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| 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 libavcodec/wmavoice.c |
| 24 * @brief Windows Media Audio Voice compatible decoder |
| 25 * @author Ronald S. Bultje <rsbultje@gmail.com> |
| 26 */ |
| 27 |
| 28 #include <math.h> |
| 29 #include "avcodec.h" |
| 30 #include "get_bits.h" |
| 31 #include "put_bits.h" |
| 32 #include "wmavoice_data.h" |
| 33 #include "celp_math.h" |
| 34 #include "celp_filters.h" |
| 35 #include "acelp_vectors.h" |
| 36 #include "acelp_filters.h" |
| 37 #include "lsp.h" |
| 38 #include "libavutil/lzo.h" |
| 39 |
| 40 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame |
| 41 #define MAX_LSPS 16 ///< maximum filter order |
| 42 #define MAX_FRAMES 3 ///< maximum number of frames per superframe |
| 43 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame |
| 44 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history |
| 45 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) |
| 46 ///< maximum number of samples per superframe |
| 47 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that |
| 48 ///< was split over two packets |
| 49 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration |
| 50 |
| 51 /** |
| 52 * Frame type VLC coding. |
| 53 */ |
| 54 static VLC frame_type_vlc; |
| 55 |
| 56 /** |
| 57 * Adaptive codebook types. |
| 58 */ |
| 59 enum { |
| 60 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) |
| 61 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which |
| 62 ///< we interpolate to get a per-sample pitch. |
| 63 ///< Signal is generated using an asymmetric sinc |
| 64 ///< window function |
| 65 ///< @note see #wmavoice_ipol1_coeffs |
| 66 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using |
| 67 ///< a Hamming sinc window function |
| 68 ///< @note see #wmavoice_ipol2_coeffs |
| 69 }; |
| 70 |
| 71 /** |
| 72 * Fixed codebook types. |
| 73 */ |
| 74 enum { |
| 75 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence |
| 76 ///< generated from a hardcoded (fixed) codebook |
| 77 ///< with per-frame (low) gain values |
| 78 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block |
| 79 ///< gain values |
| 80 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, |
| 81 ///< used in particular for low-bitrate streams |
| 82 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in |
| 83 ///< combinations of either single pulses or |
| 84 ///< pulse pairs |
| 85 }; |
| 86 |
| 87 /** |
| 88 * Description of frame types. |
| 89 */ |
| 90 static const struct frame_type_desc { |
| 91 uint8_t n_blocks; ///< amount of blocks per frame (each block |
| 92 ///< (contains 160/#n_blocks samples) |
| 93 uint8_t log_n_blocks; ///< log2(#n_blocks) |
| 94 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) |
| 95 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) |
| 96 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs |
| 97 ///< (rather than just one single pulse) |
| 98 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES |
| 99 uint16_t frame_size; ///< the amount of bits that make up the block |
| 100 ///< data (per frame) |
| 101 } frame_descs[17] = { |
| 102 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, |
| 103 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, |
| 104 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, |
| 105 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, |
| 106 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, |
| 107 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, |
| 108 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, |
| 109 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, |
| 110 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, |
| 111 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, |
| 112 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, |
| 113 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, |
| 114 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, |
| 115 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, |
| 116 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, |
| 117 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, |
| 118 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } |
| 119 }; |
| 120 |
| 121 /** |
| 122 * WMA Voice decoding context. |
| 123 */ |
| 124 typedef struct { |
| 125 /** |
| 126 * @defgroup struct_global Global values |
| 127 * Global values, specified in the stream header / extradata or used |
| 128 * all over. |
| 129 * @{ |
| 130 */ |
| 131 GetBitContext gb; ///< packet bitreader. During decoder init, |
| 132 ///< it contains the extradata from the |
| 133 ///< demuxer. During decoding, it contains |
| 134 ///< packet data. |
| 135 int8_t vbm_tree[25]; ///< converts VLC codes to frame type |
| 136 |
| 137 int spillover_bitsize; ///< number of bits used to specify |
| 138 ///< #spillover_nbits in the packet header |
| 139 ///< = ceil(log2(ctx->block_align << 3)) |
| 140 int history_nsamples; ///< number of samples in history for signal |
| 141 ///< prediction (through ACB) |
| 142 |
| 143 int do_apf; ///< whether to apply the averaged |
| 144 ///< projection filter (APF) |
| 145 |
| 146 int lsps; ///< number of LSPs per frame [10 or 16] |
| 147 int lsp_q_mode; ///< defines quantizer defaults [0, 1] |
| 148 int lsp_def_mode; ///< defines different sets of LSP defaults |
| 149 ///< [0, 1] |
| 150 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded |
| 151 ///< per-frame (independent coding) |
| 152 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded |
| 153 ///< per superframe (residual coding) |
| 154 |
| 155 int min_pitch_val; ///< base value for pitch parsing code |
| 156 int max_pitch_val; ///< max value + 1 for pitch parsing |
| 157 int pitch_nbits; ///< number of bits used to specify the |
| 158 ///< pitch value in the frame header |
| 159 int block_pitch_nbits; ///< number of bits used to specify the |
| 160 ///< first block's pitch value |
| 161 int block_pitch_range; ///< range of the block pitch |
| 162 int block_delta_pitch_nbits; ///< number of bits used to specify the |
| 163 ///< delta pitch between this and the last |
| 164 ///< block's pitch value, used in all but |
| 165 ///< first block |
| 166 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is |
| 167 ///< from -this to +this-1) |
| 168 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale |
| 169 ///< conversion |
| 170 |
| 171 /** |
| 172 * @} |
| 173 * @defgroup struct_packet Packet values |
| 174 * Packet values, specified in the packet header or related to a packet. |
| 175 * A packet is considered to be a single unit of data provided to this |
| 176 * decoder by the demuxer. |
| 177 * @{ |
| 178 */ |
| 179 int spillover_nbits; ///< number of bits of the previous packet's |
| 180 ///< last superframe preceeding this |
| 181 ///< packet's first full superframe (useful |
| 182 ///< for re-synchronization also) |
| 183 int has_residual_lsps; ///< if set, superframes contain one set of |
| 184 ///< LSPs that cover all frames, encoded as |
| 185 ///< independent and residual LSPs; if not |
| 186 ///< set, each frame contains its own, fully |
| 187 ///< independent, LSPs |
| 188 int skip_bits_next; ///< number of bits to skip at the next call |
| 189 ///< to #wmavoice_decode_packet() (since |
| 190 ///< they're part of the previous superframe) |
| 191 |
| 192 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE]; |
| 193 ///< cache for superframe data split over |
| 194 ///< multiple packets |
| 195 int sframe_cache_size; ///< set to >0 if we have data from an |
| 196 ///< (incomplete) superframe from a previous |
