1/* 2 * Windows Media Audio Voice decoder. 3 * Copyright (c) 2009 Ronald S. Bultje 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/** 23 * @file 24 * @brief Windows Media Audio Voice compatible decoder 25 * @author Ronald S. Bultje <rsbultje@gmail.com> 26 */ 27 28#include <math.h> 29 30#include "libavutil/channel_layout.h" 31#include "libavutil/float_dsp.h" 32#include "libavutil/mem_internal.h" 33#include "libavutil/thread.h" 34#include "avcodec.h" 35#include "codec_internal.h" 36#include "internal.h" 37#include "get_bits.h" 38#include "put_bits.h" 39#include "wmavoice_data.h" 40#include "celp_filters.h" 41#include "acelp_vectors.h" 42#include "acelp_filters.h" 43#include "lsp.h" 44#include "dct.h" 45#include "rdft.h" 46#include "sinewin.h" 47 48#define MAX_BLOCKS 8 ///< maximum number of blocks per frame 49#define MAX_LSPS 16 ///< maximum filter order 50#define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple 51 ///< of 16 for ASM input buffer alignment 52#define MAX_FRAMES 3 ///< maximum number of frames per superframe 53#define MAX_FRAMESIZE 160 ///< maximum number of samples per frame 54#define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history 55#define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) 56 ///< maximum number of samples per superframe 57#define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that 58 ///< was split over two packets 59#define VLC_NBITS 6 ///< number of bits to read per VLC iteration 60 61/** 62 * Frame type VLC coding. 63 */ 64static VLC frame_type_vlc; 65 66/** 67 * Adaptive codebook types. 68 */ 69enum { 70 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) 71 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which 72 ///< we interpolate to get a per-sample pitch. 73 ///< Signal is generated using an asymmetric sinc 74 ///< window function 75 ///< @note see #wmavoice_ipol1_coeffs 76 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using 77 ///< a Hamming sinc window function 78 ///< @note see #wmavoice_ipol2_coeffs 79}; 80 81/** 82 * Fixed codebook types. 83 */ 84enum { 85 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence 86 ///< generated from a hardcoded (fixed) codebook 87 ///< with per-frame (low) gain values 88 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block 89 ///< gain values 90 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, 91 ///< used in particular for low-bitrate streams 92 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in 93 ///< combinations of either single pulses or 94 ///< pulse pairs 95}; 96 97/** 98 * Description of frame types. 99 */ 100static const struct frame_type_desc { 101 uint8_t n_blocks; ///< amount of blocks per frame (each block 102 ///< (contains 160/#n_blocks samples) 103 uint8_t log_n_blocks; ///< log2(#n_blocks) 104 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) 105 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) 106 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs 107 ///< (rather than just one single pulse) 108 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES 109} frame_descs[17] = { 110 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0 }, 111 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0 }, 112 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0 }, 113 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, 114 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, 115 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0 }, 116 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, 117 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, 118 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, 119 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, 120 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, 121 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, 122 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, 123 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, 124 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, 125 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, 126 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 } 127}; 128 129/** 130 * WMA Voice decoding context. 131 */ 132typedef struct WMAVoiceContext { 133 /** 134 * @name Global values specified in the stream header / extradata or used all over. 135 * @{ 136 */ 137 GetBitContext gb; ///< packet bitreader. During decoder init, 138 ///< it contains the extradata from the 139 ///< demuxer. During decoding, it contains 140 ///< packet data. 141 int8_t vbm_tree[25]; ///< converts VLC codes to frame type 142 143 int spillover_bitsize; ///< number of bits used to specify 144 ///< #spillover_nbits in the packet header 145 ///< = ceil(log2(ctx->block_align << 3)) 146 int history_nsamples; ///< number of samples in history for signal 147 ///< prediction (through ACB) 148 149 /* postfilter specific values */ 150 int do_apf; ///< whether to apply the averaged 151 ///< projection filter (APF) 152 int denoise_strength; ///< strength of denoising in Wiener filter 153 ///< [0-11] 154 int denoise_tilt_corr; ///< Whether to apply tilt correction to the 155 ///< Wiener filter coefficients (postfilter) 156 int dc_level; ///< Predicted amount of DC noise, based 157 ///< on which a DC removal filter is used 158 159 int lsps; ///< number of LSPs per frame [10 or 16] 160 int lsp_q_mode; ///< defines quantizer defaults [0, 1] 161 int lsp_def_mode; ///< defines different sets of LSP defaults 162 ///< [0, 1] 163 164 int min_pitch_val; ///< base value for pitch parsing code 165 int max_pitch_val; ///< max value + 1 for pitch parsing 166 int pitch_nbits; ///< number of bits used to specify the 167 ///< pitch value in the frame header 168 int block_pitch_nbits; ///< number of bits used to specify the 169 ///< first block's pitch value 170 int block_pitch_range; ///< range of the block pitch 171 int block_delta_pitch_nbits; ///< number of bits used to specify the 172 ///< delta pitch between this and the last 173 ///< block's pitch value, used in all but 174 ///< first block 175 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is 176 ///< from -this to +this-1) 177 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale 178 ///< conversion 179 180 /** 181 * @} 182 * 183 * @name Packet values specified in the packet header or related to a packet. 184 * 185 * A packet is considered to be a single unit of data provided to this 186 * decoder by the demuxer. 187 * @{ 188 */ 189 int spillover_nbits; ///< number of bits of the previous packet's 190 ///< last superframe preceding this 191 ///< packet's first full superframe (useful 192 ///< for re-synchronization also) 193 int has_residual_lsps; ///< if set, superframes contain one set of 194 ///< LSPs that cover all frames, encoded as 195 ///< independent and residual LSPs; if not 196 ///< set, each frame contains its own, fully 197 ///< independent, LSPs 198 int skip_bits_next; ///< number of bits to skip at the next call 199 ///< to #wmavoice_decode_packet() (since 200 ///< they're part of the previous superframe) 201 202 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + AV_INPUT_BUFFER_PADDING_SIZE]; 203 ///< cache for superframe data split over 204 ///< multiple packets 205 int sframe_cache_size; ///< set to >0 if we have data from an 206 ///< (incomplete) superframe from a previous 207 ///< packet that spilled over in the current 208 ///< packet; specifies the amount of bits in 209 ///< #sframe_cache 210 PutBitContext pb; ///< bitstream writer for #sframe_cache 211 212 /** 213 * @} 214 * 215 * @name Frame and superframe values 216 * Superframe and frame data - these can change from frame to frame, 217 * although some of them do in that case serve as a cache / history for 218 * the next frame or superframe. 