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  1. /*
  2. * SIPR / ACELP.NET decoder
  3. *
  4. * Copyright (c) 2008 Vladimir Voroshilov
  5. * Copyright (c) 2009 Vitor Sessak
  6. *
  7. * This file is part of FFmpeg.
  8. *
  9. * FFmpeg is free software; you can redistribute it and/or
  10. * modify it under the terms of the GNU Lesser General Public
  11. * License as published by the Free Software Foundation; either
  12. * version 2.1 of the License, or (at your option) any later version.
  13. *
  14. * FFmpeg is distributed in the hope that it will be useful,
  15. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  16. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  17. * Lesser General Public License for more details.
  18. *
  19. * You should have received a copy of the GNU Lesser General Public
  20. * License along with FFmpeg; if not, write to the Free Software
  21. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  22. */
  23. #include <math.h>
  24. #include <stdint.h>
  25. #include <string.h>
  26. #include "libavutil/mathematics.h"
  27. #include "avcodec.h"
  28. #define BITSTREAM_READER_LE
  29. #include "get_bits.h"
  30. #include "dsputil.h"
  31. #include "lsp.h"
  32. #include "celp_math.h"
  33. #include "acelp_vectors.h"
  34. #include "acelp_pitch_delay.h"
  35. #include "acelp_filters.h"
  36. #include "celp_filters.h"
  37. #define MAX_SUBFRAME_COUNT 5
  38. #include "sipr.h"
  39. #include "siprdata.h"
  40. typedef struct {
  41. const char *mode_name;
  42. uint16_t bits_per_frame;
  43. uint8_t subframe_count;
  44. uint8_t frames_per_packet;
  45. float pitch_sharp_factor;
  46. /* bitstream parameters */
  47. uint8_t number_of_fc_indexes;
  48. uint8_t ma_predictor_bits; ///< size in bits of the switched MA predictor
  49. /** size in bits of the i-th stage vector of quantizer */
  50. uint8_t vq_indexes_bits[5];
  51. /** size in bits of the adaptive-codebook index for every subframe */
  52. uint8_t pitch_delay_bits[5];
  53. uint8_t gp_index_bits;
  54. uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
  55. uint8_t gc_index_bits; ///< size in bits of the gain codebook indexes
  56. } SiprModeParam;
  57. static const SiprModeParam modes[MODE_COUNT] = {
  58. [MODE_16k] = {
  59. .mode_name = "16k",
  60. .bits_per_frame = 160,
  61. .subframe_count = SUBFRAME_COUNT_16k,
  62. .frames_per_packet = 1,
  63. .pitch_sharp_factor = 0.00,
  64. .number_of_fc_indexes = 10,
  65. .ma_predictor_bits = 1,
  66. .vq_indexes_bits = {7, 8, 7, 7, 7},
  67. .pitch_delay_bits = {9, 6},
  68. .gp_index_bits = 4,
  69. .fc_index_bits = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
  70. .gc_index_bits = 5
  71. },
  72. [MODE_8k5] = {
  73. .mode_name = "8k5",
  74. .bits_per_frame = 152,
  75. .subframe_count = 3,
  76. .frames_per_packet = 1,
  77. .pitch_sharp_factor = 0.8,
  78. .number_of_fc_indexes = 3,
  79. .ma_predictor_bits = 0,
  80. .vq_indexes_bits = {6, 7, 7, 7, 5},
  81. .pitch_delay_bits = {8, 5, 5},
  82. .gp_index_bits = 0,
  83. .fc_index_bits = {9, 9, 9},
  84. .gc_index_bits = 7
  85. },
  86. [MODE_6k5] = {
  87. .mode_name = "6k5",
  88. .bits_per_frame = 232,
  89. .subframe_count = 3,
  90. .frames_per_packet = 2,
  91. .pitch_sharp_factor = 0.8,
  92. .number_of_fc_indexes = 3,
  93. .ma_predictor_bits = 0,
  94. .vq_indexes_bits = {6, 7, 7, 7, 5},
  95. .pitch_delay_bits = {8, 5, 5},
  96. .gp_index_bits = 0,
  97. .fc_index_bits = {5, 5, 5},
  98. .gc_index_bits = 7
  99. },
  100. [MODE_5k0] = {
  101. .mode_name = "5k0",
  102. .bits_per_frame = 296,
  103. .subframe_count = 5,
  104. .frames_per_packet = 2,
  105. .pitch_sharp_factor = 0.85,
  106. .number_of_fc_indexes = 1,
  107. .ma_predictor_bits = 0,
  108. .vq_indexes_bits = {6, 7, 7, 7, 5},
  109. .pitch_delay_bits = {8, 5, 8, 5, 5},
  110. .gp_index_bits = 0,
  111. .fc_index_bits = {10},
  112. .gc_index_bits = 7
  113. }
  114. };
  115. const float ff_pow_0_5[] = {
  116. 1.0/(1 << 1), 1.0/(1 << 2), 1.0/(1 << 3), 1.0/(1 << 4),
  117. 1.0/(1 << 5), 1.0/(1 << 6), 1.0/(1 << 7), 1.0/(1 << 8),