| 197 ///< packet that spilled over in the current |
| 198 ///< packet; specifies the amount of bits in |
| 199 ///< #sframe_cache |
| 200 PutBitContext pb; ///< bitstream writer for #sframe_cache |
| 201 |
| 202 /** |
| 203 * @} |
| 204 * @defgroup struct_frame Frame and superframe values |
| 205 * Superframe and frame data - these can change from frame to frame, |
| 206 * although some of them do in that case serve as a cache / history for |
| 207 * the next frame or superframe. |
| 208 * @{ |
| 209 */ |
| 210 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous |
| 211 ///< superframe |
| 212 int last_pitch_val; ///< pitch value of the previous frame |
| 213 int last_acb_type; ///< frame type [0-2] of the previous frame |
| 214 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) |
| 215 ///< << 16) / #MAX_FRAMESIZE |
| 216 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE |
| 217 |
| 218 int aw_idx_is_ext; ///< whether the AW index was encoded in |
| 219 ///< 8 bits (instead of 6) |
| 220 int aw_pulse_range; ///< the range over which #aw_pulse_set1() |
| 221 ///< can apply the pulse, relative to the |
| 222 ///< value in aw_first_pulse_off. The exact |
| 223 ///< position of the first AW-pulse is within |
| 224 ///< [pulse_off, pulse_off + this], and |
| 225 ///< depends on bitstream values; [16 or 24] |
| 226 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note |
| 227 ///< that this number can be negative (in |
| 228 ///< which case it basically means "zero") |
| 229 int aw_first_pulse_off[2]; ///< index of first sample to which to |
| 230 ///< apply AW-pulses, or -0xff if unset |
| 231 int aw_next_pulse_off_cache; ///< the position (relative to start of the |
| 232 ///< second block) at which pulses should |
| 233 ///< start to be positioned, serves as a |
| 234 ///< cache for pitch-adaptive window pulses |
| 235 ///< between blocks |
| 236 |
| 237 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is |
| 238 ///< only used for comfort noise in #pRNG() |
| 239 float gain_pred_err[6]; ///< cache for gain prediction |
| 240 float excitation_history[MAX_SIGNAL_HISTORY]; |
| 241 ///< cache of the signal of previous |
| 242 ///< superframes, used as a history for |
| 243 ///< signal generation |
| 244 float synth_history[MAX_LSPS]; ///< see #excitation_history |
| 245 /** |
| 246 * @} |
| 247 */ |
| 248 } WMAVoiceContext; |
| 249 |
| 250 /** |
| 251 * Sets up the variable bit mode (VBM) tree from container extradata. |
| 252 * @param gb bit I/O context. |
| 253 * The bit context (s->gb) should be loaded with byte 23-46 of the |
| 254 * container extradata (i.e. the ones containing the VBM tree). |
| 255 * @param vbm_tree pointer to array to which the decoded VBM tree will be |
| 256 * written. |
| 257 * @return 0 on success, <0 on error. |
| 258 */ |
| 259 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) |
| 260 { |
| 261 static const uint8_t bits[] = { |
| 262 2, 2, 2, 4, 4, 4, |
| 263 6, 6, 6, 8, 8, 8, |
| 264 10, 10, 10, 12, 12, 12, |
| 265 14, 14, 14, 14 |
| 266 }; |
| 267 static const uint16_t codes[] = { |
| 268 0x0000, 0x0001, 0x0002, // 00/01/10 |
| 269 0x000c, 0x000d, 0x000e, // 11+00/01/10 |
| 270 0x003c, 0x003d, 0x003e, // 1111+00/01/10 |
| 271 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 |
| 272 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 |
| 273 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 |
| 274 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx |
| 275 }; |
| 276 int cntr[8], n, res; |
| 277 |
| 278 memset(vbm_tree, 0xff, sizeof(vbm_tree)); |
| 279 memset(cntr, 0, sizeof(cntr)); |
| 280 for (n = 0; n < 17; n++) { |
| 281 res = get_bits(gb, 3); |
| 282 if (cntr[res] > 3) // should be >= 3 + (res == 7)) |
| 283 return -1; |
| 284 vbm_tree[res * 3 + cntr[res]++] = n; |
| 285 } |
| 286 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), |
| 287 bits, 1, 1, codes, 2, 2, 132); |
| 288 return 0; |
| 289 } |
| 290 |
| 291 /** |
| 292 * Set up decoder with parameters from demuxer (extradata etc.). |
| 293 */ |
| 294 static av_cold int wmavoice_decode_init(AVCodecContext *ctx) |
| 295 { |
| 296 int n, flags, pitch_range, lsp16_flag; |
| 297 WMAVoiceContext *s = ctx->priv_data; |
| 298 |
| 299 /** |
| 300 * Extradata layout: |
| 301 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), |
| 302 * - byte 19-22: flags field (annoyingly in LE; see below for known |
| 303 * values), |
| 304 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, |
| 305 * rest is 0). |
| 306 */ |
| 307 if (ctx->extradata_size != 46) { |
| 308 av_log(ctx, AV_LOG_ERROR, |
| 309 "Invalid extradata size %d (should be 46)\n", |
| 310 ctx->extradata_size); |
| 311 return -1; |
| 312 } |
| 313 flags = AV_RL32(ctx->extradata + 18); |
| 314 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); |
| 315 s->do_apf = flags & 0x1; |
| 316 s->lsp_q_mode = !!(flags & 0x2000); |
| 317 s->lsp_def_mode = !!(flags & 0x4000); |
| 318 lsp16_flag = flags & 0x1000; |
| 319 if (lsp16_flag) { |
| 320 s->lsps = 16; |
| 321 s->frame_lsp_bitsize = 34; |
| 322 s->sframe_lsp_bitsize = 60; |
| 323 } else { |
| 324 s->lsps = 10; |
| 325 s->frame_lsp_bitsize = 24; |
| 326 s->sframe_lsp_bitsize = 48; |
| 327 } |
| 328 for (n = 0; n < s->lsps; n++) |
| 329 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); |
| 330 |
| 331 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); |
| 332 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { |
| 333 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); |
| 334 return -1; |
| 335 } |
| 336 |
| 337 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; |
| 338 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; |
| 339 pitch_range = s->max_pitch_val - s->min_pitch_val; |
| 340 s->pitch_nbits = av_ceil_log2(pitch_range); |
| 341 s->last_pitch_val = 40; |
| 342 s->last_acb_type = ACB_TYPE_NONE; |
| 343 s->history_nsamples = s->max_pitch_val + 8; |
| 344 |
| 345 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { |
| 346 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, |
| 347 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; |
| 348 |
| 349 av_log(ctx, AV_LOG_ERROR, |
| 350 "Unsupported samplerate %d (min=%d, max=%d)\n", |
| 351 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz |
| 352 |
| 353 return -1; |
| 354 } |
| 355 |
| 356 s->block_conv_table[0] = s->min_pitch_val; |
| 357 s->block_conv_table[1] = (pitch_range * 25) >> 6; |
| 358 s->block_conv_table[2] = (pitch_range * 44) >> 6; |
| 359 s->block_conv_table[3] = s->max_pitch_val - 1; |
| 360 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; |
| 361 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); |
| 362 s->block_pitch_range = s->block_conv_table[2] + |
| 363 s->block_conv_table[3] + 1 + |
| 364 2 * (s->block_conv_table[1] - 2 * s->min_pitch
_val); |
| 365 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); |
| 366 |
| 367 ctx->sample_fmt = SAMPLE_FMT_FLT; |
| 368 |
| 369 return 0; |
| 370 } |
| 371 |
| 372 /** |
| 373 * Dequantize LSPs |
| 374 * @param lsps output pointer to the array that will hold the LSPs |
| 375 * @param num number of LSPs to be dequantized |
| 376 * @param values quantized values, contains n_stages values |
| 377 * @param sizes range (i.e. max value) of each quantized value |
| 378 * @param n_stages number of dequantization runs |
| 379 * @param table dequantization table to be used |
| 380 * @param mul_q LSF multiplier |
| 381 * @param base_q base (lowest) LSF values |
| 382 */ |
| 383 static void dequant_lsps(double *lsps, int num, |
| 384 const uint16_t *values, |
| 385 const uint16_t *sizes, |
| 386 int n_stages, const uint8_t *table, |
| 387 const double *mul_q, |