219 * @{ 220 */ 221 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous 222 ///< superframe 223 int last_pitch_val; ///< pitch value of the previous frame 224 int last_acb_type; ///< frame type [0-2] of the previous frame 225 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) 226 ///< << 16) / #MAX_FRAMESIZE 227 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE 228 229 int aw_idx_is_ext; ///< whether the AW index was encoded in 230 ///< 8 bits (instead of 6) 231 int aw_pulse_range; ///< the range over which #aw_pulse_set1() 232 ///< can apply the pulse, relative to the 233 ///< value in aw_first_pulse_off. The exact 234 ///< position of the first AW-pulse is within 235 ///< [pulse_off, pulse_off + this], and 236 ///< depends on bitstream values; [16 or 24] 237 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note 238 ///< that this number can be negative (in 239 ///< which case it basically means "zero") 240 int aw_first_pulse_off[2]; ///< index of first sample to which to 241 ///< apply AW-pulses, or -0xff if unset 242 int aw_next_pulse_off_cache; ///< the position (relative to start of the 243 ///< second block) at which pulses should 244 ///< start to be positioned, serves as a 245 ///< cache for pitch-adaptive window pulses 246 ///< between blocks 247 248 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is 249 ///< only used for comfort noise in #pRNG() 250 int nb_superframes; ///< number of superframes in current packet 251 float gain_pred_err[6]; ///< cache for gain prediction 252 float excitation_history[MAX_SIGNAL_HISTORY]; 253 ///< cache of the signal of previous 254 ///< superframes, used as a history for 255 ///< signal generation 256 float synth_history[MAX_LSPS]; ///< see #excitation_history 257 /** 258 * @} 259 * 260 * @name Postfilter values 261 * 262 * Variables used for postfilter implementation, mostly history for 263 * smoothing and so on, and context variables for FFT/iFFT. 264 * @{ 265 */ 266 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the 267 ///< postfilter (for denoise filter) 268 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert 269 ///< transform, part of postfilter) 270 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] 271 ///< range 272 float postfilter_agc; ///< gain control memory, used in 273 ///< #adaptive_gain_control() 274 float dcf_mem[2]; ///< DC filter history 275 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; 276 ///< zero filter output (i.e. excitation) 277 ///< by postfilter 278 float denoise_filter_cache[MAX_FRAMESIZE]; 279 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache 280 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80]; 281 ///< aligned buffer for LPC tilting 282 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; 283 ///< aligned buffer for denoise coefficients 284 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; 285 ///< aligned buffer for postfilter speech 286 ///< synthesis 287 /** 288 * @} 289 */ 290} WMAVoiceContext; 291 292/** 293 * Set up the variable bit mode (VBM) tree from container extradata. 294 * @param gb bit I/O context. 295 * The bit context (s->gb) should be loaded with byte 23-46 of the 296 * container extradata (i.e. the ones containing the VBM tree). 297 * @param vbm_tree pointer to array to which the decoded VBM tree will be 298 * written. 299 * @return 0 on success, <0 on error. 300 */ 301static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) 302{ 303 int cntr[8] = { 0 }, n, res; 304 305 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); 306 for (n = 0; n < 17; n++) { 307 res = get_bits(gb, 3); 308 if (cntr[res] > 3) // should be >= 3 + (res == 7)) 309 return -1; 310 vbm_tree[res * 3 + cntr[res]++] = n; 311 } 312 return 0; 313} 314 315static av_cold void wmavoice_init_static_data(void) 316{ 317 static const uint8_t bits[] = { 318 2, 2, 2, 4, 4, 4, 319 6, 6, 6, 8, 8, 8, 320 10, 10, 10, 12, 12, 12, 321 14, 14, 14, 14 322 }; 323 static const uint16_t codes[] = { 324 0x0000, 0x0001, 0x0002, // 00/01/10 325 0x000c, 0x000d, 0x000e, // 11+00/01/10 326 0x003c, 0x003d, 0x003e, // 1111+00/01/10 327 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 328 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 329 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 330 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx 331 }; 332 333 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), 334 bits, 1, 1, codes, 2, 2, 132); 335} 336 337static av_cold void wmavoice_flush(AVCodecContext *ctx) 338{ 339 WMAVoiceContext *s = ctx->priv_data; 340 int n; 341 342 s->postfilter_agc = 0; 343 s->sframe_cache_size = 0; 344 s->skip_bits_next = 0; 345 for (n = 0; n < s->lsps; n++) 346 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); 347 memset(s->excitation_history, 0, 348 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); 349 memset(s->synth_history, 0, 350 sizeof(*s->synth_history) * MAX_LSPS); 351 memset(s->gain_pred_err, 0, 352 sizeof(s->gain_pred_err)); 353 354 if (s->do_apf) { 355 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, 356 sizeof(*s->synth_filter_out_buf) * s->lsps); 357 memset(s->dcf_mem, 0, 358 sizeof(*s->dcf_mem) * 2); 359 memset(s->zero_exc_pf, 0, 360 sizeof(*s->zero_exc_pf) * s->history_nsamples); 361 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); 362 } 363} 364 365/** 366 * Set up decoder with parameters from demuxer (extradata etc.). 367 */ 368static av_cold int wmavoice_decode_init(AVCodecContext *ctx) 369{ 370 static AVOnce init_static_once = AV_ONCE_INIT; 371 int n, flags, pitch_range, lsp16_flag, ret; 372 WMAVoiceContext *s = ctx->priv_data; 373 374 ff_thread_once(&init_static_once, wmavoice_init_static_data); 375 376 /** 377 * Extradata layout: 378 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), 379 * - byte 19-22: flags field (annoyingly in LE; see below for known 380 * values), 381 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, 382 * rest is 0). 383 */ 384 if (ctx->extradata_size != 46) { 385 av_log(ctx, AV_LOG_ERROR, 386 "Invalid extradata size %d (should be 46)\n", 387 ctx->extradata_size); 388 return AVERROR_INVALIDDATA; 389 } 390 if (ctx->block_align <= 0 || ctx->block_align > (1<<22)) { 391 av_log(ctx, AV_LOG_ERROR, "Invalid block alignment %d.\n", ctx->block_align); 392 return AVERROR_INVALIDDATA; 393 } 394 395 flags = AV_RL32(ctx->extradata + 18); 396 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); 397 s->do_apf = flags & 0x1; 398 if (s->do_apf) { 399 if ((ret = ff_rdft_init(&s->rdft, 7, DFT_R2C)) < 0 || 400 (ret = ff_rdft_init(&s->irdft, 7, IDFT_C2R)) < 0 || 401 (ret = ff_dct_init (&s->dct, 6, DCT_I)) < 0 || 402 (ret = ff_dct_init (&s->dst, 6, DST_I)) < 0) 403 return ret; 404 405 ff_sine_window_init(s->cos, 256); 406 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); 407 for (n = 0; n < 255; n++) { 408 s->sin[n] = -s->sin[510 - n]; 409 s->cos[510 - n] = s->cos[n]; 410 } 411 } 412 s->denoise_strength = (flags >> 2) & 0xF; 413 if (s->denoise_strength >= 12) { 414 av_log(ctx, AV_LOG_ERROR, 415 "Invalid denoise filter strength %d (max=11)\n", 416 s->denoise_strength); 417 return AVERROR_INVALIDDATA; 418 } 419 s->denoise_tilt_corr = !!(flags & 0x40); 420 s->dc_level = (flags >> 7) & 0xF; 421 s->lsp_q_mode = !!(flags & 0x2000); 422 s->lsp_def_mode = !!(flags & 0x4000); 423 lsp16_flag = flags & 0x1000; 424 if (lsp16_flag) { 425 s->lsps = 16; 426 } else { 427 s->lsps = 10; 428 } 429 for (n = 0; n < s->lsps; n++) 430 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); 431 432 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); 433 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { 434 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); 435 return AVERROR_INVALIDDATA; 436 } 437 438 if (ctx->sample_rate >= INT_MAX / (256 * 37)) 439 return AVERROR_INVALIDDATA; 440 441 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; 442 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; 443 pitch_range = s->max_pitch_val - s->min_pitch_val; 444 if (pitch_range <= 0) { 445 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n"); 446 return AVERROR_INVALIDDATA; 447 } 448 s->pitch_nbits = av_ceil_log2(pitch_range); 449 s->last_pitch_val = 40; 450 s->last_acb_type = ACB_TYPE_NONE; 451 s->history_nsamples = s->max_pitch_val + 8; 452 453 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { 454 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, 455 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; 456 457 av_log(ctx, AV_LOG_ERROR, 458 "Unsupported samplerate %d (min=%d, max=%d)\n", 459 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz 460 461 return AVERROR(ENOSYS); 462 } 463 464 s->block_conv_table[0] = s->min_pitch_val; 465 s->block_conv_table[1] = (pitch_range * 25) >> 6; 466 s->block_conv_table[2] = (pitch_range * 44) >> 6; 467 s->block_conv_table[3] = s->max_pitch_val - 1; 468 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; 469 if (s->block_delta_pitch_hrange <= 0) { 470 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n"); 471 return AVERROR_INVALIDDATA; 472 } 473 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); 474 s->block_pitch_range = s->block_conv_table[2] + 475 s->block_conv_table[3] + 1 + 476 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); 477 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); 478 479 av_channel_layout_uninit(&ctx->ch_layout); 480 ctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO; 481 ctx->sample_fmt = AV_SAMPLE_FMT_FLT; 482 483 return 0; 484} 485 486/** 487 * @name Postfilter functions 488 * Postfilter functions (gain control, wiener denoise filter, DC filter, 489 * kalman smoothening, plus surrounding code to wrap it) 490 * @{ 491 */ 492/** 493 * Adaptive gain control (as used in postfilter). 494 * 495 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except 496 * that the energy here is calculated using sum(abs(...)), whereas the 497 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). 