| 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.h" |
| 33 | #include "avcodec.h" |
| 34 | #include "internal.h" |
| 35 | #include "get_bits.h" |
| 36 | #include "put_bits.h" |
| 37 | #include "wmavoice_data.h" |
| 38 | #include "celp_filters.h" |
| 39 | #include "acelp_vectors.h" |
| 40 | #include "acelp_filters.h" |
| 41 | #include "lsp.h" |
| 42 | #include "dct.h" |
| 43 | #include "rdft.h" |
| 44 | #include "sinewin.h" |
| 45 | |
| 46 | #define MAX_BLOCKS 8 ///< maximum number of blocks per frame |
| 47 | #define MAX_LSPS 16 ///< maximum filter order |
| 48 | #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple |
| 49 | ///< of 16 for ASM input buffer alignment |
| 50 | #define MAX_FRAMES 3 ///< maximum number of frames per superframe |
| 51 | #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame |
| 52 | #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history |
| 53 | #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) |
| 54 | ///< maximum number of samples per superframe |
| 55 | #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that |
| 56 | ///< was split over two packets |
| 57 | #define VLC_NBITS 6 ///< number of bits to read per VLC iteration |
| 58 | |
| 59 | /** |
| 60 | * Frame type VLC coding. |
| 61 | */ |
| 62 | static VLC frame_type_vlc; |
| 63 | |
| 64 | /** |
| 65 | * Adaptive codebook types. |
| 66 | */ |
| 67 | enum { |
| 68 | ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) |
| 69 | ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which |
| 70 | ///< we interpolate to get a per-sample pitch. |
| 71 | ///< Signal is generated using an asymmetric sinc |
| 72 | ///< window function |
| 73 | ///< @note see #wmavoice_ipol1_coeffs |
| 74 | ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using |
| 75 | ///< a Hamming sinc window function |
| 76 | ///< @note see #wmavoice_ipol2_coeffs |
| 77 | }; |
| 78 | |
| 79 | /** |
| 80 | * Fixed codebook types. |
| 81 | */ |
| 82 | enum { |
| 83 | FCB_TYPE_SILENCE = 0, ///< comfort noise during silence |
| 84 | ///< generated from a hardcoded (fixed) codebook |
| 85 | ///< with per-frame (low) gain values |
| 86 | FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block |
| 87 | ///< gain values |
| 88 | FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, |
| 89 | ///< used in particular for low-bitrate streams |
| 90 | FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in |
| 91 | ///< combinations of either single pulses or |
| 92 | ///< pulse pairs |
| 93 | }; |
| 94 | |
| 95 | /** |
| 96 | * Description of frame types. |
| 97 | */ |
| 98 | static const struct frame_type_desc { |
| 99 | uint8_t n_blocks; ///< amount of blocks per frame (each block |
| 100 | ///< (contains 160/#n_blocks samples) |
| 101 | uint8_t log_n_blocks; ///< log2(#n_blocks) |
| 102 | uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) |
| 103 | uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) |
| 104 | uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs |
| 105 | ///< (rather than just one single pulse) |
| 106 | ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES |
| 107 | uint16_t frame_size; ///< the amount of bits that make up the block |
| 108 | ///< data (per frame) |
| 109 | } frame_descs[17] = { |
| 110 | { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, |
| 111 | { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, |
| 112 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, |
| 113 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, |
| 114 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, |
| 115 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, |
| 116 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, |
| 117 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, |
| 118 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, |
| 119 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, |
| 120 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, |
| 121 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, |
| 122 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, |
| 123 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, |
| 124 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, |
| 125 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, |
| 126 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } |
| 127 | }; |
| 128 | |
| 129 | /** |
| 130 | * WMA Voice decoding context. |
| 131 | */ |
| 132 | typedef struct { |
| 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 | int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded |
| 164 | ///< per-frame (independent coding) |
| 165 | int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded |
| 166 | ///< per superframe (residual coding) |
| 167 | |
| 168 | int min_pitch_val; ///< base value for pitch parsing code |
| 169 | int max_pitch_val; ///< max value + 1 for pitch parsing |
| 170 | int pitch_nbits; ///< number of bits used to specify the |
| 171 | ///< pitch value in the frame header |
| 172 | int block_pitch_nbits; ///< number of bits used to specify the |
| 173 | ///< first block's pitch value |
| 174 | int block_pitch_range; ///< range of the block pitch |
| 175 | int block_delta_pitch_nbits; ///< number of bits used to specify the |
| 176 | ///< delta pitch between this and the last |
| 177 | ///< block's pitch value, used in all but |
| 178 | ///< first block |
| 179 | int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is |
| 180 | ///< from -this to +this-1) |
| 181 | uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale |
| 182 | ///< conversion |
| 183 | |
| 184 | /** |
| 185 | * @} |
| 186 | * |
| 187 | * @name Packet values specified in the packet header or related to a packet. |
| 188 | * |
| 189 | * A packet is considered to be a single unit of data provided to this |
| 190 | * decoder by the demuxer. |
| 191 | * @{ |
| 192 | */ |
| 193 | int spillover_nbits; ///< number of bits of the previous packet's |
| 194 | ///< last superframe preceding this |
| 195 | ///< packet's first full superframe (useful |
| 196 | ///< for re-synchronization also) |
| 197 | int has_residual_lsps; ///< if set, superframes contain one set of |
| 198 | ///< LSPs that cover all frames, encoded as |
| 199 | ///< independent and residual LSPs; if not |
| 200 | ///< set, each frame contains its own, fully |
| 201 | ///< independent, LSPs |
| 202 | int skip_bits_next; ///< number of bits to skip at the next call |
| 203 | ///< to #wmavoice_decode_packet() (since |
| 204 | ///< they're part of the previous superframe) |
| 205 | |
| 206 | uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE]; |
| 207 | ///< cache for superframe data split over |
| 208 | ///< multiple packets |
| 209 | int sframe_cache_size; ///< set to >0 if we have data from an |
| 210 | ///< (incomplete) superframe from a previous |
| 211 | ///< packet that spilled over in the current |
| 212 | ///< packet; specifies the amount of bits in |
| 213 | ///< #sframe_cache |
| 214 | PutBitContext pb; ///< bitstream writer for #sframe_cache |
| 215 | |
| 216 | /** |
| 217 | * @} |
| 218 | * |
| 219 | * @name Frame and superframe values |
| 220 | * Superframe and frame data - these can change from frame to frame, |
| 221 | * although some of them do in that case serve as a cache / history for |
| 222 | * the next frame or superframe. |
| 223 | * @{ |
| 224 | */ |
| 225 | double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous |
| 226 | ///< superframe |
| 227 | int last_pitch_val; ///< pitch value of the previous frame |
| 228 | int last_acb_type; ///< frame type [0-2] of the previous frame |
| 229 | int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) |
| 230 | ///< << 16) / #MAX_FRAMESIZE |
| 231 | float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE |
| 232 | |
| 233 | int aw_idx_is_ext; ///< whether the AW index was encoded in |
| 234 | ///< 8 bits (instead of 6) |
| 235 | int aw_pulse_range; ///< the range over which #aw_pulse_set1() |
| 236 | ///< can apply the pulse, relative to the |
| 237 | ///< value in aw_first_pulse_off. The exact |
| 238 | ///< position of the first AW-pulse is within |
| 239 | ///< [pulse_off, pulse_off + this], and |
| 240 | ///< depends on bitstream values; [16 or 24] |
| 241 | int aw_n_pulses[2]; ///< number of AW-pulses in each block; note |
| 242 | ///< that this number can be negative (in |
| 243 | ///< which case it basically means "zero") |
| 244 | int aw_first_pulse_off[2]; ///< index of first sample to which to |
| 245 | ///< apply AW-pulses, or -0xff if unset |
| 246 | int aw_next_pulse_off_cache; ///< the position (relative to start of the |
| 247 | ///< second block) at which pulses should |
| 248 | ///< start to be positioned, serves as a |
| 249 | ///< cache for pitch-adaptive window pulses |
| 250 | ///< between blocks |
| 251 | |
| 252 | int frame_cntr; ///< current frame index [0 - 0xFFFE]; is |
| 253 | ///< only used for comfort noise in #pRNG() |
| 254 | float gain_pred_err[6]; ///< cache for gain prediction |
| 255 | float excitation_history[MAX_SIGNAL_HISTORY]; |
| 256 | ///< cache of the signal of previous |
| 257 | ///< superframes, used as a history for |
| 258 | ///< signal generation |
| 259 | float synth_history[MAX_LSPS]; ///< see #excitation_history |
| 260 | /** |
| 261 | * @} |
| 262 | * |
| 263 | * @name Postfilter values |
| 264 | * |
| 265 | * Variables used for postfilter implementation, mostly history for |
| 266 | * smoothing and so on, and context variables for FFT/iFFT. |
| 267 | * @{ |
| 268 | */ |
| 269 | RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the |
| 270 | ///< postfilter (for denoise filter) |
| 271 | DCTContext dct, dst; ///< contexts for phase shift (in Hilbert |
| 272 | ///< transform, part of postfilter) |
| 273 | float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] |
| 274 | ///< range |
| 275 | float postfilter_agc; ///< gain control memory, used in |
| 276 | ///< #adaptive_gain_control() |
| 277 | float dcf_mem[2]; ///< DC filter history |