| 388 const double *base_q) |
| 389 { |
| 390 int n, m; |
| 391 |
| 392 memset(lsps, 0, num * sizeof(*lsps)); |
| 393 for (n = 0; n < n_stages; n++) { |
| 394 const uint8_t *t_off = &table[values[n] * num]; |
| 395 double base = base_q[n], mul = mul_q[n]; |
| 396 |
| 397 for (m = 0; m < num; m++) |
| 398 lsps[m] += base + mul * t_off[m]; |
| 399 |
| 400 table += sizes[n] * num; |
| 401 } |
| 402 } |
| 403 |
| 404 /** |
| 405 * @defgroup lsp_dequant LSP dequantization routines |
| 406 * LSP dequantization routines, for 10/16LSPs and independent/residual coding. |
| 407 * @note we assume enough bits are available, caller should check. |
| 408 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; |
| 409 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. |
| 410 * @{ |
| 411 */ |
| 412 /** |
| 413 * Parse 10 independently-coded LSPs. |
| 414 */ |
| 415 static void dequant_lsp10i(GetBitContext *gb, double *lsps) |
| 416 { |
| 417 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; |
| 418 static const double mul_lsf[4] = { |
| 419 5.2187144800e-3, 1.4626986422e-3, |
| 420 9.6179549166e-4, 1.1325736225e-3 |
| 421 }; |
| 422 static const double base_lsf[4] = { |
| 423 M_PI * -2.15522e-1, M_PI * -6.1646e-2, |
| 424 M_PI * -3.3486e-2, M_PI * -5.7408e-2 |
| 425 }; |
| 426 uint16_t v[4]; |
| 427 |
| 428 v[0] = get_bits(gb, 8); |
| 429 v[1] = get_bits(gb, 6); |
| 430 v[2] = get_bits(gb, 5); |
| 431 v[3] = get_bits(gb, 5); |
| 432 |
| 433 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, |
| 434 mul_lsf, base_lsf); |
| 435 } |
| 436 |
| 437 /** |
| 438 * Parse 10 independently-coded LSPs, and then derive the tables to |
| 439 * generate LSPs for the other frames from them (residual coding). |
| 440 */ |
| 441 static void dequant_lsp10r(GetBitContext *gb, |
| 442 double *i_lsps, const double *old, |
| 443 double *a1, double *a2, int q_mode) |
| 444 { |
| 445 static const uint16_t vec_sizes[3] = { 128, 64, 64 }; |
| 446 static const double mul_lsf[3] = { |
| 447 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 |
| 448 }; |
| 449 static const double base_lsf[3] = { |
| 450 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 |
| 451 }; |
| 452 const float (*ipol_tab)[2][10] = q_mode ? |
| 453 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; |
| 454 uint16_t interpol, v[3]; |
| 455 int n; |
| 456 |
| 457 dequant_lsp10i(gb, i_lsps); |
| 458 |
| 459 interpol = get_bits(gb, 5); |
| 460 v[0] = get_bits(gb, 7); |
| 461 v[1] = get_bits(gb, 6); |
| 462 v[2] = get_bits(gb, 6); |
| 463 |
| 464 for (n = 0; n < 10; n++) { |
| 465 double delta = old[n] - i_lsps[n]; |
| 466 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; |
| 467 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; |
| 468 } |
| 469 |
| 470 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, |
| 471 mul_lsf, base_lsf); |
| 472 } |
| 473 |
| 474 /** |
| 475 * Parse 16 independently-coded LSPs. |
| 476 */ |
| 477 static void dequant_lsp16i(GetBitContext *gb, double *lsps) |
| 478 { |
| 479 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; |
| 480 static const double mul_lsf[5] = { |
| 481 3.3439586280e-3, 6.9908173703e-4, |
| 482 3.3216608306e-3, 1.0334960326e-3, |
| 483 3.1899104283e-3 |
| 484 }; |
| 485 static const double base_lsf[5] = { |
| 486 M_PI * -1.27576e-1, M_PI * -2.4292e-2, |
| 487 M_PI * -1.28094e-1, M_PI * -3.2128e-2, |
| 488 M_PI * -1.29816e-1 |
| 489 }; |
| 490 uint16_t v[5]; |
| 491 |
| 492 v[0] = get_bits(gb, 8); |
| 493 v[1] = get_bits(gb, 6); |
| 494 v[2] = get_bits(gb, 7); |
| 495 v[3] = get_bits(gb, 6); |
| 496 v[4] = get_bits(gb, 7); |
| 497 |
| 498 dequant_lsps( lsps, 5, v, vec_sizes, 2, |
| 499 wmavoice_dq_lsp16i1, mul_lsf, base_lsf); |
| 500 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, |
| 501 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); |
| 502 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, |
| 503 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); |
| 504 } |
| 505 |
| 506 /** |
| 507 * Parse 16 independently-coded LSPs, and then derive the tables to |
| 508 * generate LSPs for the other frames from them (residual coding). |
| 509 */ |
| 510 static void dequant_lsp16r(GetBitContext *gb, |
| 511 double *i_lsps, const double *old, |
| 512 double *a1, double *a2, int q_mode) |
| 513 { |
| 514 static const uint16_t vec_sizes[3] = { 128, 128, 128 }; |
| 515 static const double mul_lsf[3] = { |
| 516 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 |
| 517 }; |
| 518 static const double base_lsf[3] = { |
| 519 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 |
| 520 }; |
| 521 const float (*ipol_tab)[2][16] = q_mode ? |
| 522 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; |
| 523 uint16_t interpol, v[3]; |
| 524 int n; |
| 525 |
| 526 dequant_lsp16i(gb, i_lsps); |
| 527 |
| 528 interpol = get_bits(gb, 5); |
| 529 v[0] = get_bits(gb, 7); |
| 530 v[1] = get_bits(gb, 7); |
| 531 v[2] = get_bits(gb, 7); |
| 532 |
| 533 for (n = 0; n < 16; n++) { |
| 534 double delta = old[n] - i_lsps[n]; |
| 535 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; |
| 536 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; |
| 537 } |
| 538 |
| 539 dequant_lsps( a2, 10, v, vec_sizes, 1, |
| 540 wmavoice_dq_lsp16r1, mul_lsf, base_lsf); |
| 541 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, |
| 542 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); |
| 543 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, |
| 544 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); |
| 545 } |
| 546 |
| 547 /** |
| 548 * @} |
| 549 * @defgroup aw Pitch-adaptive window coding functions |
| 550 * The next few functions are for pitch-adaptive window coding. |
| 551 * @{ |
| 552 */ |
| 553 /** |
| 554 * Parse the offset of the first pitch-adaptive window pulses, and |
| 555 * the distribution of pulses between the two blocks in this frame. |
| 556 * @param s WMA Voice decoding context private data |
| 557 * @param gb bit I/O context |
| 558 * @param pitch pitch for each block in this frame |
| 559 */ |
| 560 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, |
| 561 const int *pitch) |
| 562 { |
| 563 static const int16_t start_offset[94] = { |
| 564 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, |
| 565 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, |
| 566 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, |
| 567 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, |
| 568 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, |
| 569 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, |
| 570 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, |
| 571 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 |
| 572 }; |
| 573 int bits, offset; |
| 574 |
| 575 /* position of pulse */ |
| 576 s->aw_idx_is_ext = 0; |
| 577 if ((bits = get_bits(gb, 6)) >= 54) { |
| 578 s->aw_idx_is_ext = 1; |
| 579 bits += (bits - 54) * 3 + get_bits(gb, 2); |
| 580 } |
| 581 |
| 582 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count |
| 583 * the distribution of the pulses in each block contained in this frame. */ |
| 584 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; |
| 585 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; |
| 586 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pit
ch[0]; |
| 587 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; |
| 588 offset += s->aw_n_pulses[0] * pitch[0]; |
| 589 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1
]; |
| 590 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; |
| 591 |
| 592 /* if continuing from a position before the block, reset position to |