498 * 499 * @param out output buffer for filtered samples 500 * @param in input buffer containing the samples as they are after the 501 * postfilter steps so far 502 * @param speech_synth input buffer containing speech synth before postfilter 503 * @param size input buffer size 504 * @param alpha exponential filter factor 505 * @param gain_mem pointer to filter memory (single float) 506 */ 507static void adaptive_gain_control(float *out, const float *in, 508 const float *speech_synth, 509 int size, float alpha, float *gain_mem) 510{ 511 int i; 512 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; 513 float mem = *gain_mem; 514 515 for (i = 0; i < size; i++) { 516 speech_energy += fabsf(speech_synth[i]); 517 postfilter_energy += fabsf(in[i]); 518 } 519 gain_scale_factor = postfilter_energy == 0.0 ? 0.0 : 520 (1.0 - alpha) * speech_energy / postfilter_energy; 521 522 for (i = 0; i < size; i++) { 523 mem = alpha * mem + gain_scale_factor; 524 out[i] = in[i] * mem; 525 } 526 527 *gain_mem = mem; 528} 529 530/** 531 * Kalman smoothing function. 532 * 533 * This function looks back pitch +/- 3 samples back into history to find 534 * the best fitting curve (that one giving the optimal gain of the two 535 * signals, i.e. the highest dot product between the two), and then 536 * uses that signal history to smoothen the output of the speech synthesis 537 * filter. 538 * 539 * @param s WMA Voice decoding context 540 * @param pitch pitch of the speech signal 541 * @param in input speech signal 542 * @param out output pointer for smoothened signal 543 * @param size input/output buffer size 544 * 545 * @returns -1 if no smoothening took place, e.g. because no optimal 546 * fit could be found, or 0 on success. 547 */ 548static int kalman_smoothen(WMAVoiceContext *s, int pitch, 549 const float *in, float *out, int size) 550{ 551 int n; 552 float optimal_gain = 0, dot; 553 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], 554 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], 555 *best_hist_ptr = NULL; 556 557 /* find best fitting point in history */ 558 do { 559 dot = avpriv_scalarproduct_float_c(in, ptr, size); 560 if (dot > optimal_gain) { 561 optimal_gain = dot; 562 best_hist_ptr = ptr; 563 } 564 } while (--ptr >= end); 565 566 if (optimal_gain <= 0) 567 return -1; 568 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size); 569 if (dot <= 0) // would be 1.0 570 return -1; 571 572 if (optimal_gain <= dot) { 573 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 574 } else 575 dot = 0.625; 576 577 /* actual smoothing */ 578 for (n = 0; n < size; n++) 579 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); 580 581 return 0; 582} 583 584/** 585 * Get the tilt factor of a formant filter from its transfer function 586 * @see #tilt_factor() in amrnbdec.c, which does essentially the same, 587 * but somehow (??) it does a speech synthesis filter in the 588 * middle, which is missing here 589 * 590 * @param lpcs LPC coefficients 591 * @param n_lpcs Size of LPC buffer 592 * @returns the tilt factor 593 */ 594static float tilt_factor(const float *lpcs, int n_lpcs) 595{ 596 float rh0, rh1; 597 598 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs); 599 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1); 600 601 return rh1 / rh0; 602} 603 604/** 605 * Derive denoise filter coefficients (in real domain) from the LPCs. 606 */ 607static void calc_input_response(WMAVoiceContext *s, float *lpcs, 608 int fcb_type, float *coeffs, int remainder) 609{ 610 float last_coeff, min = 15.0, max = -15.0; 611 float irange, angle_mul, gain_mul, range, sq; 612 int n, idx; 613 614 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ 615 s->rdft.rdft_calc(&s->rdft, lpcs); 616#define log_range(var, assign) do { \ 617 float tmp = log10f(assign); var = tmp; \ 618 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ 619 } while (0) 620 log_range(last_coeff, lpcs[1] * lpcs[1]); 621 for (n = 1; n < 64; n++) 622 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + 623 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); 624 log_range(lpcs[0], lpcs[0] * lpcs[0]); 625#undef log_range 626 range = max - min; 627 lpcs[64] = last_coeff; 628 629 /* Now, use this spectrum to pick out these frequencies with higher 630 * (relative) power/energy (which we then take to be "not noise"), 631 * and set up a table (still in lpc[]) of (relative) gains per frequency. 632 * These frequencies will be maintained, while others ("noise") will be 633 * decreased in the filter output. */ 634 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] 635 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : 636 (5.0 / 14.7)); 637 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); 638 for (n = 0; n <= 64; n++) { 639 float pwr; 640 641 idx = lrint((max - lpcs[n]) * irange - 1); 642 idx = FFMAX(0, idx); 643 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx]; 644 lpcs[n] = angle_mul * pwr; 645 646 /* 70.57 =~ 1/log10(1.0331663) */ 647 idx = av_clipf((pwr * gain_mul - 0.0295) * 70.570526123, 0, INT_MAX / 2); 648 649 if (idx > 127) { // fall back if index falls outside table range 650 coeffs[n] = wmavoice_energy_table[127] * 651 powf(1.0331663, idx - 127); 652 } else 653 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; 654 } 655 656 /* calculate the Hilbert transform of the gains, which we do (since this 657 * is a sine input) by doing a phase shift (in theory, H(sin())=cos()). 658 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the 659 * "moment" of the LPCs in this filter. */ 660 s->dct.dct_calc(&s->dct, lpcs); 661 s->dst.dct_calc(&s->dst, lpcs); 662 663 /* Split out the coefficient indexes into phase/magnitude pairs */ 664 idx = 255 + av_clip(lpcs[64], -255, 255); 665 coeffs[0] = coeffs[0] * s->cos[idx]; 666 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); 667 last_coeff = coeffs[64] * s->cos[idx]; 668 for (n = 63;; n--) { 669 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); 670 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; 671 coeffs[n * 2] = coeffs[n] * s->cos[idx]; 672 673 if (!--n) break; 674 675 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); 676 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; 677 coeffs[n * 2] = coeffs[n] * s->cos[idx]; 678 } 679 coeffs[1] = last_coeff; 680 681 /* move into real domain */ 682 s->irdft.rdft_calc(&s->irdft, coeffs); 683 684 /* tilt correction and normalize scale */ 685 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); 686 if (s->denoise_tilt_corr) { 687 float tilt_mem = 0; 688 689 coeffs[remainder - 1] = 0; 690 ff_tilt_compensation(&tilt_mem, 691 -1.8 * tilt_factor(coeffs, remainder - 1), 692 coeffs, remainder); 693 } 694 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs, 695 remainder)); 696 for (n = 0; n < remainder; n++) 697 coeffs[n] *= sq; 698} 699 700/** 701 * This function applies a Wiener filter on the (noisy) speech signal as 702 * a means to denoise it. 703 * 704 * - take RDFT of LPCs to get the power spectrum of the noise + speech; 705 * - using this power spectrum, calculate (for each frequency) the Wiener 706 * filter gain, which depends on the frequency power and desired level 707 * of noise subtraction (when set too high, this leads to artifacts) 708 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse 709 * of 4-8kHz); 710 * - by doing a phase shift, calculate the Hilbert transform of this array 711 * of per-frequency filter-gains to get the filtering coefficients; 712 * - smoothen/normalize/de-tilt these filter coefficients as desired; 713 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT 714 * to get the denoised speech signal; 715 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond 716 * the frame boundary) are saved and applied to subsequent frames by an 717 * overlap-add method (otherwise you get clicking-artifacts). 718 * 719 * @param s WMA Voice decoding context 720 * @param fcb_type Frame (codebook) type 721 * @param synth_pf input: the noisy speech signal, output: denoised speech 722 * data; should be 16-byte aligned (for ASM purposes) 723 * @param size size of the speech data 724 * @param lpcs LPCs used to synthesize this frame's speech data 725 */ 726static void wiener_denoise(WMAVoiceContext *s, int fcb_type, 727 float *synth_pf, int size, 728 const float *lpcs) 729{ 730 int remainder, lim, n; 731 732 if (fcb_type != FCB_TYPE_SILENCE) { 733 float *tilted_lpcs = s->tilted_lpcs_pf, 734 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; 735 736 tilted_lpcs[0] = 1.0; 737 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); 738 memset(&tilted_lpcs[s->lsps + 1], 0, 739 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); 740 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), 741 tilted_lpcs, s->lsps + 2); 742 743 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame 744 * size is applied to the next frame. All input beyond this is zero, 745 * and thus all output beyond this will go towards zero, hence we can 746 * limit to min(size-1, 127-size) as a performance consideration. */ 747 remainder = FFMIN(127 - size, size - 1); 748 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); 749 750 /* apply coefficients (in frequency spectrum