| 278 | float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; |
| 279 | ///< zero filter output (i.e. excitation) |
| 280 | ///< by postfilter |
| 281 | float denoise_filter_cache[MAX_FRAMESIZE]; |
| 282 | int denoise_filter_cache_size; ///< samples in #denoise_filter_cache |
| 283 | DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80]; |
| 284 | ///< aligned buffer for LPC tilting |
| 285 | DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; |
| 286 | ///< aligned buffer for denoise coefficients |
| 287 | DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; |
| 288 | ///< aligned buffer for postfilter speech |
| 289 | ///< synthesis |
| 290 | /** |
| 291 | * @} |
| 292 | */ |
| 293 | } WMAVoiceContext; |
| 294 | |
| 295 | /** |
| 296 | * Set up the variable bit mode (VBM) tree from container extradata. |
| 297 | * @param gb bit I/O context. |
| 298 | * The bit context (s->gb) should be loaded with byte 23-46 of the |
| 299 | * container extradata (i.e. the ones containing the VBM tree). |
| 300 | * @param vbm_tree pointer to array to which the decoded VBM tree will be |
| 301 | * written. |
| 302 | * @return 0 on success, <0 on error. |
| 303 | */ |
| 304 | static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) |
| 305 | { |
| 306 | int cntr[8] = { 0 }, n, res; |
| 307 | |
| 308 | memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); |
| 309 | for (n = 0; n < 17; n++) { |
| 310 | res = get_bits(gb, 3); |
| 311 | if (cntr[res] > 3) // should be >= 3 + (res == 7)) |
| 312 | return -1; |
| 313 | vbm_tree[res * 3 + cntr[res]++] = n; |
| 314 | } |
| 315 | return 0; |
| 316 | } |
| 317 | |
| 318 | static av_cold void wmavoice_init_static_data(AVCodec *codec) |
| 319 | { |
| 320 | static const uint8_t bits[] = { |
| 321 | 2, 2, 2, 4, 4, 4, |
| 322 | 6, 6, 6, 8, 8, 8, |
| 323 | 10, 10, 10, 12, 12, 12, |
| 324 | 14, 14, 14, 14 |
| 325 | }; |
| 326 | static const uint16_t codes[] = { |
| 327 | 0x0000, 0x0001, 0x0002, // 00/01/10 |
| 328 | 0x000c, 0x000d, 0x000e, // 11+00/01/10 |
| 329 | 0x003c, 0x003d, 0x003e, // 1111+00/01/10 |
| 330 | 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 |
| 331 | 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 |
| 332 | 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 |
| 333 | 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx |
| 334 | }; |
| 335 | |
| 336 | INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), |
| 337 | bits, 1, 1, codes, 2, 2, 132); |
| 338 | } |
| 339 | |
| 340 | /** |
| 341 | * Set up decoder with parameters from demuxer (extradata etc.). |
| 342 | */ |
| 343 | static av_cold int wmavoice_decode_init(AVCodecContext *ctx) |
| 344 | { |
| 345 | int n, flags, pitch_range, lsp16_flag; |
| 346 | WMAVoiceContext *s = ctx->priv_data; |
| 347 | |
| 348 | /** |
| 349 | * Extradata layout: |
| 350 | * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), |
| 351 | * - byte 19-22: flags field (annoyingly in LE; see below for known |
| 352 | * values), |
| 353 | * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, |
| 354 | * rest is 0). |
| 355 | */ |
| 356 | if (ctx->extradata_size != 46) { |
| 357 | av_log(ctx, AV_LOG_ERROR, |
| 358 | "Invalid extradata size %d (should be 46)\n", |
| 359 | ctx->extradata_size); |
| 360 | return AVERROR_INVALIDDATA; |
| 361 | } |
| 362 | flags = AV_RL32(ctx->extradata + 18); |
| 363 | s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); |
| 364 | s->do_apf = flags & 0x1; |
| 365 | if (s->do_apf) { |
| 366 | ff_rdft_init(&s->rdft, 7, DFT_R2C); |
| 367 | ff_rdft_init(&s->irdft, 7, IDFT_C2R); |
| 368 | ff_dct_init(&s->dct, 6, DCT_I); |
| 369 | ff_dct_init(&s->dst, 6, DST_I); |
| 370 | |
| 371 | ff_sine_window_init(s->cos, 256); |
| 372 | memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); |
| 373 | for (n = 0; n < 255; n++) { |
| 374 | s->sin[n] = -s->sin[510 - n]; |
| 375 | s->cos[510 - n] = s->cos[n]; |
| 376 | } |
| 377 | } |
| 378 | s->denoise_strength = (flags >> 2) & 0xF; |
| 379 | if (s->denoise_strength >= 12) { |
| 380 | av_log(ctx, AV_LOG_ERROR, |
| 381 | "Invalid denoise filter strength %d (max=11)\n", |
| 382 | s->denoise_strength); |
| 383 | return AVERROR_INVALIDDATA; |
| 384 | } |
| 385 | s->denoise_tilt_corr = !!(flags & 0x40); |
| 386 | s->dc_level = (flags >> 7) & 0xF; |
| 387 | s->lsp_q_mode = !!(flags & 0x2000); |
| 388 | s->lsp_def_mode = !!(flags & 0x4000); |
| 389 | lsp16_flag = flags & 0x1000; |
| 390 | if (lsp16_flag) { |
| 391 | s->lsps = 16; |
| 392 | s->frame_lsp_bitsize = 34; |
| 393 | s->sframe_lsp_bitsize = 60; |
| 394 | } else { |
| 395 | s->lsps = 10; |
| 396 | s->frame_lsp_bitsize = 24; |
| 397 | s->sframe_lsp_bitsize = 48; |
| 398 | } |
| 399 | for (n = 0; n < s->lsps; n++) |
| 400 | s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); |
| 401 | |
| 402 | init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); |
| 403 | if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { |
| 404 | av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); |
| 405 | return AVERROR_INVALIDDATA; |
| 406 | } |
| 407 | |
| 408 | s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; |
| 409 | s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; |
| 410 | pitch_range = s->max_pitch_val - s->min_pitch_val; |
| 411 | if (pitch_range <= 0) { |
| 412 | av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n"); |
| 413 | return AVERROR_INVALIDDATA; |
| 414 | } |
| 415 | s->pitch_nbits = av_ceil_log2(pitch_range); |
| 416 | s->last_pitch_val = 40; |
| 417 | s->last_acb_type = ACB_TYPE_NONE; |
| 418 | s->history_nsamples = s->max_pitch_val + 8; |
| 419 | |
| 420 | if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { |
| 421 | int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, |
| 422 | max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; |
| 423 | |
| 424 | av_log(ctx, AV_LOG_ERROR, |
| 425 | "Unsupported samplerate %d (min=%d, max=%d)\n", |
| 426 | ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz |
| 427 | |
| 428 | return AVERROR(ENOSYS); |
| 429 | } |
| 430 | |
| 431 | s->block_conv_table[0] = s->min_pitch_val; |
| 432 | s->block_conv_table[1] = (pitch_range * 25) >> 6; |
| 433 | s->block_conv_table[2] = (pitch_range * 44) >> 6; |
| 434 | s->block_conv_table[3] = s->max_pitch_val - 1; |
| 435 | s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; |
| 436 | if (s->block_delta_pitch_hrange <= 0) { |
| 437 | av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n"); |
| 438 | return AVERROR_INVALIDDATA; |
| 439 | } |
| 440 | s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); |
| 441 | s->block_pitch_range = s->block_conv_table[2] + |
| 442 | s->block_conv_table[3] + 1 + |
| 443 | 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); |
| 444 | s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); |
| 445 | |
| 446 | ctx->channels = 1; |
| 447 | ctx->channel_layout = AV_CH_LAYOUT_MONO; |
| 448 | ctx->sample_fmt = AV_SAMPLE_FMT_FLT; |
| 449 | |
| 450 | return 0; |
| 451 | } |
| 452 | |
| 453 | /** |
| 454 | * @name Postfilter functions |
| 455 | * Postfilter functions (gain control, wiener denoise filter, DC filter, |
| 456 | * kalman smoothening, plus surrounding code to wrap it) |
| 457 | * @{ |
| 458 | */ |
| 459 | /** |
| 460 | * Adaptive gain control (as used in postfilter). |
| 461 | * |
| 462 | * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except |
| 463 | * that the energy here is calculated using sum(abs(...)), whereas the |
| 464 | * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). |
| 465 | * |
| 466 | * @param out output buffer for filtered samples |
| 467 | * @param in input buffer containing the samples as they are after the |
| 468 | * postfilter steps so far |
| 469 | * @param speech_synth input buffer containing speech synth before postfilter |
| 470 | * @param size input buffer size |
| 471 | * @param alpha exponential filter factor |
| 472 | * @param gain_mem pointer to filter memory (single float) |
| 473 | */ |
| 474 | static void adaptive_gain_control(float *out, const float *in, |
| 475 | const float *speech_synth, |
| 476 | int size, float alpha, float *gain_mem) |
| 477 | { |
| 478 | int i; |
| 479 | float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; |
| 480 | float mem = *gain_mem; |
| 481 | |
| 482 | for (i = 0; i < size; i++) { |
| 483 | speech_energy += fabsf(speech_synth[i]); |
| 484 | postfilter_energy += fabsf(in[i]); |
| 485 | } |
| 486 | gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; |
| 487 | |
| 488 | for (i = 0; i < size; i++) { |
| 489 | mem = alpha * mem + gain_scale_factor; |
| 490 | out[i] = in[i] * mem; |
| 491 | } |
| 492 | |
| 493 | *gain_mem = mem; |
| 494 | } |
| 495 | |
| 496 | /** |
| 497 | * Kalman smoothing function. |
| 498 | * |
| 499 | * This function looks back pitch +/- 3 samples back into history to find |
| 500 | * the best fitting curve (that one giving the optimal gain of the two |
| 501 | * signals, i.e. the highest dot product between the two), and then |
| 502 | * uses that signal history to smoothen the output of the speech synthesis |
| 503 | * filter. |
| 504 | * |
| 505 | * @param s WMA Voice decoding context |
| 506 | * @param pitch pitch of the speech signal |
| 507 | * @param in input speech signal |
| 508 | * @param out output pointer for smoothened signal |
| 509 | * @param size input/output buffer size |
| 510 | * |
| 511 | * @returns -1 if no smoothening took place, e.g. because no optimal |
| 512 | * fit could be found, or 0 on success. |
| 513 | */ |
| 514 | static int kalman_smoothen(WMAVoiceContext *s, int pitch, |
| 515 | const float *in, float *out, int size) |
| 516 | { |
| 517 | int n; |
| 518 | float optimal_gain = 0, dot; |
| 519 | const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], |
| 520 | *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], |
| 521 | *best_hist_ptr = NULL; |
| 522 | |
| 523 | /* find best fitting point in history */ |
| 524 | do { |
| 525 | dot = avpriv_scalarproduct_float_c(in, ptr, size); |
| 526 | if (dot > optimal_gain) { |
| 527 | optimal_gain = dot; |
| 528 | best_hist_ptr = ptr; |