| 593 * start of block (when corrected for the range over which it can be |
| 594 * spread in aw_pulse_set1()). */ |
| 595 if (start_offset[bits] < MAX_FRAMESIZE / 2) { |
| 596 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) |
| 597 s->aw_first_pulse_off[1] -= pitch[1]; |
| 598 if (start_offset[bits] < 0) |
| 599 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) |
| 600 s->aw_first_pulse_off[0] -= pitch[0]; |
| 601 } |
| 602 } |
| 603 |
| 604 /** |
| 605 * Apply second set of pitch-adaptive window pulses. |
| 606 * @param s WMA Voice decoding context private data |
| 607 * @param gb bit I/O context |
| 608 * @param block_idx block index in frame [0, 1] |
| 609 * @param fcb structure containing fixed codebook vector info |
| 610 */ |
| 611 static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
| 612 int block_idx, AMRFixed *fcb) |
| 613 { |
| 614 uint16_t use_mask[7]; // only 5 are used, rest is padding |
| 615 /* in this function, idx is the index in the 80-bit (+ padding) use_mask |
| 616 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits |
| 617 * of idx are the position of the bit within a particular item in the |
| 618 * array (0 being the most significant bit, and 15 being the least |
| 619 * significant bit), and the remainder (>> 4) is the index in the |
| 620 * use_mask[]-array. This is faster and uses less memory than using a |
| 621 * 80-byte/80-int array. */ |
| 622 int pulse_off = s->aw_first_pulse_off[block_idx], |
| 623 pulse_start, n, idx, range, aidx, start_off = 0; |
| 624 |
| 625 /* set offset of first pulse to within this block */ |
| 626 if (s->aw_n_pulses[block_idx] > 0) |
| 627 while (pulse_off + s->aw_pulse_range < 1) |
| 628 pulse_off += fcb->pitch_lag; |
| 629 |
| 630 /* find range per pulse */ |
| 631 if (s->aw_n_pulses[0] > 0) { |
| 632 if (block_idx == 0) { |
| 633 range = 32; |
| 634 } else /* block_idx = 1 */ { |
| 635 range = 8; |
| 636 if (s->aw_n_pulses[block_idx] > 0) |
| 637 pulse_off = s->aw_next_pulse_off_cache; |
| 638 } |
| 639 } else |
| 640 range = 16; |
| 641 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; |
| 642 |
| 643 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, |
| 644 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus |
| 645 * we exclude that range from being pulsed again in this function. */ |
| 646 memset( use_mask, -1, 5 * sizeof(use_mask[0])); |
| 647 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); |
| 648 if (s->aw_n_pulses[block_idx] > 0) |
| 649 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { |
| 650 int excl_range = s->aw_pulse_range; // always 16 or 24 |
| 651 uint16_t *use_mask_ptr = &use_mask[idx >> 4]; |
| 652 int first_sh = 16 - (idx & 15); |
| 653 *use_mask_ptr++ &= 0xFFFF << first_sh; |
| 654 excl_range -= first_sh; |
| 655 if (excl_range >= 16) { |
| 656 *use_mask_ptr++ = 0; |
| 657 *use_mask_ptr &= 0xFFFF >> (excl_range - 16); |
| 658 } else |
| 659 *use_mask_ptr &= 0xFFFF >> excl_range; |
| 660 } |
| 661 |
| 662 /* find the 'aidx'th offset that is not excluded */ |
| 663 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); |
| 664 for (n = 0; n <= aidx; pulse_start++) { |
| 665 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; |
| 666 if (idx >= MAX_FRAMESIZE / 2) { // find from zero |
| 667 if (use_mask[0]) idx = 0x0F; |
| 668 else if (use_mask[1]) idx = 0x1F; |
| 669 else if (use_mask[2]) idx = 0x2F; |
| 670 else if (use_mask[3]) idx = 0x3F; |
| 671 else if (use_mask[4]) idx = 0x4F; |
| 672 else return; |
| 673 idx -= av_log2_16bit(use_mask[idx >> 4]); |
| 674 } |
| 675 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { |
| 676 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); |
| 677 n++; |
| 678 start_off = idx; |
| 679 } |
| 680 } |
| 681 |
| 682 fcb->x[fcb->n] = start_off; |
| 683 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; |
| 684 fcb->n++; |
| 685 |
| 686 /* set offset for next block, relative to start of that block */ |
| 687 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; |
| 688 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; |
| 689 } |
| 690 |
| 691 /** |
| 692 * Apply first set of pitch-adaptive window pulses. |
| 693 * @param s WMA Voice decoding context private data |
| 694 * @param gb bit I/O context |
| 695 * @param block_idx block index in frame [0, 1] |
| 696 * @param fcb storage location for fixed codebook pulse info |
| 697 */ |
| 698 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, |
| 699 int block_idx, AMRFixed *fcb) |
| 700 { |
| 701 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); |
| 702 float v; |
| 703 |
| 704 if (s->aw_n_pulses[block_idx] > 0) { |
| 705 int n, v_mask, i_mask, sh, n_pulses; |
| 706 |
| 707 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each |
| 708 n_pulses = 3; |
| 709 v_mask = 8; |
| 710 i_mask = 7; |
| 711 sh = 4; |
| 712 } else { // 4 pulses, 1:sign + 2:index each |
| 713 n_pulses = 4; |
| 714 v_mask = 4; |
| 715 i_mask = 3; |
| 716 sh = 3; |
| 717 } |
| 718 |
| 719 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { |
| 720 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; |
| 721 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + |
| 722 s->aw_first_pulse_off[block_idx]; |
| 723 while (fcb->x[fcb->n] < 0) |
| 724 fcb->x[fcb->n] += fcb->pitch_lag; |
| 725 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) |
| 726 fcb->n++; |
| 727 } |
| 728 } else { |
| 729 int num2 = (val & 0x1FF) >> 1, delta, idx; |
| 730 |
| 731 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } |
| 732 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } |
| 733 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } |
| 734 else { delta = 7; idx = num2 + 1 - 3 * 75; } |
| 735 v = (val & 0x200) ? -1.0 : 1.0; |
| 736 |
| 737 fcb->no_repeat_mask |= 3 << fcb->n; |
| 738 fcb->x[fcb->n] = idx - delta; |
| 739 fcb->y[fcb->n] = v; |
| 740 fcb->x[fcb->n + 1] = idx; |
| 741 fcb->y[fcb->n + 1] = (val & 1) ? -v : v; |
| 742 fcb->n += 2; |
| 743 } |
| 744 } |
| 745 |
| 746 /** |
| 747 * @} |
| 748 * |
| 749 * Generate a random number from frame_cntr and block_idx, which will lief |
| 750 * in the range [0, 1000 - block_size] (so it can be used as an index in a |
| 751 * table of size 1000 of which you want to read block_size entries). |
| 752 * |
| 753 * @param frame_cntr current frame number |
| 754 * @param block_num current block index |
| 755 * @param block_size amount of entries we want to read from a table |
| 756 * that has 1000 entries |
| 757 * @returns a (non-)random number in the [0, 1000 - block_size] range. |
| 758 */ |
| 759 static int pRNG(int frame_cntr, int block_num, int block_size) |
| 760 { |
| 761 /* array to simplify the calculation of z: |
| 762 * y = (x % 9) * 5 + 6; |
| 763 * z = (49995 * x) / y; |
| 764 * Since y only has 9 values, we can remove the division by using a |
| 765 * LUT and using FASTDIV-style divisions. For each of the 9 values |
| 766 * of y, we can rewrite z as: |
| 767 * z = x * (49995 / y) + x * ((49995 % y) / y) |
| 768 * In this table, each col represents one possible value of y, the |
| 769 * first number is 49995 / y, and the second is the FASTDIV variant |
| 770 * of 49995 % y / y. */ |
| 771 static const unsigned int div_tbl[9][2] = { |
| 772 { 8332, 3 * 715827883U }, // y = 6 |
| 773 { 4545, 0 * 390451573U }, // y = 11 |
| 774 { 3124, 11 * 268435456U }, // y = 16 |
| 775 { 2380, 15 * 204522253U }, // y = 21 |
| 776 { 1922, 23 * 165191050U }, // y = 26 |
| 777 { 1612, 23 * 138547333U }, // y = 31 |
| 778 { 1388, 27 * 119304648U }, // y = 36 |
| 779 { 1219, 16 * 104755300U }, // y = 41 |
| 780 { 1086, 39 * 93368855U } // y = 46 |
| 781 }; |
| 782 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; |