domain), i.e. complex 751 * number multiplication */ 752 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); 753 s->rdft.rdft_calc(&s->rdft, synth_pf); 754 s->rdft.rdft_calc(&s->rdft, coeffs); 755 synth_pf[0] *= coeffs[0]; 756 synth_pf[1] *= coeffs[1]; 757 for (n = 1; n < 64; n++) { 758 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; 759 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; 760 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; 761 } 762 s->irdft.rdft_calc(&s->irdft, synth_pf); 763 } 764 765 /* merge filter output with the history of previous runs */ 766 if (s->denoise_filter_cache_size) { 767 lim = FFMIN(s->denoise_filter_cache_size, size); 768 for (n = 0; n < lim; n++) 769 synth_pf[n] += s->denoise_filter_cache[n]; 770 s->denoise_filter_cache_size -= lim; 771 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], 772 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); 773 } 774 775 /* move remainder of filter output into a cache for future runs */ 776 if (fcb_type != FCB_TYPE_SILENCE) { 777 lim = FFMIN(remainder, s->denoise_filter_cache_size); 778 for (n = 0; n < lim; n++) 779 s->denoise_filter_cache[n] += synth_pf[size + n]; 780 if (lim < remainder) { 781 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], 782 sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); 783 s->denoise_filter_cache_size = remainder; 784 } 785 } 786} 787 788/** 789 * Averaging projection filter, the postfilter used in WMAVoice. 790 * 791 * This uses the following steps: 792 * - A zero-synthesis filter (generate excitation from synth signal) 793 * - Kalman smoothing on excitation, based on pitch 794 * - Re-synthesized smoothened output 795 * - Iterative Wiener denoise filter 796 * - Adaptive gain filter 797 * - DC filter 798 * 799 * @param s WMAVoice decoding context 800 * @param synth Speech synthesis output (before postfilter) 801 * @param samples Output buffer for filtered samples 802 * @param size Buffer size of synth & samples 803 * @param lpcs Generated LPCs used for speech synthesis 804 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned) 805 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) 806 * @param pitch Pitch of the input signal 807 */ 808static void postfilter(WMAVoiceContext *s, const float *synth, 809 float *samples, int size, 810 const float *lpcs, float *zero_exc_pf, 811 int fcb_type, int pitch) 812{ 813 float synth_filter_in_buf[MAX_FRAMESIZE / 2], 814 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], 815 *synth_filter_in = zero_exc_pf; 816 817 av_assert0(size <= MAX_FRAMESIZE / 2); 818 819 /* generate excitation from input signal */ 820 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); 821 822 if (fcb_type >= FCB_TYPE_AW_PULSES && 823 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) 824 synth_filter_in = synth_filter_in_buf; 825 826 /* re-synthesize speech after smoothening, and keep history */ 827 ff_celp_lp_synthesis_filterf(synth_pf, lpcs, 828 synth_filter_in, size, s->lsps); 829 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], 830 sizeof(synth_pf[0]) * s->lsps); 831 832 wiener_denoise(s, fcb_type, synth_pf, size, lpcs); 833 834 adaptive_gain_control(samples, synth_pf, synth, size, 0.99, 835 &s->postfilter_agc); 836 837 if (s->dc_level > 8) { 838 /* remove ultra-low frequency DC noise / highpass filter; 839 * coefficients are identical to those used in SIPR decoding, 840 * and very closely resemble those used in AMR-NB decoding. */ 841 ff_acelp_apply_order_2_transfer_function(samples, samples, 842 (const float[2]) { -1.99997, 1.0 }, 843 (const float[2]) { -1.9330735188, 0.93589198496 }, 844 0.93980580475, s->dcf_mem, size); 845 } 846} 847/** 848 * @} 849 */ 850 851/** 852 * Dequantize LSPs 853 * @param lsps output pointer to the array that will hold the LSPs 854 * @param num number of LSPs to be dequantized 855 * @param values quantized values, contains n_stages values 856 * @param sizes range (i.e. max value) of each quantized value 857 * @param n_stages number of dequantization runs 858 * @param table dequantization table to be used 859 * @param mul_q LSF multiplier 860 * @param base_q base (lowest) LSF values 861 */ 862static void dequant_lsps(double *lsps, int num, 863 const uint16_t *values, 864 const uint16_t *sizes, 865 int n_stages, const uint8_t *table, 866 const double *mul_q, 867 const double *base_q) 868{ 869 int n, m; 870 871 memset(lsps, 0, num * sizeof(*lsps)); 872 for (n = 0; n < n_stages; n++) { 873 const uint8_t *t_off = &table[values[n] * num]; 874 double base = base_q[n], mul = mul_q[n]; 875 876 for (m = 0; m < num; m++) 877 lsps[m] += base + mul * t_off[m]; 878 879 table += sizes[n] * num; 880 } 881} 882 883/** 884 * @name LSP dequantization routines 885 * LSP dequantization routines, for 10/16LSPs and independent/residual coding. 886 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; 887 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. 888 * @{ 889 */ 890/** 891 * Parse 10 independently-coded LSPs. 892 */ 893static void dequant_lsp10i(GetBitContext *gb, double *lsps) 894{ 895 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; 896 static const double mul_lsf[4] = { 897 5.2187144800e-3, 1.4626986422e-3, 898 9.6179549166e-4, 1.1325736225e-3 899 }; 900 static const double base_lsf[4] = { 901 M_PI * -2.15522e-1, M_PI * -6.1646e-2, 902 M_PI * -3.3486e-2, M_PI * -5.7408e-2 903 }; 904 uint16_t v[4]; 905 906 v[0] = get_bits(gb, 8); 907 v[1] = get_bits(gb, 6); 908 v[2] = get_bits(gb, 5); 909 v[3] = get_bits(gb, 5); 910 911 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, 912 mul_lsf, base_lsf); 913} 914 915/** 916 * Parse 10 independently-coded LSPs, and then derive the tables to 917 * generate LSPs for the other frames from them (residual coding). 918 */ 919static void dequant_lsp10r(GetBitContext *gb, 920 double *i_lsps, const double *old, 921 double *a1, double *a2, int q_mode) 922{ 923 static const uint16_t vec_sizes[3] = { 128, 64, 64 }; 924 static const double mul_lsf[3] = { 925 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 926 }; 927 static const double base_lsf[3] = { 928 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 929 }; 930 const float (*ipol_tab)[2][10] = q_mode ? 931 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; 932 uint16_t interpol, v[3]; 933 int n; 934 935 dequant_lsp10i(gb, i_lsps); 936 937 interpol = get_bits(gb, 5); 938 v[0] = get_bits(gb, 7); 939 v[1] = get_bits(gb, 6); 940 v[2] = get_bits(gb, 6); 941 942 for (n = 0; n < 10; n++) { 943 double delta = old[n] - i_lsps[n]; 944 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; 945 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; 946 } 947 948 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, 949 mul_lsf, base_lsf); 950} 951 952/** 953 * Parse 16 independently-coded LSPs. 954 */ 955static void dequant_lsp16i(GetBitContext *gb, double *lsps) 956{ 957 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; 958 static const double mul_lsf[5] = { 959 3.3439586280e-3, 6.9908173703e-4, 960 3.3216608306e-3, 1.0334960326e-3, 961 3.1899104283e-3 962 }; 963 static const double base_lsf[5] = { 964 M_PI * -1.27576e-1, M_PI * -2.4292e-2, 965 M_PI * -1.28094e-1, M_PI * -3.2128e-2, 966 M_PI * -1.29816e-1 967 }; 968 uint16_t v[5]; 969 970 v[0] = get_bits(gb, 8); 971 v[1] = get_bits(gb, 6); 972 v[2] = get_bits(gb, 7); 973 v[3] = get_bits(gb, 6); 974 v[4] = get_bits(gb, 7); 975 976 dequant_lsps( lsps, 5, v, vec_sizes, 2, 977 wmavoice_dq_lsp16i1, mul_lsf, base_lsf); 978 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, 979 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); 980 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, 981 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); 982} 983 984/** 985 * Parse 16 independently-coded LSPs, and then derive the tables to 986 * generate LSPs for the other frames from them (residual coding). 987 */ 988static void dequant_lsp16r(GetBitContext *gb, 989 double *i_lsps, const double *old, 990 double *a1, double *a2, int q_mode) 991{ 992 static const uint16_t vec_sizes[3] = { 128, 128, 128 }; 993 static const double mul_lsf[3] = { 994 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 995 }; 996 static const double base_lsf[3] = { 997 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 998 }; 999 const float (*ipol_tab)[2][16] = q_mode ? 1000 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; 1001 uint16_t interpol, v[3]; 1002 int n; 1003 1004 dequant_lsp16i(gb, i_lsps); 1005 1006 interpol = get_bits(gb, 5); 1007 v[0] = get_bits(gb, 7); 1008 v[1] = get_bits(gb, 7); 1009 v[2] = get_bits(gb, 7); 1010 1011 for (n = 0; n < 16; n++) { 1012 double delta = old[n] - i_lsps[n]; 1013 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; 1014 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; 1015 } 1016 1017 dequant_lsps( a2, 10, v, vec_sizes, 1, 1018 wmavoice_dq_lsp16r1, mul_lsf, base_lsf); 1019 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, 1020 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); 1021 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, 1022 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); 1023} 1024 1025/** 1026 * @} 1027 * @name Pitch-adaptive window coding functions 1028 * The next few functions are for pitch-adaptive window coding. 