| 529 | } |
| 530 | } while (--ptr >= end); |
| 531 | |
| 532 | if (optimal_gain <= 0) |
| 533 | return -1; |
| 534 | dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size); |
| 535 | if (dot <= 0) // would be 1.0 |
| 536 | return -1; |
| 537 | |
| 538 | if (optimal_gain <= dot) { |
| 539 | dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 |
| 540 | } else |
| 541 | dot = 0.625; |
| 542 | |
| 543 | /* actual smoothing */ |
| 544 | for (n = 0; n < size; n++) |
| 545 | out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); |
| 546 | |
| 547 | return 0; |
| 548 | } |
| 549 | |
| 550 | /** |
| 551 | * Get the tilt factor of a formant filter from its transfer function |
| 552 | * @see #tilt_factor() in amrnbdec.c, which does essentially the same, |
| 553 | * but somehow (??) it does a speech synthesis filter in the |
| 554 | * middle, which is missing here |
| 555 | * |
| 556 | * @param lpcs LPC coefficients |
| 557 | * @param n_lpcs Size of LPC buffer |
| 558 | * @returns the tilt factor |
| 559 | */ |
| 560 | static float tilt_factor(const float *lpcs, int n_lpcs) |
| 561 | { |
| 562 | float rh0, rh1; |
| 563 | |
| 564 | rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs); |
| 565 | rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1); |
| 566 | |
| 567 | return rh1 / rh0; |
| 568 | } |
| 569 | |
| 570 | /** |
| 571 | * Derive denoise filter coefficients (in real domain) from the LPCs. |
| 572 | */ |
| 573 | static void calc_input_response(WMAVoiceContext *s, float *lpcs, |
| 574 | int fcb_type, float *coeffs, int remainder) |
| 575 | { |
| 576 | float last_coeff, min = 15.0, max = -15.0; |
| 577 | float irange, angle_mul, gain_mul, range, sq; |
| 578 | int n, idx; |
| 579 | |
| 580 | /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ |
| 581 | s->rdft.rdft_calc(&s->rdft, lpcs); |
| 582 | #define log_range(var, assign) do { \ |
| 583 | float tmp = log10f(assign); var = tmp; \ |
| 584 | max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ |
| 585 | } while (0) |
| 586 | log_range(last_coeff, lpcs[1] * lpcs[1]); |
| 587 | for (n = 1; n < 64; n++) |
| 588 | log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + |
| 589 | lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); |
| 590 | log_range(lpcs[0], lpcs[0] * lpcs[0]); |
| 591 | #undef log_range |
| 592 | range = max - min; |
| 593 | lpcs[64] = last_coeff; |
| 594 | |
| 595 | /* Now, use this spectrum to pick out these frequencies with higher |
| 596 | * (relative) power/energy (which we then take to be "not noise"), |
| 597 | * and set up a table (still in lpc[]) of (relative) gains per frequency. |
| 598 | * These frequencies will be maintained, while others ("noise") will be |
| 599 | * decreased in the filter output. */ |
| 600 | irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] |
| 601 | gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : |
| 602 | (5.0 / 14.7)); |
| 603 | angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); |
| 604 | for (n = 0; n <= 64; n++) { |
| 605 | float pwr; |
| 606 | |
| 607 | idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1); |
| 608 | pwr = wmavoice_denoise_power_table[s->denoise_strength][idx]; |
| 609 | lpcs[n] = angle_mul * pwr; |
| 610 | |
| 611 | /* 70.57 =~ 1/log10(1.0331663) */ |
| 612 | idx = (pwr * gain_mul - 0.0295) * 70.570526123; |
| 613 | if (idx > 127) { // fall back if index falls outside table range |
| 614 | coeffs[n] = wmavoice_energy_table[127] * |
| 615 | powf(1.0331663, idx - 127); |
| 616 | } else |
| 617 | coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; |
| 618 | } |
| 619 | |
| 620 | /* calculate the Hilbert transform of the gains, which we do (since this |
| 621 | * is a sine input) by doing a phase shift (in theory, H(sin())=cos()). |
| 622 | * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the |
| 623 | * "moment" of the LPCs in this filter. */ |
| 624 | s->dct.dct_calc(&s->dct, lpcs); |
| 625 | s->dst.dct_calc(&s->dst, lpcs); |
| 626 | |
| 627 | /* Split out the coefficient indexes into phase/magnitude pairs */ |
| 628 | idx = 255 + av_clip(lpcs[64], -255, 255); |
| 629 | coeffs[0] = coeffs[0] * s->cos[idx]; |
| 630 | idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); |
| 631 | last_coeff = coeffs[64] * s->cos[idx]; |
| 632 | for (n = 63;; n--) { |
| 633 | idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); |
| 634 | coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; |
| 635 | coeffs[n * 2] = coeffs[n] * s->cos[idx]; |
| 636 | |
| 637 | if (!--n) break; |
| 638 | |
| 639 | idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); |
| 640 | coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; |
| 641 | coeffs[n * 2] = coeffs[n] * s->cos[idx]; |
| 642 | } |
| 643 | coeffs[1] = last_coeff; |
| 644 | |
| 645 | /* move into real domain */ |
| 646 | s->irdft.rdft_calc(&s->irdft, coeffs); |
| 647 | |
| 648 | /* tilt correction and normalize scale */ |
| 649 | memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); |
| 650 | if (s->denoise_tilt_corr) { |
| 651 | float tilt_mem = 0; |
| 652 | |
| 653 | coeffs[remainder - 1] = 0; |
| 654 | ff_tilt_compensation(&tilt_mem, |
| 655 | -1.8 * tilt_factor(coeffs, remainder - 1), |
| 656 | coeffs, remainder); |
| 657 | } |
| 658 | sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs, |
| 659 | remainder)); |
| 660 | for (n = 0; n < remainder; n++) |
| 661 | coeffs[n] *= sq; |
| 662 | } |
| 663 | |
| 664 | /** |
| 665 | * This function applies a Wiener filter on the (noisy) speech signal as |
| 666 | * a means to denoise it. |
| 667 | * |
| 668 | * - take RDFT of LPCs to get the power spectrum of the noise + speech; |
| 669 | * - using this power spectrum, calculate (for each frequency) the Wiener |
| 670 | * filter gain, which depends on the frequency power and desired level |
| 671 | * of noise subtraction (when set too high, this leads to artifacts) |
| 672 | * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse |
| 673 | * of 4-8kHz); |
| 674 | * - by doing a phase shift, calculate the Hilbert transform of this array |
| 675 | * of per-frequency filter-gains to get the filtering coefficients; |
| 676 | * - smoothen/normalize/de-tilt these filter coefficients as desired; |
| 677 | * - take RDFT of noisy sound, apply the coefficients and take its IRDFT |
| 678 | * to get the denoised speech signal; |
| 679 | * - the leftover (i.e. output of the IRDFT on denoised speech data beyond |
| 680 | * the frame boundary) are saved and applied to subsequent frames by an |
| 681 | * overlap-add method (otherwise you get clicking-artifacts). |
| 682 | * |
| 683 | * @param s WMA Voice decoding context |
| 684 | * @param fcb_type Frame (codebook) type |
| 685 | * @param synth_pf input: the noisy speech signal, output: denoised speech |
| 686 | * data; should be 16-byte aligned (for ASM purposes) |
| 687 | * @param size size of the speech data |
| 688 | * @param lpcs LPCs used to synthesize this frame's speech data |
| 689 | */ |
| 690 | static void wiener_denoise(WMAVoiceContext *s, int fcb_type, |
| 691 | float *synth_pf, int size, |
| 692 | const float *lpcs) |
| 693 | { |
| 694 | int remainder, lim, n; |
| 695 | |
| 696 | if (fcb_type != FCB_TYPE_SILENCE) { |
| 697 | float *tilted_lpcs = s->tilted_lpcs_pf, |
| 698 | *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; |
| 699 | |
| 700 | tilted_lpcs[0] = 1.0; |
| 701 | memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); |
| 702 | memset(&tilted_lpcs[s->lsps + 1], 0, |
| 703 | sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); |
| 704 | ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), |
| 705 | tilted_lpcs, s->lsps + 2); |
| 706 | |
| 707 | /* The IRDFT output (127 samples for 7-bit filter) beyond the frame |
| 708 | * size is applied to the next frame. All input beyond this is zero, |
| 709 | * and thus all output beyond this will go towards zero, hence we can |
| 710 | * limit to min(size-1, 127-size) as a performance consideration. */ |
| 711 | remainder = FFMIN(127 - size, size - 1); |
| 712 | calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); |
| 713 | |
| 714 | /* apply coefficients (in frequency spectrum domain), i.e. complex |
| 715 | * number multiplication */ |
| 716 | memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); |
| 717 | s->rdft.rdft_calc(&s->rdft, synth_pf); |
| 718 | s->rdft.rdft_calc(&s->rdft, coeffs); |
| 719 | synth_pf[0] *= coeffs[0]; |
| 720 | synth_pf[1] *= coeffs[1]; |
| 721 | for (n = 1; n < 64; n++) { |
| 722 | float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; |
| 723 | synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; |
| 724 | synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; |
| 725 | } |
| 726 | s->irdft.rdft_calc(&s->irdft, synth_pf); |
| 727 | } |
| 728 | |
| 729 | /* merge filter output with the history of previous runs */ |
| 730 | if (s->denoise_filter_cache_size) { |
| 731 | lim = FFMIN(s->denoise_filter_cache_size, size); |
| 732 | for (n = 0; n < lim; n++) |
| 733 | synth_pf[n] += s->denoise_filter_cache[n]; |
| 734 | s->denoise_filter_cache_size -= lim; |
| 735 | memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], |
| 736 | sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); |
| 737 | } |
| 738 | |
| 739 | /* move remainder of filter output into a cache for future runs */ |
| 740 | if (fcb_type != FCB_TYPE_SILENCE) { |
| 741 | lim = FFMIN(remainder, s->denoise_filter_cache_size); |
| 742 | for (n = 0; n < lim; n++) |
| 743 | s->denoise_filter_cache[n] += synth_pf[size + n]; |
| 744 | if (lim < remainder) { |
| 745 | memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], |
| 746 | sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); |
| 747 | s->denoise_filter_cache_size = remainder; |
| 748 | } |
| 749 | } |
| 750 | } |
| 751 | |
| 752 | /** |
| 753 | * Averaging projection filter, the postfilter used in WMAVoice. |
| 754 | * |
| 755 | * This uses the following steps: |
| 756 | * - A zero-synthesis filter (generate excitation from synth signal) |