| 783 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, |
| 784 // so this is effectively a modulo (%) |
| 785 y = x - 9 * MULH(477218589, x); // x % 9 |
| 786 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); |
| 787 // z = x * 49995 / (y * 5 + 6) |
| 788 return z % (1000 - block_size); |
| 789 } |
| 790 |
| 791 /** |
| 792 * Parse hardcoded signal for a single block. |
| 793 * @note see #synth_block(). |
| 794 */ |
| 795 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, |
| 796 int block_idx, int size, |
| 797 const struct frame_type_desc *frame_desc, |
| 798 float *excitation) |
| 799 { |
| 800 float gain; |
| 801 int n, r_idx; |
| 802 |
| 803 assert(size <= MAX_FRAMESIZE); |
| 804 |
| 805 /* Set the offset from which we start reading wmavoice_std_codebook */ |
| 806 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
| 807 r_idx = pRNG(s->frame_cntr, block_idx, size); |
| 808 gain = s->silence_gain; |
| 809 } else /* FCB_TYPE_HARDCODED */ { |
| 810 r_idx = get_bits(gb, 8); |
| 811 gain = wmavoice_gain_universal[get_bits(gb, 6)]; |
| 812 } |
| 813 |
| 814 /* Clear gain prediction parameters */ |
| 815 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); |
| 816 |
| 817 /* Apply gain to hardcoded codebook and use that as excitation signal */ |
| 818 for (n = 0; n < size; n++) |
| 819 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; |
| 820 } |
| 821 |
| 822 /** |
| 823 * Parse FCB/ACB signal for a single block. |
| 824 * @note see #synth_block(). |
| 825 */ |
| 826 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
| 827 int block_idx, int size, |
| 828 int block_pitch_sh2, |
| 829 const struct frame_type_desc *frame_desc, |
| 830 float *excitation) |
| 831 { |
| 832 static const float gain_coeff[6] = { |
| 833 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 |
| 834 }; |
| 835 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; |
| 836 int n, idx, gain_weight; |
| 837 AMRFixed fcb; |
| 838 |
| 839 assert(size <= MAX_FRAMESIZE / 2); |
| 840 memset(pulses, 0, sizeof(*pulses) * size); |
| 841 |
| 842 fcb.pitch_lag = block_pitch_sh2 >> 2; |
| 843 fcb.pitch_fac = 1.0; |
| 844 fcb.no_repeat_mask = 0; |
| 845 fcb.n = 0; |
| 846 |
| 847 /* For the other frame types, this is where we apply the innovation |
| 848 * (fixed) codebook pulses of the speech signal. */ |
| 849 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| 850 aw_pulse_set1(s, gb, block_idx, &fcb); |
| 851 aw_pulse_set2(s, gb, block_idx, &fcb); |
| 852 } else /* FCB_TYPE_EXC_PULSES */ { |
| 853 int offset_nbits = 5 - frame_desc->log_n_blocks; |
| 854 |
| 855 fcb.no_repeat_mask = -1; |
| 856 /* similar to ff_decode_10_pulses_35bits(), but with single pulses |
| 857 * (instead of double) for a subset of pulses */ |
| 858 for (n = 0; n < 5; n++) { |
| 859 float sign; |
| 860 int pos1, pos2; |
| 861 |
| 862 sign = get_bits1(gb) ? 1.0 : -1.0; |
| 863 pos1 = get_bits(gb, offset_nbits); |
| 864 fcb.x[fcb.n] = n + 5 * pos1; |
| 865 fcb.y[fcb.n++] = sign; |
| 866 if (n < frame_desc->dbl_pulses) { |
| 867 pos2 = get_bits(gb, offset_nbits); |
| 868 fcb.x[fcb.n] = n + 5 * pos2; |
| 869 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; |
| 870 } |
| 871 } |
| 872 } |
| 873 ff_set_fixed_vector(pulses, &fcb, 1.0, size); |
| 874 |
| 875 /* Calculate gain for adaptive & fixed codebook signal. |
| 876 * see ff_amr_set_fixed_gain(). */ |
| 877 idx = get_bits(gb, 7); |
| 878 fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) - |
| 879 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); |
| 880 acb_gain = wmavoice_gain_codebook_acb[idx]; |
| 881 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], |
| 882 -2.9957322736 /* log(0.05) */, |
| 883 1.6094379124 /* log(5.0) */); |
| 884 |
| 885 gain_weight = 8 >> frame_desc->log_n_blocks; |
| 886 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, |
| 887 sizeof(*s->gain_pred_err) * (6 - gain_weight)); |
| 888 for (n = 0; n < gain_weight; n++) |
| 889 s->gain_pred_err[n] = pred_err; |
| 890 |
| 891 /* Calculation of adaptive codebook */ |
| 892 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
| 893 int len; |
| 894 for (n = 0; n < size; n += len) { |
| 895 int next_idx_sh16; |
| 896 int abs_idx = block_idx * size + n; |
| 897 int pitch_sh16 = (s->last_pitch_val << 16) + |
| 898 s->pitch_diff_sh16 * abs_idx; |
| 899 int pitch = (pitch_sh16 + 0x6FFF) >> 16; |
| 900 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; |
| 901 idx = idx_sh16 >> 16; |
| 902 if (s->pitch_diff_sh16) { |
| 903 if (s->pitch_diff_sh16 > 0) { |
| 904 next_idx_sh16 = (idx_sh16) &~ 0xFFFF; |
| 905 } else |
| 906 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; |
| 907 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 /
8, |
| 908 1, size - n); |
| 909 } else |
| 910 len = size; |
| 911 |
| 912 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], |
| 913 wmavoice_ipol1_coeffs, 17, |
| 914 idx, 9, len); |
| 915 } |
| 916 } else /* ACB_TYPE_HAMMING */ { |
| 917 int block_pitch = block_pitch_sh2 >> 2; |
| 918 idx = block_pitch_sh2 & 3; |
| 919 if (idx) { |
| 920 ff_acelp_interpolatef(excitation, &excitation[-block_pitch], |
| 921 wmavoice_ipol2_coeffs, 4, |
| 922 idx, 8, size); |
| 923 } else |
| 924 av_memcpy_backptr(excitation, sizeof(float) * block_pitch, |
| 925 sizeof(float) * size); |
| 926 } |
| 927 |
| 928 /* Interpolate ACB/FCB and use as excitation signal */ |
| 929 ff_weighted_vector_sumf(excitation, excitation, pulses, |
| 930 acb_gain, fcb_gain, size); |
| 931 } |
| 932 |
| 933 /** |
| 934 * Parse data in a single block. |
| 935 * @note we assume enough bits are available, caller should check. |
| 936 * |
| 937 * @param s WMA Voice decoding context private data |
| 938 * @param gb bit I/O context |
| 939 * @param block_idx index of the to-be-read block |
| 940 * @param size amount of samples to be read in this block |
| 941 * @param block_pitch_sh2 pitch for this block << 2 |
| 942 * @param lsps LSPs for (the end of) this frame |
| 943 * @param prev_lsps LSPs for the last frame |
| 944 * @param frame_desc frame type descriptor |
| 945 * @param excitation target memory for the ACB+FCB interpolated signal |
| 946 * @param synth target memory for the speech synthesis filter output |
| 947 * @return 0 on success, <0 on error. |
| 948 */ |
| 949 static void synth_block(WMAVoiceContext *s, GetBitContext *gb, |
| 950 int block_idx, int size, |
| 951 int block_pitch_sh2, |
| 952 const double *lsps, const double *prev_lsps, |
| 953 const struct frame_type_desc *frame_desc, |
| 954 float *excitation, float *synth) |
| 955 { |
| 956 double i_lsps[MAX_LSPS]; |
| 957 float lpcs[MAX_LSPS]; |
| 958 float fac; |
| 959 int n; |
| 960 |
| 961 if (frame_desc->acb_type == ACB_TYPE_NONE) |
| 962 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); |
| 963 else |
| 964 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, |
| 965 frame_desc, excitation); |
| 966 |
| 967 /* convert interpolated LSPs to LPCs */ |
| 968 fac = (block_idx + 0.5) / frame_desc->n_blocks; |
| 969 for (n = 0; n < s->lsps; n++) // LSF -> LSP |
| 970 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); |
| 971 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); |
| 972 |
| 973 /* Speech synthesis */ |
| 974 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); |
| 975 } |
| 976 |
| 977 /** |
| 978 * Synthesize output samples for a single frame. |
| 979 * @note we assume enough bits are available, caller should check. |
| 980 * |
| 981 * @param ctx WMA Voice decoder context |
| 982 * @param gb bit I/O context (s->gb or one for cross-packet superframes) |