1029 * @{ 1030 */ 1031/** 1032 * Parse the offset of the first pitch-adaptive window pulses, and 1033 * the distribution of pulses between the two blocks in this frame. 1034 * @param s WMA Voice decoding context private data 1035 * @param gb bit I/O context 1036 * @param pitch pitch for each block in this frame 1037 */ 1038static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, 1039 const int *pitch) 1040{ 1041 static const int16_t start_offset[94] = { 1042 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, 1043 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, 1044 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, 1045 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, 1046 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, 1047 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, 1048 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, 1049 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 1050 }; 1051 int bits, offset; 1052 1053 /* position of pulse */ 1054 s->aw_idx_is_ext = 0; 1055 if ((bits = get_bits(gb, 6)) >= 54) { 1056 s->aw_idx_is_ext = 1; 1057 bits += (bits - 54) * 3 + get_bits(gb, 2); 1058 } 1059 1060 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count 1061 * the distribution of the pulses in each block contained in this frame. */ 1062 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; 1063 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; 1064 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; 1065 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; 1066 offset += s->aw_n_pulses[0] * pitch[0]; 1067 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; 1068 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; 1069 1070 /* if continuing from a position before the block, reset position to 1071 * start of block (when corrected for the range over which it can be 1072 * spread in aw_pulse_set1()). */ 1073 if (start_offset[bits] < MAX_FRAMESIZE / 2) { 1074 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) 1075 s->aw_first_pulse_off[1] -= pitch[1]; 1076 if (start_offset[bits] < 0) 1077 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) 1078 s->aw_first_pulse_off[0] -= pitch[0]; 1079 } 1080} 1081 1082/** 1083 * Apply second set of pitch-adaptive window pulses. 1084 * @param s WMA Voice decoding context private data 1085 * @param gb bit I/O context 1086 * @param block_idx block index in frame [0, 1] 1087 * @param fcb structure containing fixed codebook vector info 1088 * @return -1 on error, 0 otherwise 1089 */ 1090static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, 1091 int block_idx, AMRFixed *fcb) 1092{ 1093 uint16_t use_mask_mem[9]; // only 5 are used, rest is padding 1094 uint16_t *use_mask = use_mask_mem + 2; 1095 /* in this function, idx is the index in the 80-bit (+ padding) use_mask 1096 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits 1097 * of idx are the position of the bit within a particular item in the 1098 * array (0 being the most significant bit, and 15 being the least 1099 * significant bit), and the remainder (>> 4) is the index in the 1100 * use_mask[]-array. This is faster and uses less memory than using a 1101 * 80-byte/80-int array. */ 1102 int pulse_off = s->aw_first_pulse_off[block_idx], 1103 pulse_start, n, idx, range, aidx, start_off = 0; 1104 1105 /* set offset of first pulse to within this block */ 1106 if (s->aw_n_pulses[block_idx] > 0) 1107 while (pulse_off + s->aw_pulse_range < 1) 1108 pulse_off += fcb->pitch_lag; 1109 1110 /* find range per pulse */ 1111 if (s->aw_n_pulses[0] > 0) { 1112 if (block_idx == 0) { 1113 range = 32; 1114 } else /* block_idx = 1 */ { 1115 range = 8; 1116 if (s->aw_n_pulses[block_idx] > 0) 1117 pulse_off = s->aw_next_pulse_off_cache; 1118 } 1119 } else 1120 range = 16; 1121 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; 1122 1123 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, 1124 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus 1125 * we exclude that range from being pulsed again in this function. */ 1126 memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0])); 1127 memset( use_mask, -1, 5 * sizeof(use_mask[0])); 1128 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); 1129 if (s->aw_n_pulses[block_idx] > 0) 1130 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { 1131 int excl_range = s->aw_pulse_range; // always 16 or 24 1132 uint16_t *use_mask_ptr = &use_mask[idx >> 4]; 1133 int first_sh = 16 - (idx & 15); 1134 *use_mask_ptr++ &= 0xFFFFu << first_sh; 1135 excl_range -= first_sh; 1136 if (excl_range >= 16) { 1137 *use_mask_ptr++ = 0; 1138 *use_mask_ptr &= 0xFFFF >> (excl_range - 16); 1139 } else 1140 *use_mask_ptr &= 0xFFFF >> excl_range; 1141 } 1142 1143 /* find the 'aidx'th offset that is not excluded */ 1144 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); 1145 for (n = 0; n <= aidx; pulse_start++) { 1146 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; 1147 if (idx >= MAX_FRAMESIZE / 2) { // find from zero 1148 if (use_mask[0]) idx = 0x0F; 1149 else if (use_mask[1]) idx = 0x1F; 1150 else if (use_mask[2]) idx = 0x2F; 1151 else if (use_mask[3]) idx = 0x3F; 1152 else if (use_mask[4]) idx = 0x4F; 1153 else return -1; 1154 idx -= av_log2_16bit(use_mask[idx >> 4]); 1155 } 1156 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { 1157 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); 1158 n++; 1159 start_off = idx; 1160 } 1161 } 1162 1163 fcb->x[fcb->n] = start_off; 1164 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; 1165 fcb->n++; 1166 1167 /* set offset for next block, relative to start of that block */ 1168 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; 1169 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; 1170 return 0; 1171} 1172 1173/** 1174 * Apply first set of pitch-adaptive window pulses. 1175 * @param s WMA Voice decoding context private data 1176 * @param gb bit I/O context 1177 * @param block_idx block index in frame [0, 1] 1178 * @param fcb storage location for fixed codebook pulse info 1179 */ 1180static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, 1181 int block_idx, AMRFixed *fcb) 1182{ 1183 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); 1184 float v; 1185 1186 if (s->aw_n_pulses[block_idx] > 0) { 1187 int n, v_mask, i_mask, sh, n_pulses; 1188 1189 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each 1190 n_pulses = 3; 1191 v_mask = 8; 1192 i_mask = 7; 1193 sh = 4; 1194 } else { // 4 pulses, 1:sign + 2:index each 1195 n_pulses = 4; 1196 v_mask = 4; 1197 i_mask = 3; 1198 sh = 3; 1199 } 1200 1201 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { 1202 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; 1203 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + 1204 s->aw_first_pulse_off[block_idx]; 1205 while (fcb->x[fcb->n] < 0) 1206 fcb->x[fcb->n] += fcb->pitch_lag; 1207 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) 1208 fcb->n++; 1209 } 1210 } else { 1211 int num2 = (val & 0x1FF) >> 1, delta, idx; 1212 1213 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } 1214 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } 1215 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } 1216 else { delta = 7; idx = num2 + 1 - 3 * 75; } 1217 v = (val & 0x200) ? -1.0 : 1.0; 1218 1219 fcb->no_repeat_mask |= 3 << fcb->n; 1220 fcb->x[fcb->n] = idx - delta; 1221 fcb->y[fcb->n] = v; 1222 fcb->x[fcb->n + 1] = idx; 1223 fcb->y[fcb->n + 1] = (val & 1) ? -v : v; 1224 fcb->n += 2; 1225 } 1226} 1227 1228/** 1229 * @} 1230 * 1231 * Generate a random number from frame_cntr and block_idx, which will live 1232 * in the range [0, 1000 - block_size] (so it can be used as an index in a 1233 * table of size 1000 of which you want to read block_size entries). 1234 * 1235 * @param frame_cntr current frame number 1236 * @param block_num current block index 1237 * @param block_size amount of entries we want to read from a table 1238 * that has 1000 entries 1239 * @return a (non-)random number in the [0, 1000 - block_size] range. 