| 757 | * - Kalman smoothing on excitation, based on pitch |
| 758 | * - Re-synthesized smoothened output |
| 759 | * - Iterative Wiener denoise filter |
| 760 | * - Adaptive gain filter |
| 761 | * - DC filter |
| 762 | * |
| 763 | * @param s WMAVoice decoding context |
| 764 | * @param synth Speech synthesis output (before postfilter) |
| 765 | * @param samples Output buffer for filtered samples |
| 766 | * @param size Buffer size of synth & samples |
| 767 | * @param lpcs Generated LPCs used for speech synthesis |
| 768 | * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned) |
| 769 | * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) |
| 770 | * @param pitch Pitch of the input signal |
| 771 | */ |
| 772 | static void postfilter(WMAVoiceContext *s, const float *synth, |
| 773 | float *samples, int size, |
| 774 | const float *lpcs, float *zero_exc_pf, |
| 775 | int fcb_type, int pitch) |
| 776 | { |
| 777 | float synth_filter_in_buf[MAX_FRAMESIZE / 2], |
| 778 | *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], |
| 779 | *synth_filter_in = zero_exc_pf; |
| 780 | |
| 781 | av_assert0(size <= MAX_FRAMESIZE / 2); |
| 782 | |
| 783 | /* generate excitation from input signal */ |
| 784 | ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); |
| 785 | |
| 786 | if (fcb_type >= FCB_TYPE_AW_PULSES && |
| 787 | !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) |
| 788 | synth_filter_in = synth_filter_in_buf; |
| 789 | |
| 790 | /* re-synthesize speech after smoothening, and keep history */ |
| 791 | ff_celp_lp_synthesis_filterf(synth_pf, lpcs, |
| 792 | synth_filter_in, size, s->lsps); |
| 793 | memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], |
| 794 | sizeof(synth_pf[0]) * s->lsps); |
| 795 | |
| 796 | wiener_denoise(s, fcb_type, synth_pf, size, lpcs); |
| 797 | |
| 798 | adaptive_gain_control(samples, synth_pf, synth, size, 0.99, |
| 799 | &s->postfilter_agc); |
| 800 | |
| 801 | if (s->dc_level > 8) { |
| 802 | /* remove ultra-low frequency DC noise / highpass filter; |
| 803 | * coefficients are identical to those used in SIPR decoding, |
| 804 | * and very closely resemble those used in AMR-NB decoding. */ |
| 805 | ff_acelp_apply_order_2_transfer_function(samples, samples, |
| 806 | (const float[2]) { -1.99997, 1.0 }, |
| 807 | (const float[2]) { -1.9330735188, 0.93589198496 }, |
| 808 | 0.93980580475, s->dcf_mem, size); |
| 809 | } |
| 810 | } |
| 811 | /** |
| 812 | * @} |
| 813 | */ |
| 814 | |
| 815 | /** |
| 816 | * Dequantize LSPs |
| 817 | * @param lsps output pointer to the array that will hold the LSPs |
| 818 | * @param num number of LSPs to be dequantized |
| 819 | * @param values quantized values, contains n_stages values |
| 820 | * @param sizes range (i.e. max value) of each quantized value |
| 821 | * @param n_stages number of dequantization runs |
| 822 | * @param table dequantization table to be used |
| 823 | * @param mul_q LSF multiplier |
| 824 | * @param base_q base (lowest) LSF values |
| 825 | */ |
| 826 | static void dequant_lsps(double *lsps, int num, |
| 827 | const uint16_t *values, |
| 828 | const uint16_t *sizes, |
| 829 | int n_stages, const uint8_t *table, |
| 830 | const double *mul_q, |
| 831 | const double *base_q) |
| 832 | { |
| 833 | int n, m; |
| 834 | |
| 835 | memset(lsps, 0, num * sizeof(*lsps)); |
| 836 | for (n = 0; n < n_stages; n++) { |
| 837 | const uint8_t *t_off = &table[values[n] * num]; |
| 838 | double base = base_q[n], mul = mul_q[n]; |
| 839 | |
| 840 | for (m = 0; m < num; m++) |
| 841 | lsps[m] += base + mul * t_off[m]; |
| 842 | |
| 843 | table += sizes[n] * num; |
| 844 | } |
| 845 | } |
| 846 | |
| 847 | /** |
| 848 | * @name LSP dequantization routines |
| 849 | * LSP dequantization routines, for 10/16LSPs and independent/residual coding. |
| 850 | * @note we assume enough bits are available, caller should check. |
| 851 | * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; |
| 852 | * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. |
| 853 | * @{ |
| 854 | */ |
| 855 | /** |
| 856 | * Parse 10 independently-coded LSPs. |
| 857 | */ |
| 858 | static void dequant_lsp10i(GetBitContext *gb, double *lsps) |
| 859 | { |
| 860 | static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; |
| 861 | static const double mul_lsf[4] = { |
| 862 | 5.2187144800e-3, 1.4626986422e-3, |
| 863 | 9.6179549166e-4, 1.1325736225e-3 |
| 864 | }; |
| 865 | static const double base_lsf[4] = { |
| 866 | M_PI * -2.15522e-1, M_PI * -6.1646e-2, |
| 867 | M_PI * -3.3486e-2, M_PI * -5.7408e-2 |
| 868 | }; |
| 869 | uint16_t v[4]; |
| 870 | |
| 871 | v[0] = get_bits(gb, 8); |
| 872 | v[1] = get_bits(gb, 6); |
| 873 | v[2] = get_bits(gb, 5); |
| 874 | v[3] = get_bits(gb, 5); |
| 875 | |
| 876 | dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, |
| 877 | mul_lsf, base_lsf); |
| 878 | } |
| 879 | |
| 880 | /** |
| 881 | * Parse 10 independently-coded LSPs, and then derive the tables to |
| 882 | * generate LSPs for the other frames from them (residual coding). |
| 883 | */ |
| 884 | static void dequant_lsp10r(GetBitContext *gb, |
| 885 | double *i_lsps, const double *old, |
| 886 | double *a1, double *a2, int q_mode) |
| 887 | { |
| 888 | static const uint16_t vec_sizes[3] = { 128, 64, 64 }; |
| 889 | static const double mul_lsf[3] = { |
| 890 | 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 |
| 891 | }; |
| 892 | static const double base_lsf[3] = { |
| 893 | M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 |
| 894 | }; |
| 895 | const float (*ipol_tab)[2][10] = q_mode ? |
| 896 | wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; |
| 897 | uint16_t interpol, v[3]; |
| 898 | int n; |
| 899 | |
| 900 | dequant_lsp10i(gb, i_lsps); |
| 901 | |
| 902 | interpol = get_bits(gb, 5); |
| 903 | v[0] = get_bits(gb, 7); |
| 904 | v[1] = get_bits(gb, 6); |
| 905 | v[2] = get_bits(gb, 6); |
| 906 | |
| 907 | for (n = 0; n < 10; n++) { |
| 908 | double delta = old[n] - i_lsps[n]; |
| 909 | a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; |
| 910 | a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; |
| 911 | } |
| 912 | |
| 913 | dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, |
| 914 | mul_lsf, base_lsf); |
| 915 | } |
| 916 | |
| 917 | /** |
| 918 | * Parse 16 independently-coded LSPs. |
| 919 | */ |
| 920 | static void dequant_lsp16i(GetBitContext *gb, double *lsps) |
| 921 | { |
| 922 | static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; |
| 923 | static const double mul_lsf[5] = { |
| 924 | 3.3439586280e-3, 6.9908173703e-4, |
| 925 | 3.3216608306e-3, 1.0334960326e-3, |
| 926 | 3.1899104283e-3 |
| 927 | }; |
| 928 | static const double base_lsf[5] = { |
| 929 | M_PI * -1.27576e-1, M_PI * -2.4292e-2, |
| 930 | M_PI * -1.28094e-1, M_PI * -3.2128e-2, |
| 931 | M_PI * -1.29816e-1 |
| 932 | }; |
| 933 | uint16_t v[5]; |
| 934 | |
| 935 | v[0] = get_bits(gb, 8); |
| 936 | v[1] = get_bits(gb, 6); |
| 937 | v[2] = get_bits(gb, 7); |
| 938 | v[3] = get_bits(gb, 6); |
| 939 | v[4] = get_bits(gb, 7); |
| 940 | |
| 941 | dequant_lsps( lsps, 5, v, vec_sizes, 2, |
| 942 | wmavoice_dq_lsp16i1, mul_lsf, base_lsf); |
| 943 | dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, |
| 944 | wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); |
| 945 | dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, |
| 946 | wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); |
| 947 | } |
| 948 | |
| 949 | /** |
| 950 | * Parse 16 independently-coded LSPs, and then derive the tables to |
| 951 | * generate LSPs for the other frames from them (residual coding). |
| 952 | */ |
| 953 | static void dequant_lsp16r(GetBitContext *gb, |
| 954 | double *i_lsps, const double *old, |
| 955 | double *a1, double *a2, int q_mode) |
| 956 | { |
| 957 | static const uint16_t vec_sizes[3] = { 128, 128, 128 }; |
| 958 | static const double mul_lsf[3] = { |
| 959 | 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 |
| 960 | }; |
| 961 | static const double base_lsf[3] = { |
| 962 | M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 |
| 963 | }; |
| 964 | const float (*ipol_tab)[2][16] = q_mode ? |
| 965 | wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; |
| 966 | uint16_t interpol, v[3]; |
| 967 | int n; |
| 968 | |
| 969 | dequant_lsp16i(gb, i_lsps); |
| 970 | |
| 971 | interpol = get_bits(gb, 5); |
| 972 | v[0] = get_bits(gb, 7); |
| 973 | v[1] = get_bits(gb, 7); |
| 974 | v[2] = get_bits(gb, 7); |
| 975 | |
| 976 | for (n = 0; n < 16; n++) { |
| 977 | double delta = old[n] - i_lsps[n]; |
| 978 | a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; |
| 979 | a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; |
| 980 | } |
| 981 | |
| 982 | dequant_lsps( a2, 10, v, vec_sizes, 1, |
| 983 | wmavoice_dq_lsp16r1, mul_lsf, base_lsf); |
| 984 | dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, |
| 985 | wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); |
| 986 | dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, |
| 987 | wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); |
| 988 | } |
| 989 | |
| 990 | /** |
| 991 | * @} |
| 992 | * @name Pitch-adaptive window coding functions |
| 993 | * The next few functions are for pitch-adaptive window coding. |
| 994 | * @{ |
| 995 | */ |
| 996 | /** |
| 997 | * Parse the offset of the first pitch-adaptive window pulses, and |
| 998 | * the distribution of pulses between the two blocks in this frame. |
| 999 | * @param s WMA Voice decoding context private data |
| 1000 | * @param gb bit I/O context |
| 1001 | * @param pitch pitch for each block in this frame |
| 1002 | */ |
| 1003 | static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, |
| 1004 | const int *pitch) |
| 1005 | { |
| 1006 | static const int16_t start_offset[94] = { |
| 1007 | -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, |
| 1008 | 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, |
| 1009 | 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, |
| 1010 | 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, |