| 983 * @param samples pointer to output sample buffer, has space for at least 160 |
| 984 * samples |
| 985 * @param lsps LSP array |
| 986 * @param prev_lsps array of previous frame's LSPs |
| 987 * @param excitation target buffer for excitation signal |
| 988 * @param synth target buffer for synthesized speech data |
| 989 * @return 0 on success, <0 on error. |
| 990 */ |
| 991 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, |
| 992 float *samples, |
| 993 const double *lsps, const double *prev_lsps, |
| 994 float *excitation, float *synth) |
| 995 { |
| 996 WMAVoiceContext *s = ctx->priv_data; |
| 997 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val; |
| 998 int pitch[MAX_BLOCKS], last_block_pitch; |
| 999 |
| 1000 /* Parse frame type ("frame header"), see frame_descs */ |
| 1001 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], |
| 1002 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; |
| 1003 |
| 1004 if (bd_idx < 0) { |
| 1005 av_log(ctx, AV_LOG_ERROR, |
| 1006 "Invalid frame type VLC code, skipping\n"); |
| 1007 return -1; |
| 1008 } |
| 1009 |
| 1010 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ |
| 1011 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { |
| 1012 /* Pitch is provided per frame, which is interpreted as the pitch of |
| 1013 * the last sample of the last block of this frame. We can interpolate |
| 1014 * the pitch of other blocks (and even pitch-per-sample) by gradually |
| 1015 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ |
| 1016 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; |
| 1017 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; |
| 1018 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); |
| 1019 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); |
| 1020 if (s->last_acb_type == ACB_TYPE_NONE || |
| 1021 20 * abs(cur_pitch_val - s->last_pitch_val) > |
| 1022 (cur_pitch_val + s->last_pitch_val)) |
| 1023 s->last_pitch_val = cur_pitch_val; |
| 1024 |
| 1025 /* pitch per block */ |
| 1026 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
| 1027 int fac = n * 2 + 1; |
| 1028 |
| 1029 pitch[n] = (MUL16(fac, cur_pitch_val) + |
| 1030 MUL16((n_blocks_x2 - fac), s->last_pitch_val) + |
| 1031 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; |
| 1032 } |
| 1033 |
| 1034 /* "pitch-diff-per-sample" for calculation of pitch per sample */ |
| 1035 s->pitch_diff_sh16 = |
| 1036 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; |
| 1037 } |
| 1038 |
| 1039 /* Global gain (if silence) and pitch-adaptive window coordinates */ |
| 1040 switch (frame_descs[bd_idx].fcb_type) { |
| 1041 case FCB_TYPE_SILENCE: |
| 1042 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; |
| 1043 break; |
| 1044 case FCB_TYPE_AW_PULSES: |
| 1045 aw_parse_coords(s, gb, pitch); |
| 1046 break; |
| 1047 } |
| 1048 |
| 1049 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
| 1050 int bl_pitch_sh2; |
| 1051 |
| 1052 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ |
| 1053 switch (frame_descs[bd_idx].acb_type) { |
| 1054 case ACB_TYPE_HAMMING: { |
| 1055 /* Pitch is given per block. Per-block pitches are encoded as an |
| 1056 * absolute value for the first block, and then delta values |
| 1057 * relative to this value) for all subsequent blocks. The scale of |
| 1058 * this pitch value is semi-logaritmic compared to its use in the |
| 1059 * decoder, so we convert it to normal scale also. */ |
| 1060 int block_pitch, |
| 1061 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, |
| 1062 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, |
| 1063 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; |
| 1064 |
| 1065 if (n == 0) { |
| 1066 block_pitch = get_bits(gb, s->block_pitch_nbits); |
| 1067 } else |
| 1068 block_pitch = last_block_pitch - s->block_delta_pitch_hrange + |
| 1069 get_bits(gb, s->block_delta_pitch_nbits); |
| 1070 /* Convert last_ so that any next delta is within _range */ |
| 1071 last_block_pitch = av_clip(block_pitch, |
| 1072 s->block_delta_pitch_hrange, |
| 1073 s->block_pitch_range - |
| 1074 s->block_delta_pitch_hrange); |
| 1075 |
| 1076 /* Convert semi-log-style scale back to normal scale */ |
| 1077 if (block_pitch < t1) { |
| 1078 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; |
| 1079 } else { |
| 1080 block_pitch -= t1; |
| 1081 if (block_pitch < t2) { |
| 1082 bl_pitch_sh2 = |
| 1083 (s->block_conv_table[1] << 2) + (block_pitch << 1); |
| 1084 } else { |
| 1085 block_pitch -= t2; |
| 1086 if (block_pitch < t3) { |
| 1087 bl_pitch_sh2 = |
| 1088 (s->block_conv_table[2] + block_pitch) << 2; |
| 1089 } else |
| 1090 bl_pitch_sh2 = s->block_conv_table[3] << 2; |
| 1091 } |
| 1092 } |
| 1093 pitch[n] = bl_pitch_sh2 >> 2; |
| 1094 break; |
| 1095 } |
| 1096 |
| 1097 case ACB_TYPE_ASYMMETRIC: { |
| 1098 bl_pitch_sh2 = pitch[n] << 2; |
| 1099 break; |
| 1100 } |
| 1101 |
| 1102 default: // ACB_TYPE_NONE has no pitch |
| 1103 bl_pitch_sh2 = 0; |
| 1104 break; |
| 1105 } |
| 1106 |
| 1107 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, |
| 1108 lsps, prev_lsps, &frame_descs[bd_idx], |
| 1109 &excitation[n * block_nsamples], |
| 1110 &synth[n * block_nsamples]); |
| 1111 } |
| 1112 |
| 1113 /* Averaging projection filter, if applicable. Else, just copy samples |
| 1114 * from synthesis buffer */ |
| 1115 if (s->do_apf) { |
| 1116 // FIXME this is where APF would take place, currently not implemented |
| 1117 av_log_missing_feature(ctx, "APF", 0); |
| 1118 s->do_apf = 0; |
| 1119 } //else |
| 1120 for (n = 0; n < 160; n++) |
| 1121 samples[n] = av_clipf(synth[n], -1.0, 1.0); |
| 1122 |
| 1123 /* Cache values for next frame */ |
| 1124 s->frame_cntr++; |
| 1125 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) |
| 1126 s->last_acb_type = frame_descs[bd_idx].acb_type; |
| 1127 switch (frame_descs[bd_idx].acb_type) { |
| 1128 case ACB_TYPE_NONE: |
| 1129 s->last_pitch_val = 0; |
| 1130 break; |
| 1131 case ACB_TYPE_ASYMMETRIC: |
| 1132 s->last_pitch_val = cur_pitch_val; |
| 1133 break; |
| 1134 case ACB_TYPE_HAMMING: |
| 1135 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; |
| 1136 break; |
| 1137 } |
| 1138 |
| 1139 return 0; |
| 1140 } |
| 1141 |
| 1142 /** |
| 1143 * Ensure minimum value for first item, maximum value for last value, |
| 1144 * proper spacing between each value and proper ordering. |
| 1145 * |
| 1146 * @param lsps array of LSPs |
| 1147 * @param num size of LSP array |
| 1148 * |
| 1149 * @note basically a double version of #ff_acelp_reorder_lsf(), might be |
| 1150 * useful to put in a generic location later on. Parts are also |
| 1151 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), |
| 1152 * which is in float. |
| 1153 */ |
| 1154 static void stabilize_lsps(double *lsps, int num) |
| 1155 { |
| 1156 int n, m, l; |
| 1157 |
| 1158 /* set minimum value for first, maximum value for last and minimum |
| 1159 * spacing between LSF values. |
| 1160 * Very similar to ff_set_min_dist_lsf(), but in double. */ |
| 1161 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); |
| 1162 for (n = 1; n < num; n++) |
| 1163 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); |
| 1164 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); |
| 1165 |
| 1166 /* reorder (looks like one-time / non-recursed bubblesort). |
| 1167 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ |
| 1168 for (n = 1; n < num; n++) { |
| 1169 if (lsps[n] < lsps[n - 1]) { |
| 1170 for (m = 1; m < num; m++) { |
| 1171 double tmp = lsps[m]; |
| 1172 for (l = m - 1; l >= 0; l--) { |
| 1173 if (lsps[l] <= tmp) break; |
| 1174 lsps[l + 1] = lsps[l]; |
| 1175 } |
| 1176 lsps[l + 1] = tmp; |
| 1177 } |
| 1178 break; |
| 1179 } |