1240 */ 1241static int pRNG(int frame_cntr, int block_num, int block_size) 1242{ 1243 /* array to simplify the calculation of z: 1244 * y = (x % 9) * 5 + 6; 1245 * z = (49995 * x) / y; 1246 * Since y only has 9 values, we can remove the division by using a 1247 * LUT and using FASTDIV-style divisions. For each of the 9 values 1248 * of y, we can rewrite z as: 1249 * z = x * (49995 / y) + x * ((49995 % y) / y) 1250 * In this table, each col represents one possible value of y, the 1251 * first number is 49995 / y, and the second is the FASTDIV variant 1252 * of 49995 % y / y. */ 1253 static const unsigned int div_tbl[9][2] = { 1254 { 8332, 3 * 715827883U }, // y = 6 1255 { 4545, 0 * 390451573U }, // y = 11 1256 { 3124, 11 * 268435456U }, // y = 16 1257 { 2380, 15 * 204522253U }, // y = 21 1258 { 1922, 23 * 165191050U }, // y = 26 1259 { 1612, 23 * 138547333U }, // y = 31 1260 { 1388, 27 * 119304648U }, // y = 36 1261 { 1219, 16 * 104755300U }, // y = 41 1262 { 1086, 39 * 93368855U } // y = 46 1263 }; 1264 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; 1265 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, 1266 // so this is effectively a modulo (%) 1267 y = x - 9 * MULH(477218589, x); // x % 9 1268 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); 1269 // z = x * 49995 / (y * 5 + 6) 1270 return z % (1000 - block_size); 1271} 1272 1273/** 1274 * Parse hardcoded signal for a single block. 1275 * @note see #synth_block(). 1276 */ 1277static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, 1278 int block_idx, int size, 1279 const struct frame_type_desc *frame_desc, 1280 float *excitation) 1281{ 1282 float gain; 1283 int n, r_idx; 1284 1285 av_assert0(size <= MAX_FRAMESIZE); 1286 1287 /* Set the offset from which we start reading wmavoice_std_codebook */ 1288 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { 1289 r_idx = pRNG(s->frame_cntr, block_idx, size); 1290 gain = s->silence_gain; 1291 } else /* FCB_TYPE_HARDCODED */ { 1292 r_idx = get_bits(gb, 8); 1293 gain = wmavoice_gain_universal[get_bits(gb, 6)]; 1294 } 1295 1296 /* Clear gain prediction parameters */ 1297 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); 1298 1299 /* Apply gain to hardcoded codebook and use that as excitation signal */ 1300 for (n = 0; n < size; n++) 1301 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; 1302} 1303 1304/** 1305 * Parse FCB/ACB signal for a single block. 1306 * @note see #synth_block(). 1307 */ 1308static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, 1309 int block_idx, int size, 1310 int block_pitch_sh2, 1311 const struct frame_type_desc *frame_desc, 1312 float *excitation) 1313{ 1314 static const float gain_coeff[6] = { 1315 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 1316 }; 1317 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; 1318 int n, idx, gain_weight; 1319 AMRFixed fcb; 1320 1321 av_assert0(size <= MAX_FRAMESIZE / 2); 1322 memset(pulses, 0, sizeof(*pulses) * size); 1323 1324 fcb.pitch_lag = block_pitch_sh2 >> 2; 1325 fcb.pitch_fac = 1.0; 1326 fcb.no_repeat_mask = 0; 1327 fcb.n = 0; 1328 1329 /* For the other frame types, this is where we apply the innovation 1330 * (fixed) codebook pulses of the speech signal. */ 1331 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { 1332 aw_pulse_set1(s, gb, block_idx, &fcb); 1333 if (aw_pulse_set2(s, gb, block_idx, &fcb)) { 1334 /* Conceal the block with silence and return. 1335 * Skip the correct amount of bits to read the next 1336 * block from the correct offset. */ 1337 int r_idx = pRNG(s->frame_cntr, block_idx, size); 1338 1339 for (n = 0; n < size; n++) 1340 excitation[n] = 1341 wmavoice_std_codebook[r_idx + n] * s->silence_gain; 1342 skip_bits(gb, 7 + 1); 1343 return; 1344 } 1345 } else /* FCB_TYPE_EXC_PULSES */ { 1346 int offset_nbits = 5 - frame_desc->log_n_blocks; 1347 1348 fcb.no_repeat_mask = -1; 1349 /* similar to ff_decode_10_pulses_35bits(), but with single pulses 1350 * (instead of double) for a subset of pulses */ 1351 for (n = 0; n < 5; n++) { 1352 float sign; 1353 int pos1, pos2; 1354 1355 sign = get_bits1(gb) ? 1.0 : -1.0; 1356 pos1 = get_bits(gb, offset_nbits); 1357 fcb.x[fcb.n] = n + 5 * pos1; 1358 fcb.y[fcb.n++] = sign; 1359 if (n < frame_desc->dbl_pulses) { 1360 pos2 = get_bits(gb, offset_nbits); 1361 fcb.x[fcb.n] = n + 5 * pos2; 1362 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; 1363 } 1364 } 1365 } 1366 ff_set_fixed_vector(pulses, &fcb, 1.0, size); 1367 1368 /* Calculate gain for adaptive & fixed codebook signal. 1369 * see ff_amr_set_fixed_gain(). */ 1370 idx = get_bits(gb, 7); 1371 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, 1372 gain_coeff, 6) - 1373 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); 1374 acb_gain = wmavoice_gain_codebook_acb[idx]; 1375 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], 1376 -2.9957322736 /* log(0.05) */, 1377 1.6094379124 /* log(5.0) */); 1378 1379 gain_weight = 8 >> frame_desc->log_n_blocks; 1380 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, 1381 sizeof(*s->gain_pred_err) * (6 - gain_weight)); 1382 for (n = 0; n < gain_weight; n++) 1383 s->gain_pred_err[n] = pred_err; 1384 1385 /* Calculation of adaptive codebook */ 1386 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { 1387 int len; 1388 for (n = 0; n < size; n += len) { 1389 int next_idx_sh16; 1390 int abs_idx = block_idx * size + n; 1391 int pitch_sh16 = (s->last_pitch_val << 16) + 1392 s->pitch_diff_sh16 * abs_idx; 1393 int pitch = (pitch_sh16 + 0x6FFF) >> 16; 1394 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; 1395 idx = idx_sh16 >> 16; 1396 if (s->pitch_diff_sh16) { 1397 if (s->pitch_diff_sh16 > 0) { 1398 next_idx_sh16 = (idx_sh16) &~ 0xFFFF; 1399 } else 1400 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; 1401 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, 1402 1, size - n); 1403 } else 1404 len = size; 1405 1406 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], 1407 wmavoice_ipol1_coeffs, 17, 1408 idx, 9, len); 1409 } 1410 } else /* ACB_TYPE_HAMMING */ { 1411 int block_pitch = block_pitch_sh2 >> 2; 1412 idx = block_pitch_sh2 & 3; 1413 if (idx) { 1414 ff_acelp_interpolatef(excitation, &excitation[-block_pitch], 1415 wmavoice_ipol2_coeffs, 4, 1416 idx, 8, size); 1417 } else 1418 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, 1419 sizeof(float) * size); 1420 } 1421 1422 /* Interpolate ACB/FCB and use as excitation signal */ 1423 ff_weighted_vector_sumf(excitation, excitation, pulses, 1424 acb_gain, fcb_gain, size); 1425} 1426 1427/** 1428 * Parse data in a single block. 1429 * 1430 * @param s WMA Voice decoding context private data 1431 * @param gb bit I/O context 1432 * @param block_idx index of the to-be-read block 1433 * @param size amount of samples to be read in this block 1434 * @param block_pitch_sh2 pitch for this block << 2 1435 * @param lsps LSPs for (the end of) this frame 1436 * @param prev_lsps LSPs for the last frame 1437 * @param frame_desc frame type descriptor 1438 * @param excitation target memory for the ACB+FCB interpolated signal 1439 * @param synth target memory for the speech synthesis filter output 1440 * @return 0 on success, <0 on error. 1441 */ 1442static void synth_block(WMAVoiceContext *s, GetBitContext *gb, 1443 int block_idx, int size, 1444 int block_pitch_sh2, 1445 const double *lsps, const double *prev_lsps, 1446 const struct frame_type_desc *frame_desc, 1447 float *excitation, float *synth) 1448{ 1449 double i_lsps[MAX_LSPS]; 1450 float lpcs[MAX_LSPS]; 1451 float fac; 1452 int n; 1453 1454 if (frame_desc->acb_type == ACB_TYPE_NONE) 1455 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); 1456 else 1457 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, 1458 frame_desc, excitation); 1459 1460 /* convert interpolated LSPs to LPCs */ 1461 fac = (block_idx + 0.5) / frame_desc->n_blocks; 1462 for (n = 0; n < s->lsps; n++) // LSF -> LSP 1463 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); 1464 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); 1465 1466 /* Speech synthesis */ 1467 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); 1468} 1469 1470/** 1471 * Synthesize output samples for a single frame. 1472 * 1473 * @param ctx WMA Voice decoder context 1474 * @param gb bit I/O context (s->gb or one for cross-packet superframes) 1475 * @param frame_idx Frame number within superframe [0-2] 1476 * @param samples pointer to output sample buffer, has space for at least 160 1477 * samples 1478 * @param lsps LSP array 1479 * @param prev_lsps array of previous frame's LSPs 1480 * @param excitation target buffer for excitation signal 1481 * @param synth target buffer for synthesized speech data 1482 * @return 0 on success, <0 on error. 