| 1011 | 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, |
| 1012 | 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, |
| 1013 | 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, |
| 1014 | 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 |
| 1015 | }; |
| 1016 | int bits, offset; |
| 1017 | |
| 1018 | /* position of pulse */ |
| 1019 | s->aw_idx_is_ext = 0; |
| 1020 | if ((bits = get_bits(gb, 6)) >= 54) { |
| 1021 | s->aw_idx_is_ext = 1; |
| 1022 | bits += (bits - 54) * 3 + get_bits(gb, 2); |
| 1023 | } |
| 1024 | |
| 1025 | /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count |
| 1026 | * the distribution of the pulses in each block contained in this frame. */ |
| 1027 | s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; |
| 1028 | for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; |
| 1029 | s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; |
| 1030 | s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; |
| 1031 | offset += s->aw_n_pulses[0] * pitch[0]; |
| 1032 | s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; |
| 1033 | s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; |
| 1034 | |
| 1035 | /* if continuing from a position before the block, reset position to |
| 1036 | * start of block (when corrected for the range over which it can be |
| 1037 | * spread in aw_pulse_set1()). */ |
| 1038 | if (start_offset[bits] < MAX_FRAMESIZE / 2) { |
| 1039 | while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) |
| 1040 | s->aw_first_pulse_off[1] -= pitch[1]; |
| 1041 | if (start_offset[bits] < 0) |
| 1042 | while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) |
| 1043 | s->aw_first_pulse_off[0] -= pitch[0]; |
| 1044 | } |
| 1045 | } |
| 1046 | |
| 1047 | /** |
| 1048 | * Apply second set of pitch-adaptive window pulses. |
| 1049 | * @param s WMA Voice decoding context private data |
| 1050 | * @param gb bit I/O context |
| 1051 | * @param block_idx block index in frame [0, 1] |
| 1052 | * @param fcb structure containing fixed codebook vector info |
| 1053 | * @return -1 on error, 0 otherwise |
| 1054 | */ |
| 1055 | static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, |
| 1056 | int block_idx, AMRFixed *fcb) |
| 1057 | { |
| 1058 | uint16_t use_mask_mem[9]; // only 5 are used, rest is padding |
| 1059 | uint16_t *use_mask = use_mask_mem + 2; |
| 1060 | /* in this function, idx is the index in the 80-bit (+ padding) use_mask |
| 1061 | * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits |
| 1062 | * of idx are the position of the bit within a particular item in the |
| 1063 | * array (0 being the most significant bit, and 15 being the least |
| 1064 | * significant bit), and the remainder (>> 4) is the index in the |
| 1065 | * use_mask[]-array. This is faster and uses less memory than using a |
| 1066 | * 80-byte/80-int array. */ |
| 1067 | int pulse_off = s->aw_first_pulse_off[block_idx], |
| 1068 | pulse_start, n, idx, range, aidx, start_off = 0; |
| 1069 | |
| 1070 | /* set offset of first pulse to within this block */ |
| 1071 | if (s->aw_n_pulses[block_idx] > 0) |
| 1072 | while (pulse_off + s->aw_pulse_range < 1) |
| 1073 | pulse_off += fcb->pitch_lag; |
| 1074 | |
| 1075 | /* find range per pulse */ |
| 1076 | if (s->aw_n_pulses[0] > 0) { |
| 1077 | if (block_idx == 0) { |
| 1078 | range = 32; |
| 1079 | } else /* block_idx = 1 */ { |
| 1080 | range = 8; |
| 1081 | if (s->aw_n_pulses[block_idx] > 0) |
| 1082 | pulse_off = s->aw_next_pulse_off_cache; |
| 1083 | } |
| 1084 | } else |
| 1085 | range = 16; |
| 1086 | pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; |
| 1087 | |
| 1088 | /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, |
| 1089 | * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus |
| 1090 | * we exclude that range from being pulsed again in this function. */ |
| 1091 | memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0])); |
| 1092 | memset( use_mask, -1, 5 * sizeof(use_mask[0])); |
| 1093 | memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); |
| 1094 | if (s->aw_n_pulses[block_idx] > 0) |
| 1095 | for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { |
| 1096 | int excl_range = s->aw_pulse_range; // always 16 or 24 |
| 1097 | uint16_t *use_mask_ptr = &use_mask[idx >> 4]; |
| 1098 | int first_sh = 16 - (idx & 15); |
| 1099 | *use_mask_ptr++ &= 0xFFFFu << first_sh; |
| 1100 | excl_range -= first_sh; |
| 1101 | if (excl_range >= 16) { |
| 1102 | *use_mask_ptr++ = 0; |
| 1103 | *use_mask_ptr &= 0xFFFF >> (excl_range - 16); |
| 1104 | } else |
| 1105 | *use_mask_ptr &= 0xFFFF >> excl_range; |
| 1106 | } |
| 1107 | |
| 1108 | /* find the 'aidx'th offset that is not excluded */ |
| 1109 | aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); |
| 1110 | for (n = 0; n <= aidx; pulse_start++) { |
| 1111 | for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; |
| 1112 | if (idx >= MAX_FRAMESIZE / 2) { // find from zero |
| 1113 | if (use_mask[0]) idx = 0x0F; |
| 1114 | else if (use_mask[1]) idx = 0x1F; |
| 1115 | else if (use_mask[2]) idx = 0x2F; |
| 1116 | else if (use_mask[3]) idx = 0x3F; |
| 1117 | else if (use_mask[4]) idx = 0x4F; |
| 1118 | else return -1; |
| 1119 | idx -= av_log2_16bit(use_mask[idx >> 4]); |
| 1120 | } |
| 1121 | if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { |
| 1122 | use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); |
| 1123 | n++; |
| 1124 | start_off = idx; |
| 1125 | } |
| 1126 | } |
| 1127 | |
| 1128 | fcb->x[fcb->n] = start_off; |
| 1129 | fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; |
| 1130 | fcb->n++; |
| 1131 | |
| 1132 | /* set offset for next block, relative to start of that block */ |
| 1133 | n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; |
| 1134 | s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; |
| 1135 | return 0; |
| 1136 | } |
| 1137 | |
| 1138 | /** |
| 1139 | * Apply first set of pitch-adaptive window pulses. |
| 1140 | * @param s WMA Voice decoding context private data |
| 1141 | * @param gb bit I/O context |
| 1142 | * @param block_idx block index in frame [0, 1] |
| 1143 | * @param fcb storage location for fixed codebook pulse info |
| 1144 | */ |
| 1145 | static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, |
| 1146 | int block_idx, AMRFixed *fcb) |
| 1147 | { |
| 1148 | int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); |
| 1149 | float v; |
| 1150 | |
| 1151 | if (s->aw_n_pulses[block_idx] > 0) { |
| 1152 | int n, v_mask, i_mask, sh, n_pulses; |
| 1153 | |
| 1154 | if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each |
| 1155 | n_pulses = 3; |
| 1156 | v_mask = 8; |
| 1157 | i_mask = 7; |
| 1158 | sh = 4; |
| 1159 | } else { // 4 pulses, 1:sign + 2:index each |
| 1160 | n_pulses = 4; |
| 1161 | v_mask = 4; |
| 1162 | i_mask = 3; |
| 1163 | sh = 3; |
| 1164 | } |
| 1165 | |
| 1166 | for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { |
| 1167 | fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; |
| 1168 | fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + |
| 1169 | s->aw_first_pulse_off[block_idx]; |
| 1170 | while (fcb->x[fcb->n] < 0) |
| 1171 | fcb->x[fcb->n] += fcb->pitch_lag; |
| 1172 | if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) |
| 1173 | fcb->n++; |
| 1174 | } |
| 1175 | } else { |
| 1176 | int num2 = (val & 0x1FF) >> 1, delta, idx; |
| 1177 | |
| 1178 | if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } |
| 1179 | else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } |
| 1180 | else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } |
| 1181 | else { delta = 7; idx = num2 + 1 - 3 * 75; } |
| 1182 | v = (val & 0x200) ? -1.0 : 1.0; |
| 1183 | |
| 1184 | fcb->no_repeat_mask |= 3 << fcb->n; |
| 1185 | fcb->x[fcb->n] = idx - delta; |
| 1186 | fcb->y[fcb->n] = v; |
| 1187 | fcb->x[fcb->n + 1] = idx; |
| 1188 | fcb->y[fcb->n + 1] = (val & 1) ? -v : v; |
| 1189 | fcb->n += 2; |
| 1190 | } |
| 1191 | } |
| 1192 | |
| 1193 | /** |
| 1194 | * @} |
| 1195 | * |
| 1196 | * Generate a random number from frame_cntr and block_idx, which will lief |
| 1197 | * in the range [0, 1000 - block_size] (so it can be used as an index in a |
| 1198 | * table of size 1000 of which you want to read block_size entries). |
| 1199 | * |
| 1200 | * @param frame_cntr current frame number |
| 1201 | * @param block_num current block index |
| 1202 | * @param block_size amount of entries we want to read from a table |
| 1203 | * that has 1000 entries |
| 1204 | * @return a (non-)random number in the [0, 1000 - block_size] range. |
| 1205 | */ |
| 1206 | static int pRNG(int frame_cntr, int block_num, int block_size) |
| 1207 | { |
| 1208 | /* array to simplify the calculation of z: |
| 1209 | * y = (x % 9) * 5 + 6; |
| 1210 | * z = (49995 * x) / y; |
| 1211 | * Since y only has 9 values, we can remove the division by using a |
| 1212 | * LUT and using FASTDIV-style divisions. For each of the 9 values |
| 1213 | * of y, we can rewrite z as: |
| 1214 | * z = x * (49995 / y) + x * ((49995 % y) / y) |
| 1215 | * In this table, each col represents one possible value of y, the |
| 1216 | * first number is 49995 / y, and the second is the FASTDIV variant |
| 1217 | * of 49995 % y / y. */ |
| 1218 | static const unsigned int div_tbl[9][2] = { |
| 1219 | { 8332, 3 * 715827883U }, // y = 6 |
| 1220 | { 4545, 0 * 390451573U }, // y = 11 |
| 1221 | { 3124, 11 * 268435456U }, // y = 16 |
| 1222 | { 2380, 15 * 204522253U }, // y = 21 |
| 1223 | { 1922, 23 * 165191050U }, // y = 26 |
| 1224 | { 1612, 23 * 138547333U }, // y = 31 |
| 1225 | { 1388, 27 * 119304648U }, // y = 36 |
| 1226 | { 1219, 16 * 104755300U }, // y = 41 |
| 1227 | { 1086, 39 * 93368855U } // y = 46 |
| 1228 | }; |
| 1229 | unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; |
| 1230 | if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, |
| 1231 | // so this is effectively a modulo (%) |
| 1232 | y = x - 9 * MULH(477218589, x); // x % 9 |
| 1233 | z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); |
| 1234 | // z = x * 49995 / (y * 5 + 6) |