| 1180 } |
| 1181 } |
| 1182 |
| 1183 /** |
| 1184 * Test if there's enough bits to read 1 superframe. |
| 1185 * |
| 1186 * @param orig_gb bit I/O context used for reading. This function |
| 1187 * does not modify the state of the bitreader; it |
| 1188 * only uses it to copy the current stream position |
| 1189 * @param s WMA Voice decoding context private data |
| 1190 * @returns -1 if unsupported, 1 on not enough bits or 0 if OK. |
| 1191 */ |
| 1192 static int check_bits_for_superframe(GetBitContext *orig_gb, |
| 1193 WMAVoiceContext *s) |
| 1194 { |
| 1195 GetBitContext s_gb, *gb = &s_gb; |
| 1196 int n, need_bits, bd_idx; |
| 1197 const struct frame_type_desc *frame_desc; |
| 1198 |
| 1199 /* initialize a copy */ |
| 1200 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); |
| 1201 skip_bits_long(gb, get_bits_count(orig_gb)); |
| 1202 assert(get_bits_left(gb) == get_bits_left(orig_gb)); |
| 1203 |
| 1204 /* superframe header */ |
| 1205 if (get_bits_left(gb) < 14) |
| 1206 return 1; |
| 1207 if (!get_bits1(gb)) |
| 1208 return -1; // WMAPro-in-WMAVoice superframe |
| 1209 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe |
| 1210 if (s->has_residual_lsps) { // residual LSPs (for all frames) |
| 1211 if (get_bits_left(gb) < s->sframe_lsp_bitsize) |
| 1212 return 1; |
| 1213 skip_bits_long(gb, s->sframe_lsp_bitsize); |
| 1214 } |
| 1215 |
| 1216 /* frames */ |
| 1217 for (n = 0; n < MAX_FRAMES; n++) { |
| 1218 int aw_idx_is_ext = 0; |
| 1219 |
| 1220 if (!s->has_residual_lsps) { // independent LSPs (per-frame) |
| 1221 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; |
| 1222 skip_bits_long(gb, s->frame_lsp_bitsize); |
| 1223 } |
| 1224 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; |
| 1225 if (bd_idx < 0) |
| 1226 return -1; // invalid frame type VLC code |
| 1227 frame_desc = &frame_descs[bd_idx]; |
| 1228 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
| 1229 if (get_bits_left(gb) < s->pitch_nbits) |
| 1230 return 1; |
| 1231 skip_bits_long(gb, s->pitch_nbits); |
| 1232 } |
| 1233 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
| 1234 skip_bits(gb, 8); |
| 1235 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| 1236 int tmp = get_bits(gb, 6); |
| 1237 if (tmp >= 0x36) { |
| 1238 skip_bits(gb, 2); |
| 1239 aw_idx_is_ext = 1; |
| 1240 } |
| 1241 } |
| 1242 |
| 1243 /* blocks */ |
| 1244 if (frame_desc->acb_type == ACB_TYPE_HAMMING) { |
| 1245 need_bits = s->block_pitch_nbits + |
| 1246 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; |
| 1247 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| 1248 need_bits = 2 * !aw_idx_is_ext; |
| 1249 } else |
| 1250 need_bits = 0; |
| 1251 need_bits += frame_desc->frame_size; |
| 1252 if (get_bits_left(gb) < need_bits) |
| 1253 return 1; |
| 1254 skip_bits_long(gb, need_bits); |
| 1255 } |
| 1256 |
| 1257 return 0; |
| 1258 } |
| 1259 |
| 1260 /** |
| 1261 * Synthesize output samples for a single superframe. If we have any data |
| 1262 * cached in s->sframe_cache, that will be used instead of whatever is loaded |
| 1263 * in s->gb. |
| 1264 * |
| 1265 * WMA Voice superframes contain 3 frames, each containing 160 audio samples, |
| 1266 * to give a total of 480 samples per frame. See #synth_frame() for frame |
| 1267 * parsing. In addition to 3 frames, superframes can also contain the LSPs |
| 1268 * (if these are globally specified for all frames (residually); they can |
| 1269 * also be specified individually per-frame. See the s->has_residual_lsps |
| 1270 * option), and can specify the number of samples encoded in this superframe |
| 1271 * (if less than 480), usually used to prevent blanks at track boundaries. |
| 1272 * |
| 1273 * @param ctx WMA Voice decoder context |
| 1274 * @param samples pointer to output buffer for voice samples |
| 1275 * @param data_size pointer containing the size of #samples on input, and the |
| 1276 * amount of #samples filled on output |
| 1277 * @return 0 on success, <0 on error or 1 if there was not enough data to |
| 1278 * fully parse the superframe |
| 1279 */ |
| 1280 static int synth_superframe(AVCodecContext *ctx, |
| 1281 float *samples, int *data_size) |
| 1282 { |
| 1283 WMAVoiceContext *s = ctx->priv_data; |
| 1284 GetBitContext *gb = &s->gb, s_gb; |
| 1285 int n, res, n_samples = 480; |
| 1286 double lsps[MAX_FRAMES][MAX_LSPS]; |
| 1287 const double *mean_lsf = s->lsps == 16 ? |
| 1288 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mo
de]; |
| 1289 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; |
| 1290 float synth[MAX_LSPS + MAX_SFRAMESIZE]; |
| 1291 |
| 1292 memcpy(synth, s->synth_history, |
| 1293 s->lsps * sizeof(*synth)); |
| 1294 memcpy(excitation, s->excitation_history, |
| 1295 s->history_nsamples * sizeof(*excitation)); |
| 1296 |
| 1297 if (s->sframe_cache_size > 0) { |
| 1298 gb = &s_gb; |
| 1299 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); |
| 1300 s->sframe_cache_size = 0; |
| 1301 } |
| 1302 |
| 1303 if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; |
| 1304 |
| 1305 /* First bit is speech/music bit, it differentiates between WMAVoice |
| 1306 * speech samples (the actual codec) and WMAVoice music samples, which |
| 1307 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in |
| 1308 * the wild yet. */ |
| 1309 if (!get_bits1(gb)) { |
| 1310 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1); |
| 1311 return -1; |
| 1312 } |
| 1313 |
| 1314 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ |
| 1315 if (get_bits1(gb)) { |
| 1316 if ((n_samples = get_bits(gb, 12)) > 480) { |
| 1317 av_log(ctx, AV_LOG_ERROR, |
| 1318 "Superframe encodes >480 samples (%d), not allowed\n", |
| 1319 n_samples); |
| 1320 return -1; |
| 1321 } |
| 1322 } |
| 1323 /* Parse LSPs, if global for the superframe (can also be per-frame). */ |
| 1324 if (s->has_residual_lsps) { |
| 1325 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; |
| 1326 |
| 1327 for (n = 0; n < s->lsps; n++) |
| 1328 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; |
| 1329 |
| 1330 if (s->lsps == 10) { |
| 1331 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
| 1332 } else /* s->lsps == 16 */ |
| 1333 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
| 1334 |
| 1335 for (n = 0; n < s->lsps; n++) { |
| 1336 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); |
| 1337 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); |
| 1338 lsps[2][n] += mean_lsf[n]; |
| 1339 } |
| 1340 for (n = 0; n < 3; n++) |
| 1341 stabilize_lsps(lsps[n], s->lsps); |
| 1342 } |
| 1343 |
| 1344 /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ |
| 1345 for (n = 0; n < 3; n++) { |
| 1346 if (!s->has_residual_lsps) { |
| 1347 int m; |
| 1348 |
| 1349 if (s->lsps == 10) { |
| 1350 dequant_lsp10i(gb, lsps[n]); |
| 1351 } else /* s->lsps == 16 */ |
| 1352 dequant_lsp16i(gb, lsps[n]); |
| 1353 |
| 1354 for (m = 0; m < s->lsps; m++) |
| 1355 lsps[n][m] += mean_lsf[m]; |
| 1356 stabilize_lsps(lsps[n], s->lsps); |
| 1357 } |
| 1358 |
| 1359 if ((res = synth_frame(ctx, gb, |
| 1360 &samples[n * MAX_FRAMESIZE], |
| 1361 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], |
| 1362 &excitation[s->history_nsamples + n * MAX_FRAMESI
ZE], |
| 1363 &synth[s->lsps + n * MAX_FRAMESIZE]))) |
| 1364 return res; |
| 1365 } |
| 1366 |
| 1367 /* Statistics? FIXME - we don't check for length, a slight overrun |
| 1368 * will be caught by internal buffer padding, and anything else |
| 1369 * will be skipped, not read. */ |
| 1370 if (get_bits1(gb)) { |
| 1371 res = get_bits(gb, 4); |
| 1372 skip_bits(gb, 10 * (res + 1)); |
| 1373 } |
| 1374 |
| 1375 /* Specify nr. of output samples */ |
