codebook[5][get_bits(&gb, 3)];
92 dec->vector[4] = ts_codebook[4][get_bits(&gb, 4)];
93 dec->vector[3] = ts_codebook[3][get_bits(&gb, 4)];
94 dec->vector[2] = ts_codebook[2][get_bits(&gb, 4)];
95 dec->vector[1] = ts_codebook[1][get_bits(&gb, 5)];
96 dec->vector[0] = ts_codebook[0][get_bits(&gb, 5)];
97 dec->flag = get_bits1(&gb);
99 dec->offset1[0] = get_bits(&gb, 4) << 4;
100 dec->offset2[3] = get_bits(&gb, 7);
101 dec->offset2[2] = get_bits(&gb, 7);
102 dec->offset2[1] = get_bits(&gb, 7);
103 dec->offset2[0] = get_bits(&gb, 7);
105 dec->offset1[1] = get_bits(&gb, 4);
106 dec->pulseval[1] = get_bits(&gb, 14);
107 dec->pulseval[0] = get_bits(&gb, 14);
109 dec->offset1[1] |= get_bits(&gb, 4) << 4;
110 dec->pulseval[3] = get_bits(&gb, 14);
111 dec->pulseval[2] = get_bits(&gb, 14);
113 dec->offset1[0] |= get_bits1(&gb);
114 dec->pulsepos[0] = get_bits_long(&gb, 27);
115 dec->pulseoff[0] = get_bits(&gb, 4);
117 dec->offset1[0] |= get_bits1(&gb) << 1;
118 dec->pulsepos[1] = get_bits_long(&gb, 27);
119 dec->pulseoff[1] = get_bits(&gb, 4);
121 dec->offset1[0] |= get_bits1(&gb) << 2;
122 dec->pulsepos[2] = get_bits_long(&gb, 27);
123 dec->pulseoff[2] = get_bits(&gb, 4);
125 dec->offset1[0] |= get_bits1(&gb) << 3;
126 dec->pulsepos[3] = get_bits_long(&gb, 27);
127 dec->pulseoff[3] = get_bits(&gb, 4);
130 static void truespeech_correlate_filter(TSContext *dec)
132 int16_t tmp[8];
133 int i, j;
135 for(i = 0; i < 8; i++){
136 if(i > 0){
137 memcpy(tmp, dec->cvector, i * sizeof(*tmp));
138 for(j = 0; j < i; j++)
139 dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
140 (dec->cvector[j] << 15) + 0x4000) >> 15;
142 dec->cvector[i] = (8 - dec->vector[i]) >> 3;
144 for(i = 0; i < 8; i++)
145 dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
147 dec->filtval = dec->vector[0];
150 static void truespeech_filters_merge(TSContext *dec)
152 int i;
154 if(!dec->flag){
155 for(i = 0; i < 8; i++){
156 dec->filters[i + 0] = dec->prevfilt[i];
157 dec->filters[i + 8] = dec->prevfilt[i];
159 }else{
160 for(i = 0; i < 8; i++){
161 dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
162 dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
165 for(i = 0; i < 8; i++){
166 dec->filters[i + 16] = dec->cvector[i];
167 dec->filters[i + 24] = dec->cvector[i];
171 static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
173 int16_t tmp[146 + 60], *ptr0, *ptr1;
174 const int16_t *filter;
175 int i, t, off;
177 t = dec->offset2[quart];
178 if(t == 127){
179 memset(dec->newvec, 0, 60 * sizeof(*dec->newvec));
180 return;
182 for(i = 0; i < 146; i++)
183 tmp[i] = dec->filtbuf[i];
184 off = (t / 25) + dec->offset1[quart >> 1] + 18;
185 off = av_clip(off, 0, 145);
186 ptr0 = tmp + 145 - off;
187 ptr1 = tmp + 146;
188 filter = ts_order2_coeffs + (t % 25) * 2;
189 for(i = 0; i < 60; i++){
190 t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
191 ptr0++;
192 dec->newvec[i] = t;
193 ptr1[i] = t;
197 static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
199 int16_t tmp[7];
200 int i, j, t;
201 const int16_t *ptr1;
202 int16_t *ptr2;
203 int coef;
205 memset(out, 0, 60 * sizeof(*out));
206 for(i = 0; i < 7; i++) {
207 t = dec->pulseval[quart] & 3;
208 dec->pulseval[quart] >>= 2;
209 tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
212 coef = dec->pulsepos[quart] >> 15;
213 ptr1 = ts_pulse_values + 30;
214 ptr2 = tmp;
215 for(i = 0, j = 3; (i < 30) && (j > 0); i++){
216 t = *ptr1++;
217 if(coef >= t)
218 coef -= t;
219 else{
220 out[i] = *ptr2++;