| 1235 | return z % (1000 - block_size); |
| 1236 | } |
| 1237 | |
| 1238 | /** |
| 1239 | * Parse hardcoded signal for a single block. |
| 1240 | * @note see #synth_block(). |
| 1241 | */ |
| 1242 | static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, |
| 1243 | int block_idx, int size, |
| 1244 | const struct frame_type_desc *frame_desc, |
| 1245 | float *excitation) |
| 1246 | { |
| 1247 | float gain; |
| 1248 | int n, r_idx; |
| 1249 | |
| 1250 | av_assert0(size <= MAX_FRAMESIZE); |
| 1251 | |
| 1252 | /* Set the offset from which we start reading wmavoice_std_codebook */ |
| 1253 | if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
| 1254 | r_idx = pRNG(s->frame_cntr, block_idx, size); |
| 1255 | gain = s->silence_gain; |
| 1256 | } else /* FCB_TYPE_HARDCODED */ { |
| 1257 | r_idx = get_bits(gb, 8); |
| 1258 | gain = wmavoice_gain_universal[get_bits(gb, 6)]; |
| 1259 | } |
| 1260 | |
| 1261 | /* Clear gain prediction parameters */ |
| 1262 | memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); |
| 1263 | |
| 1264 | /* Apply gain to hardcoded codebook and use that as excitation signal */ |
| 1265 | for (n = 0; n < size; n++) |
| 1266 | excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; |
| 1267 | } |
| 1268 | |
| 1269 | /** |
| 1270 | * Parse FCB/ACB signal for a single block. |
| 1271 | * @note see #synth_block(). |
| 1272 | */ |
| 1273 | static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, |
| 1274 | int block_idx, int size, |
| 1275 | int block_pitch_sh2, |
| 1276 | const struct frame_type_desc *frame_desc, |
| 1277 | float *excitation) |
| 1278 | { |
| 1279 | static const float gain_coeff[6] = { |
| 1280 | 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 |
| 1281 | }; |
| 1282 | float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; |
| 1283 | int n, idx, gain_weight; |
| 1284 | AMRFixed fcb; |
| 1285 | |
| 1286 | av_assert0(size <= MAX_FRAMESIZE / 2); |
| 1287 | memset(pulses, 0, sizeof(*pulses) * size); |
| 1288 | |
| 1289 | fcb.pitch_lag = block_pitch_sh2 >> 2; |
| 1290 | fcb.pitch_fac = 1.0; |
| 1291 | fcb.no_repeat_mask = 0; |
| 1292 | fcb.n = 0; |
| 1293 | |
| 1294 | /* For the other frame types, this is where we apply the innovation |
| 1295 | * (fixed) codebook pulses of the speech signal. */ |
| 1296 | if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| 1297 | aw_pulse_set1(s, gb, block_idx, &fcb); |
| 1298 | if (aw_pulse_set2(s, gb, block_idx, &fcb)) { |
| 1299 | /* Conceal the block with silence and return. |
| 1300 | * Skip the correct amount of bits to read the next |
| 1301 | * block from the correct offset. */ |
| 1302 | int r_idx = pRNG(s->frame_cntr, block_idx, size); |
| 1303 | |
| 1304 | for (n = 0; n < size; n++) |
| 1305 | excitation[n] = |
| 1306 | wmavoice_std_codebook[r_idx + n] * s->silence_gain; |
| 1307 | skip_bits(gb, 7 + 1); |
| 1308 | return; |
| 1309 | } |
| 1310 | } else /* FCB_TYPE_EXC_PULSES */ { |
| 1311 | int offset_nbits = 5 - frame_desc->log_n_blocks; |
| 1312 | |
| 1313 | fcb.no_repeat_mask = -1; |
| 1314 | /* similar to ff_decode_10_pulses_35bits(), but with single pulses |
| 1315 | * (instead of double) for a subset of pulses */ |
| 1316 | for (n = 0; n < 5; n++) { |
| 1317 | float sign; |
| 1318 | int pos1, pos2; |
| 1319 | |
| 1320 | sign = get_bits1(gb) ? 1.0 : -1.0; |
| 1321 | pos1 = get_bits(gb, offset_nbits); |
| 1322 | fcb.x[fcb.n] = n + 5 * pos1; |
| 1323 | fcb.y[fcb.n++] = sign; |
| 1324 | if (n < frame_desc->dbl_pulses) { |
| 1325 | pos2 = get_bits(gb, offset_nbits); |
| 1326 | fcb.x[fcb.n] = n + 5 * pos2; |
| 1327 | fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; |
| 1328 | } |
| 1329 | } |
| 1330 | } |
| 1331 | ff_set_fixed_vector(pulses, &fcb, 1.0, size); |
| 1332 | |
| 1333 | /* Calculate gain for adaptive & fixed codebook signal. |
| 1334 | * see ff_amr_set_fixed_gain(). */ |
| 1335 | idx = get_bits(gb, 7); |
| 1336 | fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, |
| 1337 | gain_coeff, 6) - |
| 1338 | 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); |
| 1339 | acb_gain = wmavoice_gain_codebook_acb[idx]; |
| 1340 | pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], |
| 1341 | -2.9957322736 /* log(0.05) */, |
| 1342 | 1.6094379124 /* log(5.0) */); |
| 1343 | |
| 1344 | gain_weight = 8 >> frame_desc->log_n_blocks; |
| 1345 | memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, |
| 1346 | sizeof(*s->gain_pred_err) * (6 - gain_weight)); |
| 1347 | for (n = 0; n < gain_weight; n++) |
| 1348 | s->gain_pred_err[n] = pred_err; |
| 1349 | |
| 1350 | /* Calculation of adaptive codebook */ |
| 1351 | if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
| 1352 | int len; |
| 1353 | for (n = 0; n < size; n += len) { |
| 1354 | int next_idx_sh16; |
| 1355 | int abs_idx = block_idx * size + n; |
| 1356 | int pitch_sh16 = (s->last_pitch_val << 16) + |
| 1357 | s->pitch_diff_sh16 * abs_idx; |
| 1358 | int pitch = (pitch_sh16 + 0x6FFF) >> 16; |
| 1359 | int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; |
| 1360 | idx = idx_sh16 >> 16; |
| 1361 | if (s->pitch_diff_sh16) { |
| 1362 | if (s->pitch_diff_sh16 > 0) { |
| 1363 | next_idx_sh16 = (idx_sh16) &~ 0xFFFF; |
| 1364 | } else |
| 1365 | next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; |
| 1366 | len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, |
| 1367 | 1, size - n); |
| 1368 | } else |
| 1369 | len = size; |
| 1370 | |
| 1371 | ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], |
| 1372 | wmavoice_ipol1_coeffs, 17, |
| 1373 | idx, 9, len); |
| 1374 | } |
| 1375 | } else /* ACB_TYPE_HAMMING */ { |
| 1376 | int block_pitch = block_pitch_sh2 >> 2; |
| 1377 | idx = block_pitch_sh2 & 3; |
| 1378 | if (idx) { |
| 1379 | ff_acelp_interpolatef(excitation, &excitation[-block_pitch], |
| 1380 | wmavoice_ipol2_coeffs, 4, |
| 1381 | idx, 8, size); |
| 1382 | } else |
| 1383 | av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, |
| 1384 | sizeof(float) * size); |
| 1385 | } |
| 1386 | |
| 1387 | /* Interpolate ACB/FCB and use as excitation signal */ |
| 1388 | ff_weighted_vector_sumf(excitation, excitation, pulses, |
| 1389 | acb_gain, fcb_gain, size); |
| 1390 | } |
| 1391 | |
| 1392 | /** |
| 1393 | * Parse data in a single block. |
| 1394 | * @note we assume enough bits are available, caller should check. |
| 1395 | * |
| 1396 | * @param s WMA Voice decoding context private data |
| 1397 | * @param gb bit I/O context |
| 1398 | * @param block_idx index of the to-be-read block |
| 1399 | * @param size amount of samples to be read in this block |
| 1400 | * @param block_pitch_sh2 pitch for this block << 2 |
| 1401 | * @param lsps LSPs for (the end of) this frame |
| 1402 | * @param prev_lsps LSPs for the last frame |
| 1403 | * @param frame_desc frame type descriptor |
| 1404 | * @param excitation target memory for the ACB+FCB interpolated signal |
| 1405 | * @param synth target memory for the speech synthesis filter output |
| 1406 | * @return 0 on success, <0 on error. |
| 1407 | */ |
| 1408 | static void synth_block(WMAVoiceContext *s, GetBitContext *gb, |
| 1409 | int block_idx, int size, |
| 1410 | int block_pitch_sh2, |
| 1411 | const double *lsps, const double *prev_lsps, |
| 1412 | const struct frame_type_desc *frame_desc, |
| 1413 | float *excitation, float *synth) |
| 1414 | { |
| 1415 | double i_lsps[MAX_LSPS]; |
| 1416 | float lpcs[MAX_LSPS]; |
| 1417 | float fac; |
| 1418 | int n; |
| 1419 | |
| 1420 | if (frame_desc->acb_type == ACB_TYPE_NONE) |
| 1421 | synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); |
| 1422 | else |
| 1423 | synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, |
| 1424 | frame_desc, excitation); |
| 1425 | |
| 1426 | /* convert interpolated LSPs to LPCs */ |
| 1427 | fac = (block_idx + 0.5) / frame_desc->n_blocks; |
| 1428 | for (n = 0; n < s->lsps; n++) // LSF -> LSP |
| 1429 | i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); |
| 1430 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); |
| 1431 | |
| 1432 | /* Speech synthesis */ |
| 1433 | ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); |
| 1434 | } |
| 1435 | |
| 1436 | /** |
| 1437 | * Synthesize output samples for a single frame. |
| 1438 | * @note we assume enough bits are available, caller should check. |
| 1439 | * |
| 1440 | * @param ctx WMA Voice decoder context |
| 1441 | * @param gb bit I/O context (s->gb or one for cross-packet superframes) |
| 1442 | * @param frame_idx Frame number within superframe [0-2] |
| 1443 | * @param samples pointer to output sample buffer, has space for at least 160 |
| 1444 | * samples |
| 1445 | * @param lsps LSP array |
| 1446 | * @param prev_lsps array of previous frame's LSPs |
| 1447 | * @param excitation target buffer for excitation signal |
| 1448 | * @param synth target buffer for synthesized speech data |
| 1449 | * @return 0 on success, <0 on error. |
| 1450 | */ |
| 1451 | static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
| 1452 | float *samples, |
| 1453 | const double *lsps, const double *prev_lsps, |
| 1454 | float *excitation, float *synth) |
| 1455 | { |
| 1456 | WMAVoiceContext *s = ctx->priv_data; |
| 1457 | int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val); |
| 1458 | int pitch[MAX_BLOCKS], av_uninit(last_block_pitch); |
| 1459 | |
| 1460 | /* Parse frame type ("frame header"), see frame_descs */ |
| 1461 | int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; |
| 1462 | |
| 1463 | if (bd_idx < 0) { |
| 1464 | av_log(ctx, AV_LOG_ERROR, |
| 1465 | "Invalid frame type VLC code, skipping\n"); |
| 1466 | return AVERROR_INVALIDDATA; |
| 1467 | } |
| 1468 | |
| 1469 | block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; |
| 1470 | |
| 1471 | /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ |
| 1472 | if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { |
| 1473 | /* Pitch is provided per frame, which is interpreted as the pitch of |
| 1474 | * the last sample of the last block of this frame. We can interpolate |
| 1475 | * the pitch of other blocks (and even pitch-per-sample) by gradually |