1483 */ 1484static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, 1485 float *samples, 1486 const double *lsps, const double *prev_lsps, 1487 float *excitation, float *synth) 1488{ 1489 WMAVoiceContext *s = ctx->priv_data; 1490 int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val); 1491 int pitch[MAX_BLOCKS], av_uninit(last_block_pitch); 1492 1493 /* Parse frame type ("frame header"), see frame_descs */ 1494 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; 1495 1496 pitch[0] = INT_MAX; 1497 1498 if (bd_idx < 0) { 1499 av_log(ctx, AV_LOG_ERROR, 1500 "Invalid frame type VLC code, skipping\n"); 1501 return AVERROR_INVALIDDATA; 1502 } 1503 1504 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; 1505 1506 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ 1507 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { 1508 /* Pitch is provided per frame, which is interpreted as the pitch of 1509 * the last sample of the last block of this frame. We can interpolate 1510 * the pitch of other blocks (and even pitch-per-sample) by gradually 1511 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ 1512 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; 1513 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; 1514 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); 1515 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); 1516 if (s->last_acb_type == ACB_TYPE_NONE || 1517 20 * abs(cur_pitch_val - s->last_pitch_val) > 1518 (cur_pitch_val + s->last_pitch_val)) 1519 s->last_pitch_val = cur_pitch_val; 1520 1521 /* pitch per block */ 1522 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { 1523 int fac = n * 2 + 1; 1524 1525 pitch[n] = (MUL16(fac, cur_pitch_val) + 1526 MUL16((n_blocks_x2 - fac), s->last_pitch_val) + 1527 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; 1528 } 1529 1530 /* "pitch-diff-per-sample" for calculation of pitch per sample */ 1531 s->pitch_diff_sh16 = 1532 (cur_pitch_val - s->last_pitch_val) * (1 << 16) / MAX_FRAMESIZE; 1533 } 1534 1535 /* Global gain (if silence) and pitch-adaptive window coordinates */ 1536 switch (frame_descs[bd_idx].fcb_type) { 1537 case FCB_TYPE_SILENCE: 1538 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; 1539 break; 1540 case FCB_TYPE_AW_PULSES: 1541 aw_parse_coords(s, gb, pitch); 1542 break; 1543 } 1544 1545 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { 1546 int bl_pitch_sh2; 1547 1548 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ 1549 switch (frame_descs[bd_idx].acb_type) { 1550 case ACB_TYPE_HAMMING: { 1551 /* Pitch is given per block. Per-block pitches are encoded as an 1552 * absolute value for the first block, and then delta values 1553 * relative to this value) for all subsequent blocks. The scale of 1554 * this pitch value is semi-logarithmic compared to its use in the 1555 * decoder, so we convert it to normal scale also. */ 1556 int block_pitch, 1557 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, 1558 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, 1559 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; 1560 1561 if (n == 0) { 1562 block_pitch = get_bits(gb, s->block_pitch_nbits); 1563 } else 1564 block_pitch = last_block_pitch - s->block_delta_pitch_hrange + 1565 get_bits(gb, s->block_delta_pitch_nbits); 1566 /* Convert last_ so that any next delta is within _range */ 1567 last_block_pitch = av_clip(block_pitch, 1568 s->block_delta_pitch_hrange, 1569 s->block_pitch_range - 1570 s->block_delta_pitch_hrange); 1571 1572 /* Convert semi-log-style scale back to normal scale */ 1573 if (block_pitch < t1) { 1574 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; 1575 } else { 1576 block_pitch -= t1; 1577 if (block_pitch < t2) { 1578 bl_pitch_sh2 = 1579 (s->block_conv_table[1] << 2) + (block_pitch << 1); 1580 } else { 1581 block_pitch -= t2; 1582 if (block_pitch < t3) { 1583 bl_pitch_sh2 = 1584 (s->block_conv_table[2] + block_pitch) << 2; 1585 } else 1586 bl_pitch_sh2 = s->block_conv_table[3] << 2; 1587 } 1588 } 1589 pitch[n] = bl_pitch_sh2 >> 2; 1590 break; 1591 } 1592 1593 case ACB_TYPE_ASYMMETRIC: { 1594 bl_pitch_sh2 = pitch[n] << 2; 1595 break; 1596 } 1597 1598 default: // ACB_TYPE_NONE has no pitch 1599 bl_pitch_sh2 = 0; 1600 break; 1601 } 1602 1603 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, 1604 lsps, prev_lsps, &frame_descs[bd_idx], 1605 &excitation[n * block_nsamples], 1606 &synth[n * block_nsamples]); 1607 } 1608 1609 /* Averaging projection filter, if applicable. Else, just copy samples 1610 * from synthesis buffer */ 1611 if (s->do_apf) { 1612 double i_lsps[MAX_LSPS]; 1613 float lpcs[MAX_LSPS]; 1614 1615 if(frame_descs[bd_idx].fcb_type >= FCB_TYPE_AW_PULSES && pitch[0] == INT_MAX) 1616 return AVERROR_INVALIDDATA; 1617 1618 for (n = 0; n < s->lsps; n++) // LSF -> LSP 1619 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); 1620 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); 1621 postfilter(s, synth, samples, 80, lpcs, 1622 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], 1623 frame_descs[bd_idx].fcb_type, pitch[0]); 1624 1625 for (n = 0; n < s->lsps; n++) // LSF -> LSP 1626 i_lsps[n] = cos(lsps[n]); 1627 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); 1628 postfilter(s, &synth[80], &samples[80], 80, lpcs, 1629 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], 1630 frame_descs[bd_idx].fcb_type, pitch[0]); 1631 } else 1632 memcpy(samples, synth, 160 * sizeof(synth[0])); 1633 1634 /* Cache values for next frame */ 1635 s->frame_cntr++; 1636 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) 1637 s->last_acb_type = frame_descs[bd_idx].acb_type; 1638 switch (frame_descs[bd_idx].acb_type) { 1639 case ACB_TYPE_NONE: 1640 s->last_pitch_val = 0; 1641 break; 1642 case ACB_TYPE_ASYMMETRIC: 1643 s->last_pitch_val = cur_pitch_val; 1644 break; 1645 case ACB_TYPE_HAMMING: 1646 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; 1647 break; 1648 } 1649 1650 return 0; 1651} 1652 1653/** 1654 * Ensure minimum value for first item, maximum value for last value, 1655 * proper spacing between each value and proper ordering. 1656 * 1657 * @param lsps array of LSPs 1658 * @param num size of LSP array 1659 * 1660 * @note basically a double version of #ff_acelp_reorder_lsf(), might be 1661 * useful to put in a generic location later on. Parts are also 1662 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), 1663 * which is in float. 1664 */ 1665static void stabilize_lsps(double *lsps, int num) 1666{ 1667 int n, m, l; 1668 1669 /* set minimum value for first, maximum value for last and minimum 1670 * spacing between LSF values. 1671 * Very similar to ff_set_min_dist_lsf(), but in double. */ 1672 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); 1673 for (n = 1; n < num; n++) 1674 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); 1675 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); 1676 1677 /* reorder (looks like one-time / non-recursed bubblesort). 1678 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ 1679 for (n = 1; n < num; n++) { 1680 if (lsps[n] < lsps[n - 1]) { 1681 for (m = 1; m < num; m++) { 1682 double tmp = lsps[m]; 1683 for (l = m - 1; l >= 0; l--) { 1684 if (lsps[l] <= tmp) break; 1685 lsps[l + 1] = lsps[l]; 1686 } 1687 lsps[l + 1] = tmp; 1688 } 1689 break; 1690 } 1691 } 1692} 1693 1694/** 1695 * Synthesize output samples for a single superframe. If we have any data 1696 * cached in s->sframe_cache, that will be used instead of whatever is loaded 1697 * in s->gb. 1698 * 1699 * WMA Voice superframes contain 3 frames, each containing 160 audio samples, 1700 * to give a total of 480 samples per frame. See #synth_frame() for frame 1701 * parsing. In addition to 3 frames, superframes can also contain the LSPs 1702 * (if these are globally specified for all frames (residually); they can 1703 * also be specified individually per-frame. See the s->has_residual_lsps 1704 * option), and can specify the number of samples encoded in this superframe 1705 * (if less than 480), usually used to prevent blanks at track boundaries. 1706 * 1707 * @param ctx WMA Voice decoder context 1708 * @return 0 on success, <0 on error or 1 if there was not enough data to 1709 * fully parse the superframe 1710 */ 1711static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, 1712 int *got_frame_ptr) 1713{ 1714 WMAVoiceContext *s = ctx->priv_data; 1715 GetBitContext *gb = &s->gb, s_gb; 1716 int n, res, n_samples = MAX_SFRAMESIZE; 1717 double lsps[MAX_FRAMES][MAX_LSPS]; 1718 const double *mean_lsf = s->lsps == 16 ? 