  118. 1.0/(1 << 9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
  119. 1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
  120. };
  121. static void dequant(float *out, const int *idx, const float *cbs[])
  122. {
  123. int i;
  124. int stride = 2;
  125. int num_vec = 5;
  126. for (i = 0; i < num_vec; i++)
  127. memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
  128. }
  129. static void lsf_decode_fp(float *lsfnew, float *lsf_history,
  130. const SiprParameters *parm)
  131. {
  132. int i;
  133. float lsf_tmp[LP_FILTER_ORDER];
  134. dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
  135. for (i = 0; i < LP_FILTER_ORDER; i++)
  136. lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
  137. ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1);
  138. /* Note that a minimum distance is not enforced between the last value and
  139. the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
  140. ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1);
  141. lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
  142. memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
  143. for (i = 0; i < LP_FILTER_ORDER - 1; i++)
  144. lsfnew[i] = cos(lsfnew[i]);
  145. lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
  146. }
  147. /** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
  148. static void pitch_sharpening(int pitch_lag_int, float beta,
  149. float *fixed_vector)
  150. {
  151. int i;
  152. for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
  153. fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
  154. }
  155. /**
  156. * Extract decoding parameters from the input bitstream.
  157. * @param parms parameters structure
  158. * @param pgb pointer to initialized GetBitContext structure
  159. */
  160. static void decode_parameters(SiprParameters* parms, GetBitContext *pgb,
  161. const SiprModeParam *p)
  162. {
  163. int i, j;
  164. if (p->ma_predictor_bits)
  165. parms->ma_pred_switch = get_bits(pgb, p->ma_predictor_bits);
  166. for (i = 0; i < 5; i++)
  167. parms->vq_indexes[i] = get_bits(pgb, p->vq_indexes_bits[i]);
  168. for (i = 0; i < p->subframe_count; i++) {
  169. parms->pitch_delay[i] = get_bits(pgb, p->pitch_delay_bits[i]);
  170. if (p->gp_index_bits)
  171. parms->gp_index[i] = get_bits(pgb, p->gp_index_bits);
  172. for (j = 0; j < p->number_of_fc_indexes; j++)
  173. parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
  174. parms->gc_index[i] = get_bits(pgb, p->gc_index_bits);
  175. }
  176. }
  177. static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
  178. int num_subfr)
  179. {
  180. double lsfint[LP_FILTER_ORDER];
  181. int i,j;
  182. float t, t0 = 1.0 / num_subfr;
  183. t = t0 * 0.5;
  184. for (i = 0; i < num_subfr; i++) {
  185. for (j = 0; j < LP_FILTER_ORDER; j++)
  186. lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
  187. ff_amrwb_lsp2lpc(lsfint, Az, LP_FILTER_ORDER);
  188. Az += LP_FILTER_ORDER;
  189. t += t0;
  190. }
  191. }
  192. /**
  193. * Evaluate the adaptive impulse response.
  194. */
  195. static void eval_ir(const float *Az, int pitch_lag, float *freq,
  196. float pitch_sharp_factor)
  197. {
  198. float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
  199. int i;
  200. tmp1[0] = 1.;
  201. for (i = 0; i < LP_FILTER_ORDER; i++) {
  202. tmp1[i+1] = Az[i] * ff_pow_0_55[i];
  203. tmp2[i ] = Az[i] * ff_pow_0_7 [i];
  204. }
  205. memset(tmp1 + 11, 0, 37 * sizeof(float));
  206. ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
  207. LP_FILTER_ORDER);
  208. pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
  209. }
  210. /**
  211. * Evaluate the convolution of a vector with a sparse vector.
  212. */
  213. static void convolute_with_sparse(float *out, const AMRFixed *pulses,
  214. const float *shape, int length)
  215. {
  216. int i, j;
  217. memset(out, 0, length*sizeof(float));
  218. for (i = 0; i < pulses->n; i++)
  219. for (j = pulses->x[i]; j < length; j++)
  220. out[j] += pulses->y[i] * shape[j - pulses->x[i]];
  221. }
  222. /**
  223. * Apply postfilter, very similar to AMR one.