| 1376 *data_size = n_samples * sizeof(float); |
| 1377 |
| 1378 /* Update history */ |
| 1379 memcpy(s->prev_lsps, lsps[2], |
| 1380 s->lsps * sizeof(*s->prev_lsps)); |
| 1381 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], |
| 1382 s->lsps * sizeof(*synth)); |
| 1383 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], |
| 1384 s->history_nsamples * sizeof(*excitation)); |
| 1385 |
| 1386 return 0; |
| 1387 } |
| 1388 |
| 1389 /** |
| 1390 * Parse the packet header at the start of each packet (input data to this |
| 1391 * decoder). |
| 1392 * |
| 1393 * @param s WMA Voice decoding context private data |
| 1394 * @returns 1 if not enough bits were available, or 0 on success. |
| 1395 */ |
| 1396 static int parse_packet_header(WMAVoiceContext *s) |
| 1397 { |
| 1398 GetBitContext *gb = &s->gb; |
| 1399 unsigned int res; |
| 1400 |
| 1401 if (get_bits_left(gb) < 11) |
| 1402 return 1; |
| 1403 skip_bits(gb, 4); // packet sequence number |
| 1404 s->has_residual_lsps = get_bits1(gb); |
| 1405 do { |
| 1406 res = get_bits(gb, 6); // number of superframes per packet |
| 1407 // (minus first one if there is spillover) |
| 1408 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) |
| 1409 return 1; |
| 1410 } while (res == 0x3F); |
| 1411 s->spillover_nbits = get_bits(gb, s->spillover_bitsize); |
| 1412 |
| 1413 return 0; |
| 1414 } |
| 1415 |
| 1416 /** |
| 1417 * Copy (unaligned) bits from gb/data/size to pb. |
| 1418 * |
| 1419 * @param pb target buffer to copy bits into |
| 1420 * @param data source buffer to copy bits from |
| 1421 * @param size size of the source data, in bytes |
| 1422 * @param gb bit I/O context specifying the current position in the source. |
| 1423 * data. This function might use this to align the bit position to |
| 1424 * a whole-byte boundary before calling #ff_copy_bits() on aligned |
| 1425 * source data |
| 1426 * @param nbits the amount of bits to copy from source to target |
| 1427 * |
| 1428 * @note after calling this function, the current position in the input bit |
| 1429 * I/O context is undefined. |
| 1430 */ |
| 1431 static void copy_bits(PutBitContext *pb, |
| 1432 const uint8_t *data, int size, |
| 1433 GetBitContext *gb, int nbits) |
| 1434 { |
| 1435 int rmn_bytes, rmn_bits; |
| 1436 |
| 1437 rmn_bits = rmn_bytes = get_bits_left(gb); |
| 1438 if (rmn_bits < nbits) |
| 1439 return; |
| 1440 rmn_bits &= 7; rmn_bytes >>= 3; |
| 1441 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) |
| 1442 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); |
| 1443 ff_copy_bits(pb, data + size - rmn_bytes, |
| 1444 FFMIN(nbits - rmn_bits, rmn_bytes << 3)); |
| 1445 } |
| 1446 |
| 1447 /** |
| 1448 * Packet decoding: a packet is anything that the (ASF) demuxer contains, |
| 1449 * and we expect that the demuxer / application provides it to us as such |
| 1450 * (else you'll probably get garbage as output). Every packet has a size of |
| 1451 * ctx->block_align bytes, starts with a packet header (see |
| 1452 * #parse_packet_header()), and then a series of superframes. Superframe |
| 1453 * boundaries may exceed packets, i.e. superframes can split data over |
| 1454 * multiple (two) packets. |
| 1455 * |
| 1456 * For more information about frames, see #synth_superframe(). |
| 1457 */ |
| 1458 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, |
| 1459 int *data_size, AVPacket *avpkt) |
| 1460 { |
| 1461 WMAVoiceContext *s = ctx->priv_data; |
| 1462 GetBitContext *gb = &s->gb; |
| 1463 int size, res, pos; |
| 1464 |
| 1465 if (*data_size < 480 * sizeof(float)) { |
| 1466 av_log(ctx, AV_LOG_ERROR, |
| 1467 "Output buffer too small (%d given - %lu needed)\n", |
| 1468 *data_size, 480 * sizeof(float)); |
| 1469 return -1; |
| 1470 } |
| 1471 *data_size = 0; |
| 1472 |
| 1473 /* Packets are sometimes a multiple of ctx->block_align, with a packet |
| 1474 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer |
| 1475 * feeds us ASF packets, which may concatenate multiple "codec" packets |
| 1476 * in a single "muxer" packet, so we artificially emulate that by |
| 1477 * capping the packet size at ctx->block_align. */ |
| 1478 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); |
| 1479 if (!size) |
| 1480 return 0; |
| 1481 init_get_bits(&s->gb, avpkt->data, size << 3); |
| 1482 |
| 1483 /* size == ctx->block_align is used to indicate whether we are dealing with |
| 1484 * a new packet or a packet of which we already read the packet header |
| 1485 * previously. */ |
| 1486 if (size == ctx->block_align) { // new packet header |
| 1487 if ((res = parse_packet_header(s)) < 0) |
| 1488 return res; |
| 1489 |
| 1490 /* If the packet header specifies a s->spillover_nbits, then we want |
| 1491 * to push out all data of the previous packet (+ spillover) before |
| 1492 * continuing to parse new superframes in the current packet. */ |
| 1493 if (s->spillover_nbits > 0) { |
| 1494 if (s->sframe_cache_size > 0) { |
| 1495 int cnt = get_bits_count(gb); |
| 1496 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); |
| 1497 flush_put_bits(&s->pb); |
| 1498 s->sframe_cache_size += s->spillover_nbits; |
| 1499 if ((res = synth_superframe(ctx, data, data_size)) == 0 && |
| 1500 *data_size > 0) { |
| 1501 cnt += s->spillover_nbits; |
| 1502 s->skip_bits_next = cnt & 7; |
| 1503 return cnt >> 3; |
| 1504 } else |
| 1505 skip_bits_long (gb, s->spillover_nbits - cnt + |
| 1506 get_bits_count(gb)); // resync |
| 1507 } else |
| 1508 skip_bits_long(gb, s->spillover_nbits); // resync |
| 1509 } |
| 1510 } else if (s->skip_bits_next) |
| 1511 skip_bits(gb, s->skip_bits_next); |
| 1512 |
| 1513 /* Try parsing superframes in current packet */ |
| 1514 s->sframe_cache_size = 0; |
| 1515 s->skip_bits_next = 0; |
| 1516 pos = get_bits_left(gb); |
| 1517 if ((res = synth_superframe(ctx, data, data_size)) < 0) { |
| 1518 return res; |
| 1519 } else if (*data_size > 0) { |
| 1520 int cnt = get_bits_count(gb); |
| 1521 s->skip_bits_next = cnt & 7; |
| 1522 return cnt >> 3; |
| 1523 } else if ((s->sframe_cache_size = pos) > 0) { |
| 1524 /* rewind bit reader to start of last (incomplete) superframe... */ |
| 1525 init_get_bits(gb, avpkt->data, size << 3); |
| 1526 skip_bits_long(gb, (size << 3) - pos); |
| 1527 assert(get_bits_left(gb) == pos); |
| 1528 |
| 1529 /* ...and cache it for spillover in next packet */ |
| 1530 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); |
| 1531 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); |
| 1532 // FIXME bad - just copy bytes as whole and add use the |
| 1533 // skip_bits_next field |
| 1534 } |
| 1535 |
| 1536 return size; |
| 1537 } |
| 1538 |
| 1539 static av_cold void wmavoice_flush(AVCodecContext *ctx) |
| 1540 { |
| 1541 WMAVoiceContext *s = ctx->priv_data; |
| 1542 int n; |
| 1543 |
| 1544 s->sframe_cache_size = 0; |
| 1545 s->skip_bits_next = 0; |
| 1546 for (n = 0; n < s->lsps; n++) |
| 1547 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); |
| 1548 memset(s->excitation_history, 0, |
| 1549 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); |
| 1550 memset(s->synth_history, 0, |
| 1551 sizeof(*s->synth_history) * MAX_LSPS); |
| 1552 memset(s->gain_pred_err, 0, |
| 1553 sizeof(s->gain_pred_err)); |
| 1554 } |
| 1555 |
| 1556 AVCodec wmavoice_decoder = { |
| 1557 "wmavoice", |
| 1558 CODEC_TYPE_AUDIO, |
| 1559 CODEC_ID_WMAVOICE, |
| 1560 sizeof(WMAVoiceContext), |
| 1561 wmavoice_decode_init, |
| 1562 NULL, |
| 1563 NULL, |
| 1564 wmavoice_decode_packet, |
| 1565 CODEC_CAP_SUBFRAMES, |
| 1566 .flush = wmavoice_flush, |
| 1567 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), |
| 1568 }; |
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