| 1476 | * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ |
| 1477 | n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; |
| 1478 | log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; |
| 1479 | cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); |
| 1480 | cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); |
| 1481 | if (s->last_acb_type == ACB_TYPE_NONE || |
| 1482 | 20 * abs(cur_pitch_val - s->last_pitch_val) > |
| 1483 | (cur_pitch_val + s->last_pitch_val)) |
| 1484 | s->last_pitch_val = cur_pitch_val; |
| 1485 | |
| 1486 | /* pitch per block */ |
| 1487 | for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
| 1488 | int fac = n * 2 + 1; |
| 1489 | |
| 1490 | pitch[n] = (MUL16(fac, cur_pitch_val) + |
| 1491 | MUL16((n_blocks_x2 - fac), s->last_pitch_val) + |
| 1492 | frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; |
| 1493 | } |
| 1494 | |
| 1495 | /* "pitch-diff-per-sample" for calculation of pitch per sample */ |
| 1496 | s->pitch_diff_sh16 = |
| 1497 | ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; |
| 1498 | } |
| 1499 | |
| 1500 | /* Global gain (if silence) and pitch-adaptive window coordinates */ |
| 1501 | switch (frame_descs[bd_idx].fcb_type) { |
| 1502 | case FCB_TYPE_SILENCE: |
| 1503 | s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; |
| 1504 | break; |
| 1505 | case FCB_TYPE_AW_PULSES: |
| 1506 | aw_parse_coords(s, gb, pitch); |
| 1507 | break; |
| 1508 | } |
| 1509 | |
| 1510 | for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
| 1511 | int bl_pitch_sh2; |
| 1512 | |
| 1513 | /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ |
| 1514 | switch (frame_descs[bd_idx].acb_type) { |
| 1515 | case ACB_TYPE_HAMMING: { |
| 1516 | /* Pitch is given per block. Per-block pitches are encoded as an |
| 1517 | * absolute value for the first block, and then delta values |
| 1518 | * relative to this value) for all subsequent blocks. The scale of |
| 1519 | * this pitch value is semi-logaritmic compared to its use in the |
| 1520 | * decoder, so we convert it to normal scale also. */ |
| 1521 | int block_pitch, |
| 1522 | t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, |
| 1523 | t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, |
| 1524 | t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; |
| 1525 | |
| 1526 | if (n == 0) { |
| 1527 | block_pitch = get_bits(gb, s->block_pitch_nbits); |
| 1528 | } else |
| 1529 | block_pitch = last_block_pitch - s->block_delta_pitch_hrange + |
| 1530 | get_bits(gb, s->block_delta_pitch_nbits); |
| 1531 | /* Convert last_ so that any next delta is within _range */ |
| 1532 | last_block_pitch = av_clip(block_pitch, |
| 1533 | s->block_delta_pitch_hrange, |
| 1534 | s->block_pitch_range - |
| 1535 | s->block_delta_pitch_hrange); |
| 1536 | |
| 1537 | /* Convert semi-log-style scale back to normal scale */ |
| 1538 | if (block_pitch < t1) { |
| 1539 | bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; |
| 1540 | } else { |
| 1541 | block_pitch -= t1; |
| 1542 | if (block_pitch < t2) { |
| 1543 | bl_pitch_sh2 = |
| 1544 | (s->block_conv_table[1] << 2) + (block_pitch << 1); |
| 1545 | } else { |
| 1546 | block_pitch -= t2; |
| 1547 | if (block_pitch < t3) { |
| 1548 | bl_pitch_sh2 = |
| 1549 | (s->block_conv_table[2] + block_pitch) << 2; |
| 1550 | } else |
| 1551 | bl_pitch_sh2 = s->block_conv_table[3] << 2; |
| 1552 | } |
| 1553 | } |
| 1554 | pitch[n] = bl_pitch_sh2 >> 2; |
| 1555 | break; |
| 1556 | } |
| 1557 | |
| 1558 | case ACB_TYPE_ASYMMETRIC: { |
| 1559 | bl_pitch_sh2 = pitch[n] << 2; |
| 1560 | break; |
| 1561 | } |
| 1562 | |
| 1563 | default: // ACB_TYPE_NONE has no pitch |
| 1564 | bl_pitch_sh2 = 0; |
| 1565 | break; |
| 1566 | } |
| 1567 | |
| 1568 | synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, |
| 1569 | lsps, prev_lsps, &frame_descs[bd_idx], |
| 1570 | &excitation[n * block_nsamples], |
| 1571 | &synth[n * block_nsamples]); |
| 1572 | } |
| 1573 | |
| 1574 | /* Averaging projection filter, if applicable. Else, just copy samples |
| 1575 | * from synthesis buffer */ |
| 1576 | if (s->do_apf) { |
| 1577 | double i_lsps[MAX_LSPS]; |
| 1578 | float lpcs[MAX_LSPS]; |
| 1579 | |
| 1580 | for (n = 0; n < s->lsps; n++) // LSF -> LSP |
| 1581 | i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); |
| 1582 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); |
| 1583 | postfilter(s, synth, samples, 80, lpcs, |
| 1584 | &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], |
| 1585 | frame_descs[bd_idx].fcb_type, pitch[0]); |
| 1586 | |
| 1587 | for (n = 0; n < s->lsps; n++) // LSF -> LSP |
| 1588 | i_lsps[n] = cos(lsps[n]); |
| 1589 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); |
| 1590 | postfilter(s, &synth[80], &samples[80], 80, lpcs, |
| 1591 | &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], |
| 1592 | frame_descs[bd_idx].fcb_type, pitch[0]); |
| 1593 | } else |
| 1594 | memcpy(samples, synth, 160 * sizeof(synth[0])); |
| 1595 | |
| 1596 | /* Cache values for next frame */ |
| 1597 | s->frame_cntr++; |
| 1598 | if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) |
| 1599 | s->last_acb_type = frame_descs[bd_idx].acb_type; |
| 1600 | switch (frame_descs[bd_idx].acb_type) { |
| 1601 | case ACB_TYPE_NONE: |
| 1602 | s->last_pitch_val = 0; |
| 1603 | break; |
| 1604 | case ACB_TYPE_ASYMMETRIC: |
| 1605 | s->last_pitch_val = cur_pitch_val; |
| 1606 | break; |
| 1607 | case ACB_TYPE_HAMMING: |
| 1608 | s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; |
| 1609 | break; |
| 1610 | } |
| 1611 | |
| 1612 | return 0; |
| 1613 | } |
| 1614 | |
| 1615 | /** |
| 1616 | * Ensure minimum value for first item, maximum value for last value, |
| 1617 | * proper spacing between each value and proper ordering. |
| 1618 | * |
| 1619 | * @param lsps array of LSPs |
| 1620 | * @param num size of LSP array |
| 1621 | * |
| 1622 | * @note basically a double version of #ff_acelp_reorder_lsf(), might be |
| 1623 | * useful to put in a generic location later on. Parts are also |
| 1624 | * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), |
| 1625 | * which is in float. |
| 1626 | */ |
| 1627 | static void stabilize_lsps(double *lsps, int num) |
| 1628 | { |
| 1629 | int n, m, l; |
| 1630 | |
| 1631 | /* set minimum value for first, maximum value for last and minimum |
| 1632 | * spacing between LSF values. |
| 1633 | * Very similar to ff_set_min_dist_lsf(), but in double. */ |
| 1634 | lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); |
| 1635 | for (n = 1; n < num; n++) |
| 1636 | lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); |
| 1637 | lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); |
| 1638 | |
| 1639 | /* reorder (looks like one-time / non-recursed bubblesort). |
| 1640 | * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ |
| 1641 | for (n = 1; n < num; n++) { |
| 1642 | if (lsps[n] < lsps[n - 1]) { |
| 1643 | for (m = 1; m < num; m++) { |
| 1644 | double tmp = lsps[m]; |
| 1645 | for (l = m - 1; l >= 0; l--) { |
| 1646 | if (lsps[l] <= tmp) break; |
| 1647 | lsps[l + 1] = lsps[l]; |
| 1648 | } |
| 1649 | lsps[l + 1] = tmp; |
| 1650 | } |
| 1651 | break; |
| 1652 | } |
| 1653 | } |
| 1654 | } |
| 1655 | |
| 1656 | /** |
| 1657 | * Test if there's enough bits to read 1 superframe. |
| 1658 | * |
| 1659 | * @param orig_gb bit I/O context used for reading. This function |
| 1660 | * does not modify the state of the bitreader; it |
| 1661 | * only uses it to copy the current stream position |
| 1662 | * @param s WMA Voice decoding context private data |
| 1663 | * @return < 0 on error, 1 on not enough bits or 0 if OK. |
| 1664 | */ |
| 1665 | static int check_bits_for_superframe(GetBitContext *orig_gb, |
| 1666 | WMAVoiceContext *s) |
| 1667 | { |
| 1668 | GetBitContext s_gb, *gb = &s_gb; |
| 1669 | int n, need_bits, bd_idx; |
| 1670 | const struct frame_type_desc *frame_desc; |
| 1671 | |
| 1672 | /* initialize a copy */ |
| 1673 | init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); |
| 1674 | skip_bits_long(gb, get_bits_count(orig_gb)); |
| 1675 | av_assert1(get_bits_left(gb) == get_bits_left(orig_gb)); |
| 1676 | |
| 1677 | /* superframe header */ |
| 1678 | if (get_bits_left(gb) < 14) |
| 1679 | return 1; |
| 1680 | if (!get_bits1(gb)) |
| 1681 | return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe |
| 1682 | if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe |
| 1683 | if (s->has_residual_lsps) { // residual LSPs (for all frames) |
| 1684 | if (get_bits_left(gb) < s->sframe_lsp_bitsize) |
| 1685 | return 1; |
| 1686 | skip_bits_long(gb, s->sframe_lsp_bitsize); |
| 1687 | } |
| 1688 | |
| 1689 | /* frames */ |
| 1690 | for (n = 0; n < MAX_FRAMES; n++) { |
| 1691 | int aw_idx_is_ext = 0; |
| 1692 | |
| 1693 | if (!s->has_residual_lsps) { // independent LSPs (per-frame) |
| 1694 | if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; |
| 1695 | skip_bits_long(gb, s->frame_lsp_bitsize); |
| 1696 | } |
| 1697 | bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; |
| 1698 | if (bd_idx < 0) |
| 1699 | return AVERROR_INVALIDDATA; // invalid frame type VLC code |
| 1700 | frame_desc = &frame_descs[bd_idx]; |
| 1701 | if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
| 1702 | if (get_bits_left(gb) < s->pitch_nbits) |
| 1703 | return 1; |
| 1704 | skip_bits_long(gb, s->pitch_nbits); |
| 1705 | } |
| 1706 | if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
| 1707 | skip_bits(gb, 8); |
| 1708 | } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| 1709 | int tmp = get_bits(gb, 6); |
| 1710 | if (tmp >= 0x36) { |
| 1711 | skip_bits(gb, 2); |
| 1712 | aw_idx_is_ext = 1; |
| 1713 | } |
| 1714 | } |
| 1715 | |
| 1716 | /* blocks */ |
| 1717 | if (frame_desc->acb_type == ACB_TYPE_HAMMING) { |
| 1718 | need_bits = s->block_pitch_nbits + |
| 1719 | (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; |
| 1720 | } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
| 1721 | need_bits = 2 * !aw_idx_is_ext; |
| 1722 | } else |