  224. */
  225. static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
  226. {
  227. float buf[SUBFR_SIZE + LP_FILTER_ORDER];
  228. float *pole_out = buf + LP_FILTER_ORDER;
  229. float lpc_n[LP_FILTER_ORDER];
  230. float lpc_d[LP_FILTER_ORDER];
  231. int i;
  232. for (i = 0; i < LP_FILTER_ORDER; i++) {
  233. lpc_d[i] = lpc[i] * ff_pow_0_75[i];
  234. lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
  235. };
  236. memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
  237. LP_FILTER_ORDER*sizeof(float));
  238. ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
  239. LP_FILTER_ORDER);
  240. memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
  241. LP_FILTER_ORDER*sizeof(float));
  242. ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
  243. memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
  244. LP_FILTER_ORDER*sizeof(*pole_out));
  245. memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
  246. LP_FILTER_ORDER*sizeof(*pole_out));
  247. ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
  248. LP_FILTER_ORDER);
  249. }
  250. static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
  251. SiprMode mode, int low_gain)
  252. {
  253. int i;
  254. switch (mode) {
  255. case MODE_6k5:
  256. for (i = 0; i < 3; i++) {
  257. fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
  258. fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
  259. }
  260. fixed_sparse->n = 3;
  261. break;
  262. case MODE_8k5:
  263. for (i = 0; i < 3; i++) {
  264. fixed_sparse->x[2*i ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
  265. fixed_sparse->x[2*i + 1] = 3 * ( pulses[i] & 0xf) + i;
  266. fixed_sparse->y[2*i ] = (pulses[i] & 0x100) ? -1.0: 1.0;
  267. fixed_sparse->y[2*i + 1] =
  268. (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
  269. -fixed_sparse->y[2*i ] : fixed_sparse->y[2*i];
  270. }
  271. fixed_sparse->n = 6;
  272. break;
  273. case MODE_5k0:
  274. default:
  275. if (low_gain) {
  276. int offset = (pulses[0] & 0x200) ? 2 : 0;
  277. int val = pulses[0];
  278. for (i = 0; i < 3; i++) {
  279. int index = (val & 0x7) * 6 + 4 - i*2;
  280. fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
  281. fixed_sparse->x[i] = index;
  282. val >>= 3;
  283. }
  284. fixed_sparse->n = 3;
  285. } else {
  286. int pulse_subset = (pulses[0] >> 8) & 1;
  287. fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
  288. fixed_sparse->x[1] = ( pulses[0] & 15) * 3 + pulse_subset + 1;
  289. fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
  290. fixed_sparse->y[1] = -fixed_sparse->y[0];
  291. fixed_sparse->n = 2;
  292. }
  293. break;
  294. }
  295. }
  296. static void decode_frame(SiprContext *ctx, SiprParameters *params,
  297. float *out_data)
  298. {
  299. int i, j;
  300. int subframe_count = modes[ctx->mode].subframe_count;
  301. int frame_size = subframe_count * SUBFR_SIZE;
  302. float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT];
  303. float *excitation;
  304. float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
  305. float lsf_new[LP_FILTER_ORDER];
  306. float *impulse_response = ir_buf + LP_FILTER_ORDER;
  307. float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
  308. // memory alignment
  309. int t0_first = 0;
  310. AMRFixed fixed_cb;
  311. memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
  312. lsf_decode_fp(lsf_new, ctx->lsf_history, params);
  313. sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
  314. memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
  315. excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
  316. for (i = 0; i < subframe_count; i++) {
  317. float *pAz = Az + i*LP_FILTER_ORDER;
  318. float fixed_vector[SUBFR_SIZE];
  319. int T0,T0_frac;
  320. float pitch_gain, gain_code, avg_energy;
  321. ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
  322. ctx->mode == MODE_5k0, 6);
  323. if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
  324. t0_first = T0;
  325. ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
  326. ff_b60_sinc, 6,
  327. 2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
  328. SUBFR_SIZE);
  329. decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
  330. ctx->past_pitch_gain < 0.8);
  331. eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
  332. convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
  333. SUBFR_SIZE);
  334. avg_energy =
  335. (0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/
  336. SUBFR_SIZE;
  337. ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
  338. gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
  339. avg_energy, ctx->energy_history,
  340. 34 - 15.0/(0.05*M_LN10/M_LN2),
  341. pred);
  342. ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