1719 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; 1720 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; 1721 float synth[MAX_LSPS + MAX_SFRAMESIZE]; 1722 float *samples; 1723 1724 memcpy(synth, s->synth_history, 1725 s->lsps * sizeof(*synth)); 1726 memcpy(excitation, s->excitation_history, 1727 s->history_nsamples * sizeof(*excitation)); 1728 1729 if (s->sframe_cache_size > 0) { 1730 gb = &s_gb; 1731 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); 1732 s->sframe_cache_size = 0; 1733 } 1734 1735 /* First bit is speech/music bit, it differentiates between WMAVoice 1736 * speech samples (the actual codec) and WMAVoice music samples, which 1737 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in 1738 * the wild yet. */ 1739 if (!get_bits1(gb)) { 1740 avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); 1741 return AVERROR_PATCHWELCOME; 1742 } 1743 1744 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ 1745 if (get_bits1(gb)) { 1746 if ((n_samples = get_bits(gb, 12)) > MAX_SFRAMESIZE) { 1747 av_log(ctx, AV_LOG_ERROR, 1748 "Superframe encodes > %d samples (%d), not allowed\n", 1749 MAX_SFRAMESIZE, n_samples); 1750 return AVERROR_INVALIDDATA; 1751 } 1752 } 1753 1754 /* Parse LSPs, if global for the superframe (can also be per-frame). */ 1755 if (s->has_residual_lsps) { 1756 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; 1757 1758 for (n = 0; n < s->lsps; n++) 1759 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; 1760 1761 if (s->lsps == 10) { 1762 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); 1763 } else /* s->lsps == 16 */ 1764 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); 1765 1766 for (n = 0; n < s->lsps; n++) { 1767 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); 1768 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); 1769 lsps[2][n] += mean_lsf[n]; 1770 } 1771 for (n = 0; n < 3; n++) 1772 stabilize_lsps(lsps[n], s->lsps); 1773 } 1774 1775 /* synth_superframe can run multiple times per packet 1776 * free potential previous frame */ 1777 av_frame_unref(frame); 1778 1779 /* get output buffer */ 1780 frame->nb_samples = MAX_SFRAMESIZE; 1781 if ((res = ff_get_buffer(ctx, frame, 0)) < 0) 1782 return res; 1783 frame->nb_samples = n_samples; 1784 samples = (float *)frame->data[0]; 1785 1786 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */ 1787 for (n = 0; n < 3; n++) { 1788 if (!s->has_residual_lsps) { 1789 int m; 1790 1791 if (s->lsps == 10) { 1792 dequant_lsp10i(gb, lsps[n]); 1793 } else /* s->lsps == 16 */ 1794 dequant_lsp16i(gb, lsps[n]); 1795 1796 for (m = 0; m < s->lsps; m++) 1797 lsps[n][m] += mean_lsf[m]; 1798 stabilize_lsps(lsps[n], s->lsps); 1799 } 1800 1801 if ((res = synth_frame(ctx, gb, n, 1802 &samples[n * MAX_FRAMESIZE], 1803 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], 1804 &excitation[s->history_nsamples + n * MAX_FRAMESIZE], 1805 &synth[s->lsps + n * MAX_FRAMESIZE]))) { 1806 *got_frame_ptr = 0; 1807 return res; 1808 } 1809 } 1810 1811 /* Statistics? FIXME - we don't check for length, a slight overrun 1812 * will be caught by internal buffer padding, and anything else 1813 * will be skipped, not read. */ 1814 if (get_bits1(gb)) { 1815 res = get_bits(gb, 4); 1816 skip_bits(gb, 10 * (res + 1)); 1817 } 1818 1819 if (get_bits_left(gb) < 0) { 1820 wmavoice_flush(ctx); 1821 return AVERROR_INVALIDDATA; 1822 } 1823 1824 *got_frame_ptr = 1; 1825 1826 /* Update history */ 1827 memcpy(s->prev_lsps, lsps[2], 1828 s->lsps * sizeof(*s->prev_lsps)); 1829 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], 1830 s->lsps * sizeof(*synth)); 1831 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], 1832 s->history_nsamples * sizeof(*excitation)); 1833 if (s->do_apf) 1834 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], 1835 s->history_nsamples * sizeof(*s->zero_exc_pf)); 1836 1837 return 0; 1838} 1839 1840/** 1841 * Parse the packet header at the start of each packet (input data to this 1842 * decoder). 1843 * 1844 * @param s WMA Voice decoding context private data 1845 * @return <0 on error, nb_superframes on success. 1846 */ 1847static int parse_packet_header(WMAVoiceContext *s) 1848{ 1849 GetBitContext *gb = &s->gb; 1850 unsigned int res, n_superframes = 0; 1851 1852 skip_bits(gb, 4); // packet sequence number 1853 s->has_residual_lsps = get_bits1(gb); 1854 do { 1855 if (get_bits_left(gb) < 6 + s->spillover_bitsize) 1856 return AVERROR_INVALIDDATA; 1857 1858 res = get_bits(gb, 6); // number of superframes per packet 1859 // (minus first one if there is spillover) 1860 n_superframes += res; 1861 } while (res == 0x3F); 1862 s->spillover_nbits = get_bits(gb, s->spillover_bitsize); 1863 1864 return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA; 1865} 1866 1867/** 1868 * Copy (unaligned) bits from gb/data/size to pb. 1869 * 1870 * @param pb target buffer to copy bits into 1871 * @param data source buffer to copy bits from 1872 * @param size size of the source data, in bytes 1873 * @param gb bit I/O context specifying the current position in the source. 1874 * data. This function might use this to align the bit position to 1875 * a whole-byte boundary before calling #ff_copy_bits() on aligned 1876 * source data 1877 * @param nbits the amount of bits to copy from source to target 1878 * 1879 * @note after calling this function, the current position in the input bit 1880 * I/O context is undefined. 1881 */ 1882static void copy_bits(PutBitContext *pb, 1883 const uint8_t *data, int size, 1884 GetBitContext *gb, int nbits) 1885{ 1886 int rmn_bytes, rmn_bits; 1887 1888 rmn_bits = rmn_bytes = get_bits_left(gb); 1889 if (rmn_bits < nbits) 1890 return; 1891 if (nbits > put_bits_left(pb)) 1892 return; 1893 rmn_bits &= 7; rmn_bytes >>= 3; 1894 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) 1895 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); 1896 ff_copy_bits(pb, data + size - rmn_bytes, 1897 FFMIN(nbits - rmn_bits, rmn_bytes << 3)); 1898} 1899 1900/** 1901 * Packet decoding: a packet is anything that the (ASF) demuxer contains, 1902 * and we expect that the demuxer / application provides it to us as such 1903 * (else you'll probably get garbage as output). Every packet has a size of 1904 * ctx->block_align bytes, starts with a packet header (see 1905 * #parse_packet_header()), and then a series of superframes. Superframe 1906 * boundaries may exceed packets, i.e. superframes can split data over 1907 * multiple (two) packets. 1908 * 1909 * For more information about frames, see #synth_superframe(). 1910 */ 1911static int wmavoice_decode_packet(AVCodecContext *ctx, AVFrame *frame, 1912 int *got_frame_ptr, AVPacket *avpkt) 1913{ 1914 WMAVoiceContext *s = ctx->priv_data; 1915 GetBitContext *gb = &s->gb; 1916 int size, res, pos; 1917 1918 /* Packets are sometimes a multiple of ctx->block_align, with a packet 1919 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer 1920 * feeds us ASF packets, which may concatenate multiple "codec" packets 1921 * in a single "muxer" packet, so we artificially emulate that by 1922 * capping the packet size at ctx->block_align. */ 1923 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); 1924 init_get_bits8(&s->gb, avpkt->data, size); 1925 1926 /* size == ctx->block_align is used to indicate whether we are dealing with 1927 * a new packet or a packet of which we already read the packet header 1928 * previously. */ 1929 if (!(size % ctx->block_align)) { // new packet header 1930 if (!size) { 1931 s->spillover_nbits = 0; 1932 s->nb_superframes = 0; 1933 } else { 1934 if ((res = parse_packet_header(s)) < 0) 1935 return res; 1936 s->nb_superframes = res; 1937 } 1938 1939 /* If the packet header specifies a s->spillover_nbits, then we want 1940 * to push out all data of the previous packet (+ spillover) before 1941 * continuing to parse new superframes in the current packet. */ 1942 if (s->sframe_cache_size > 0) { 1943 int cnt = get_bits_count(gb); 1944 if (cnt + s->spillover_nbits > avpkt->size * 8) { 1945 s->spillover_nbits = avpkt->size * 8 - cnt; 1946 } 1947 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); 1948 flush_put_bits(&s->pb); 1949 s->sframe_cache_size += s->spillover_nbits; 1950 if ((res = synth_superframe(ctx, frame, got_frame_ptr)) == 0 && 1951 *got_frame_ptr) { 1952 cnt += s->spillover_nbits; 1953 s->skip_bits_next = cnt & 7; 1954 res = cnt >> 3; 1955 return res; 1956 } else 1957 skip_bits_long (gb, s->spillover_nbits - cnt + 1958 get_bits_count(gb)); // resync 1959 } else if (s->spillover_nbits) { 1960 skip_bits_long(gb, s->spillover_nbits); // resync 1961 } 1962 } else if (s->skip_bits_next) 1963 skip_bits(gb, s->skip_bits_next); 1964 1965 /* Try parsing superframes in current packet */ 1966 s->sframe_cache_size = 0; 1967 s->skip_bits_next = 0; 1968 pos = get_bits_left(gb); 1969 if (s->nb_superframes-- == 0) { 1970 *got_frame_ptr = 0; 1971 return size; 1972 } else if (s->nb_superframes > 0) { 1973 if ((res = synth_superframe(ctx, frame, got_frame_ptr)) < 0) { 1974 return res; 1975 } else if (*got_frame_ptr) { 1976 int cnt = get_bits_count(gb); 1977 s->skip_bits_next = cnt & 7; 1978 res = cnt >> 3; 1979 return res; 1980 } 1981 } else if ((s->sframe_cache_size = pos) > 0) { 1982 /* ... cache it for spillover in next packet */ 1983 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); 1984 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); 1985 // FIXME bad - just copy bytes as whole and add use the 1986 // skip_bits_next field 1987 } 1988 1989 return size; 1990} 1991 1992static av_cold int wmavoice_decode_end(AVCodecContext *ctx) 1993{ 1994 WMAVoiceContext *s = ctx->priv_data; 1995 1996 if (s->do_apf) { 1997 ff_rdft_end(&s->rdft); 1998 ff_rdft_end(&s->irdft); 1999 ff_dct_end(&s->dct); 2000 ff_dct_end(&s->dst); 2001 } 2002 2003 return 0; 2004} 2005 2006const FFCodec ff_wmavoice_decoder = { 2007 .p.name = "wmavoice", 2008 .p.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), 2009 .p.type = AVMEDIA_TYPE_AUDIO, 2010 .p.id = AV_CODEC_ID_WMAVOICE, 2011 .priv_data_size = sizeof(WMAVoiceContext), 2012 .init = wmavoice_decode_init, 2013 .close = wmavoice_decode_end, 2014 FF_CODEC_DECODE_CB(wmavoice_decode_packet), 2015 .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, 2016 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 2017 .flush = wmavoice_flush, 2018}; 2019