221 ptr1 += 30;
222 j--;
225 coef = dec->pulsepos[quart] & 0x7FFF;
226 ptr1 = ts_pulse_values;
227 for(i = 30, j = 4; (i < 60) && (j > 0); i++){
228 t = *ptr1++;
229 if(coef >= t)
230 coef -= t;
231 else{
232 out[i] = *ptr2++;
233 ptr1 += 30;
234 j--;
240 static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
242 int i;
244 memmove(dec->filtbuf, &dec->filtbuf[60], 86 * sizeof(*dec->filtbuf));
245 for(i = 0; i < 60; i++){
246 dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
247 out[i] += dec->newvec[i];
251 static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
253 int i,k;
254 int t[8];
255 int16_t *ptr0, *ptr1;
257 ptr0 = dec->tmp1;
258 ptr1 = dec->filters + quart * 8;
259 for(i = 0; i < 60; i++){
260 int sum = 0;
261 for(k = 0; k < 8; k++)
262 sum += ptr0[k] * ptr1[k];
263 sum = (sum + (out[i] << 12) + 0x800) >> 12;
264 out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
265 for(k = 7; k > 0; k--)
266 ptr0[k] = ptr0[k - 1];
267 ptr0[0] = out[i];
270 for(i = 0; i < 8; i++)
271 t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
273 ptr0 = dec->tmp2;
274 for(i = 0; i < 60; i++){
275 int sum = 0;
276 for(k = 0; k < 8; k++)
277 sum += ptr0[k] * t[k];
278 for(k = 7; k > 0; k--)
279 ptr0[k] = ptr0[k - 1];
280 ptr0[0] = out[i];
281 out[i] = ((out[i] << 12) - sum) >> 12;
284 for(i = 0; i < 8; i++)
285 t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
287 ptr0 = dec->tmp3;
288 for(i = 0; i < 60; i++){
289 int sum = out[i] << 12;
290 for(k = 0; k < 8; k++)
291 sum += ptr0[k] * t[k];
292 for(k = 7; k > 0; k--)
293 ptr0[k] = ptr0[k - 1];
294 ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
296 sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
297 sum = sum - (sum >> 3);
298 out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
302 static void truespeech_save_prevvec(TSContext *c)
304 int i;
306 for(i = 0; i < 8; i++)
307 c->prevfilt[i] = c->cvector[i];
310 static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
311 int *got_frame_ptr, AVPacket *avpkt)
313 const uint8_t *buf = avpkt->data;
314 int buf_size = avpkt->size;
315 TSContext *c = avctx->priv_data;
317 int i, j;
318 int16_t *samples;
319 int iterations, ret;
321 iterations = buf_size / 32;
323 if (!iterations) {
324 av_log(avctx, AV_LOG_ERROR,
325 "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
326 return -1;
329 /* get output buffer */
330 c->frame.nb_samples = iterations * 240;
331 if ((ret = ff_get_buffer(avctx, &c->frame)) < 0) {
332 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
333 return ret;
335 samples = (int16_t *)c->frame.data[0];
337 memset(samples, 0, iterations * 240 * sizeof(*samples));
339 for(j = 0; j < iterations; j++) {
340 truespeech_read_frame(c, buf);
341 buf += 32;
343 truespeech_correlate_filter(c);
344 truespeech_filters_merge(c);
346 for(i = 0; i < 4; i++) {
347 truespeech_apply_twopoint_filter(c, i);
348 truespeech_place_pulses (c, samples, i);
349 truespeech_update_filters(c, samples, i);
350 truespeech_synth (c, samples, i);
351 samples += 60;
354 truespeech_save_prevvec(c);
357 *got_frame_ptr = 1;
358 *(AVFrame *)data = c->frame;
360 return buf_size;
363 AVCodec ff_truespeech_decoder = {
364 .name = "truespeech",
365 .type = AVMEDIA_TYPE_AUDIO,
366 .id = AV_CODEC_ID_TRUESPEECH,
367 .priv_data_size = sizeof(TSContext),
368 .init = truespeech_decode_init,
369 .decode = truespeech_decode_frame,
370 .capabilities = CODEC_CAP_DR1,
371 .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),