  343. pitch_gain, gain_code, SUBFR_SIZE);
  344. pitch_gain *= 0.5 * pitch_gain;
  345. pitch_gain = FFMIN(pitch_gain, 0.4);
  346. ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
  347. ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
  348. gain_code *= ctx->gain_mem;
  349. for (j = 0; j < SUBFR_SIZE; j++)
  350. fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
  351. if (ctx->mode == MODE_5k0) {
  352. postfilter_5k0(ctx, pAz, fixed_vector);
  353. ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
  354. pAz, excitation, SUBFR_SIZE,
  355. LP_FILTER_ORDER);
  356. }
  357. ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
  358. SUBFR_SIZE, LP_FILTER_ORDER);
  359. excitation += SUBFR_SIZE;
  360. }
  361. memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
  362. LP_FILTER_ORDER * sizeof(float));
  363. if (ctx->mode == MODE_5k0) {
  364. for (i = 0; i < subframe_count; i++) {
  365. float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
  366. ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
  367. SUBFR_SIZE);
  368. ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
  369. &synth[i * SUBFR_SIZE], energy,
  370. SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
  371. }
  372. memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
  373. LP_FILTER_ORDER*sizeof(float));
  374. }
  375. memmove(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
  376. (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
  377. ff_acelp_apply_order_2_transfer_function(out_data, synth,
  378. (const float[2]) {-1.99997 , 1.000000000},
  379. (const float[2]) {-1.93307352, 0.935891986},
  380. 0.939805806,
  381. ctx->highpass_filt_mem,
  382. frame_size);
  383. }
  384. static av_cold int sipr_decoder_init(AVCodecContext * avctx)
  385. {
  386. SiprContext *ctx = avctx->priv_data;
  387. int i;
  388. switch (avctx->block_align) {
  389. case 20: ctx->mode = MODE_16k; break;
  390. case 19: ctx->mode = MODE_8k5; break;
  391. case 29: ctx->mode = MODE_6k5; break;
  392. case 37: ctx->mode = MODE_5k0; break;
  393. default:
  394. av_log(avctx, AV_LOG_ERROR, "Invalid block_align: %d\n", avctx->block_align);
  395. if (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
  396. else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
  397. else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
  398. else ctx->mode = MODE_5k0;
  399. }
  400. av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
  401. if (ctx->mode == MODE_16k) {
  402. ff_sipr_init_16k(ctx);
  403. ctx->decode_frame = ff_sipr_decode_frame_16k;
  404. } else {
  405. ctx->decode_frame = decode_frame;
  406. }
  407. for (i = 0; i < LP_FILTER_ORDER; i++)
  408. ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
  409. for (i = 0; i < 4; i++)
  410. ctx->energy_history[i] = -14;
  411. avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
  412. avcodec_get_frame_defaults(&ctx->frame);
  413. avctx->coded_frame = &ctx->frame;
  414. return 0;
  415. }
  416. static int sipr_decode_frame(AVCodecContext *avctx, void *data,
  417. int *got_frame_ptr, AVPacket *avpkt)
  418. {
  419. SiprContext *ctx = avctx->priv_data;
  420. const uint8_t *buf=avpkt->data;
  421. SiprParameters parm;
  422. const SiprModeParam *mode_par = &modes[ctx->mode];
  423. GetBitContext gb;
  424. float *samples;
  425. int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
  426. int i, ret;
  427. ctx->avctx = avctx;
  428. if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
  429. av_log(avctx, AV_LOG_ERROR,
  430. "Error processing packet: packet size (%d) too small\n",
  431. avpkt->size);
  432. return -1;
  433. }
  434. /* get output buffer */
  435. ctx->frame.nb_samples = mode_par->frames_per_packet * subframe_size *
  436. mode_par->subframe_count;
  437. if ((ret = avctx->get_buffer(avctx, &ctx->frame)) < 0) {
  438. av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
  439. return ret;
  440. }
  441. samples = (float *)ctx->frame.data[0];
  442. init_get_bits(&gb, buf, mode_par->bits_per_frame);
  443. for (i = 0; i < mode_par->frames_per_packet; i++) {
  444. decode_parameters(&parm, &gb, mode_par);
  445. ctx->decode_frame(ctx, &parm, samples);
  446. samples += subframe_size * mode_par->subframe_count;
  447. }
  448. *got_frame_ptr = 1;
  449. *(AVFrame *)data = ctx->frame;
  450. return mode_par->bits_per_frame >> 3;
  451. }
  452. AVCodec ff_sipr_decoder = {
  453. .name = "sipr",
  454. .type = AVMEDIA_TYPE_AUDIO,
  455. .id = AV_CODEC_ID_SIPR,
  456. .priv_data_size = sizeof(SiprContext),
  457. .init = sipr_decoder_init,
  458. .decode = sipr_decode_frame,
  459. .capabilities = CODEC_CAP_DR1,
  460. .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
  461. };