| 1723 | need_bits = 0; |
| 1724 | need_bits += frame_desc->frame_size; |
| 1725 | if (get_bits_left(gb) < need_bits) |
| 1726 | return 1; |
| 1727 | skip_bits_long(gb, need_bits); |
| 1728 | } |
| 1729 | |
| 1730 | return 0; |
| 1731 | } |
| 1732 | |
| 1733 | /** |
| 1734 | * Synthesize output samples for a single superframe. If we have any data |
| 1735 | * cached in s->sframe_cache, that will be used instead of whatever is loaded |
| 1736 | * in s->gb. |
| 1737 | * |
| 1738 | * WMA Voice superframes contain 3 frames, each containing 160 audio samples, |
| 1739 | * to give a total of 480 samples per frame. See #synth_frame() for frame |
| 1740 | * parsing. In addition to 3 frames, superframes can also contain the LSPs |
| 1741 | * (if these are globally specified for all frames (residually); they can |
| 1742 | * also be specified individually per-frame. See the s->has_residual_lsps |
| 1743 | * option), and can specify the number of samples encoded in this superframe |
| 1744 | * (if less than 480), usually used to prevent blanks at track boundaries. |
| 1745 | * |
| 1746 | * @param ctx WMA Voice decoder context |
| 1747 | * @return 0 on success, <0 on error or 1 if there was not enough data to |
| 1748 | * fully parse the superframe |
| 1749 | */ |
| 1750 | static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, |
| 1751 | int *got_frame_ptr) |
| 1752 | { |
| 1753 | WMAVoiceContext *s = ctx->priv_data; |
| 1754 | GetBitContext *gb = &s->gb, s_gb; |
| 1755 | int n, res, n_samples = 480; |
| 1756 | double lsps[MAX_FRAMES][MAX_LSPS]; |
| 1757 | const double *mean_lsf = s->lsps == 16 ? |
| 1758 | wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; |
| 1759 | float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; |
| 1760 | float synth[MAX_LSPS + MAX_SFRAMESIZE]; |
| 1761 | float *samples; |
| 1762 | |
| 1763 | memcpy(synth, s->synth_history, |
| 1764 | s->lsps * sizeof(*synth)); |
| 1765 | memcpy(excitation, s->excitation_history, |
| 1766 | s->history_nsamples * sizeof(*excitation)); |
| 1767 | |
| 1768 | if (s->sframe_cache_size > 0) { |
| 1769 | gb = &s_gb; |
| 1770 | init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); |
| 1771 | s->sframe_cache_size = 0; |
| 1772 | } |
| 1773 | |
| 1774 | if ((res = check_bits_for_superframe(gb, s)) == 1) { |
| 1775 | *got_frame_ptr = 0; |
| 1776 | return 1; |
| 1777 | } else if (res < 0) |
| 1778 | return res; |
| 1779 | |
| 1780 | /* First bit is speech/music bit, it differentiates between WMAVoice |
| 1781 | * speech samples (the actual codec) and WMAVoice music samples, which |
| 1782 | * are really WMAPro-in-WMAVoice-superframes. I've never seen those in |
| 1783 | * the wild yet. */ |
| 1784 | if (!get_bits1(gb)) { |
| 1785 | avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); |
| 1786 | return AVERROR_PATCHWELCOME; |
| 1787 | } |
| 1788 | |
| 1789 | /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ |
| 1790 | if (get_bits1(gb)) { |
| 1791 | if ((n_samples = get_bits(gb, 12)) > 480) { |
| 1792 | av_log(ctx, AV_LOG_ERROR, |
| 1793 | "Superframe encodes >480 samples (%d), not allowed\n", |
| 1794 | n_samples); |
| 1795 | return AVERROR_INVALIDDATA; |
| 1796 | } |
| 1797 | } |
| 1798 | /* Parse LSPs, if global for the superframe (can also be per-frame). */ |
| 1799 | if (s->has_residual_lsps) { |
| 1800 | double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; |
| 1801 | |
| 1802 | for (n = 0; n < s->lsps; n++) |
| 1803 | prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; |
| 1804 | |
| 1805 | if (s->lsps == 10) { |
| 1806 | dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
| 1807 | } else /* s->lsps == 16 */ |
| 1808 | dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); |
| 1809 | |
| 1810 | for (n = 0; n < s->lsps; n++) { |
| 1811 | lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); |
| 1812 | lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); |
| 1813 | lsps[2][n] += mean_lsf[n]; |
| 1814 | } |
| 1815 | for (n = 0; n < 3; n++) |
| 1816 | stabilize_lsps(lsps[n], s->lsps); |
| 1817 | } |
| 1818 | |
| 1819 | /* get output buffer */ |
| 1820 | frame->nb_samples = 480; |
| 1821 | if ((res = ff_get_buffer(ctx, frame, 0)) < 0) |
| 1822 | return res; |
| 1823 | frame->nb_samples = n_samples; |
| 1824 | samples = (float *)frame->data[0]; |
| 1825 | |
| 1826 | /* Parse frames, optionally preceded by per-frame (independent) LSPs. */ |
| 1827 | for (n = 0; n < 3; n++) { |
| 1828 | if (!s->has_residual_lsps) { |
| 1829 | int m; |
| 1830 | |
| 1831 | if (s->lsps == 10) { |
| 1832 | dequant_lsp10i(gb, lsps[n]); |
| 1833 | } else /* s->lsps == 16 */ |
| 1834 | dequant_lsp16i(gb, lsps[n]); |
| 1835 | |
| 1836 | for (m = 0; m < s->lsps; m++) |
| 1837 | lsps[n][m] += mean_lsf[m]; |
| 1838 | stabilize_lsps(lsps[n], s->lsps); |
| 1839 | } |
| 1840 | |
| 1841 | if ((res = synth_frame(ctx, gb, n, |
| 1842 | &samples[n * MAX_FRAMESIZE], |
| 1843 | lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], |
| 1844 | &excitation[s->history_nsamples + n * MAX_FRAMESIZE], |
| 1845 | &synth[s->lsps + n * MAX_FRAMESIZE]))) { |
| 1846 | *got_frame_ptr = 0; |
| 1847 | return res; |
| 1848 | } |
| 1849 | } |
| 1850 | |
| 1851 | /* Statistics? FIXME - we don't check for length, a slight overrun |
| 1852 | * will be caught by internal buffer padding, and anything else |
| 1853 | * will be skipped, not read. */ |
| 1854 | if (get_bits1(gb)) { |
| 1855 | res = get_bits(gb, 4); |
| 1856 | skip_bits(gb, 10 * (res + 1)); |
| 1857 | } |
| 1858 | |
| 1859 | *got_frame_ptr = 1; |
| 1860 | |
| 1861 | /* Update history */ |
| 1862 | memcpy(s->prev_lsps, lsps[2], |
| 1863 | s->lsps * sizeof(*s->prev_lsps)); |
| 1864 | memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], |
| 1865 | s->lsps * sizeof(*synth)); |
| 1866 | memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], |
| 1867 | s->history_nsamples * sizeof(*excitation)); |
| 1868 | if (s->do_apf) |
| 1869 | memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], |
| 1870 | s->history_nsamples * sizeof(*s->zero_exc_pf)); |
| 1871 | |
| 1872 | return 0; |
| 1873 | } |
| 1874 | |
| 1875 | /** |
| 1876 | * Parse the packet header at the start of each packet (input data to this |
| 1877 | * decoder). |
| 1878 | * |
| 1879 | * @param s WMA Voice decoding context private data |
| 1880 | * @return 1 if not enough bits were available, or 0 on success. |
| 1881 | */ |
| 1882 | static int parse_packet_header(WMAVoiceContext *s) |
| 1883 | { |
| 1884 | GetBitContext *gb = &s->gb; |
| 1885 | unsigned int res; |
| 1886 | |
| 1887 | if (get_bits_left(gb) < 11) |
| 1888 | return 1; |
| 1889 | skip_bits(gb, 4); // packet sequence number |
| 1890 | s->has_residual_lsps = get_bits1(gb); |
| 1891 | do { |
| 1892 | res = get_bits(gb, 6); // number of superframes per packet |
| 1893 | // (minus first one if there is spillover) |
| 1894 | if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) |
| 1895 | return 1; |
| 1896 | } while (res == 0x3F); |
| 1897 | s->spillover_nbits = get_bits(gb, s->spillover_bitsize); |
| 1898 | |
| 1899 | return 0; |
| 1900 | } |
| 1901 | |
| 1902 | /** |
| 1903 | * Copy (unaligned) bits from gb/data/size to pb. |
| 1904 | * |
| 1905 | * @param pb target buffer to copy bits into |
| 1906 | * @param data source buffer to copy bits from |
| 1907 | * @param size size of the source data, in bytes |
| 1908 | * @param gb bit I/O context specifying the current position in the source. |
| 1909 | * data. This function might use this to align the bit position to |
| 1910 | * a whole-byte boundary before calling #avpriv_copy_bits() on aligned |
| 1911 | * source data |
| 1912 | * @param nbits the amount of bits to copy from source to target |
| 1913 | * |
| 1914 | * @note after calling this function, the current position in the input bit |
| 1915 | * I/O context is undefined. |
| 1916 | */ |
| 1917 | static void copy_bits(PutBitContext *pb, |
| 1918 | const uint8_t *data, int size, |
| 1919 | GetBitContext *gb, int nbits) |
| 1920 | { |
| 1921 | int rmn_bytes, rmn_bits; |
| 1922 | |
| 1923 | rmn_bits = rmn_bytes = get_bits_left(gb); |
| 1924 | if (rmn_bits < nbits) |
| 1925 | return; |
| 1926 | if (nbits > pb->size_in_bits - put_bits_count(pb)) |
| 1927 | return; |
| 1928 | rmn_bits &= 7; rmn_bytes >>= 3; |
| 1929 | if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) |
| 1930 | put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); |
| 1931 | avpriv_copy_bits(pb, data + size - rmn_bytes, |
| 1932 | FFMIN(nbits - rmn_bits, rmn_bytes << 3)); |
| 1933 | } |
| 1934 | |
| 1935 | /** |
| 1936 | * Packet decoding: a packet is anything that the (ASF) demuxer contains, |
| 1937 | * and we expect that the demuxer / application provides it to us as such |
| 1938 | * (else you'll probably get garbage as output). Every packet has a size of |
| 1939 | * ctx->block_align bytes, starts with a packet header (see |
| 1940 | * #parse_packet_header()), and then a series of superframes. Superframe |
| 1941 | * boundaries may exceed packets, i.e. superframes can split data over |
| 1942 | * multiple (two) packets. |
| 1943 | * |
| 1944 | * For more information about frames, see #synth_superframe(). |
| 1945 | */ |
| 1946 | static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, |
| 1947 | int *got_frame_ptr, AVPacket *avpkt) |
| 1948 | { |
| 1949 | WMAVoiceContext *s = ctx->priv_data; |
| 1950 | GetBitContext *gb = &s->gb; |
| 1951 | int size, res, pos; |
| 1952 | |
| 1953 | /* Packets are sometimes a multiple of ctx->block_align, with a packet |
| 1954 | * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer |
| 1955 | * feeds us ASF packets, which may concatenate multiple "codec" packets |
| 1956 | * in a single "muxer" packet, so we artificially emulate that by |
| 1957 | * capping the packet size at ctx->block_align. */ |
| 1958 | for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); |
| 1959 | if (!size) { |
| 1960 | *got_frame_ptr = 0; |
| 1961 | return 0; |
| 1962 | } |
| 1963 | init_get_bits(&s->gb, avpkt->data, size << 3); |
| 1964 | |
| 1965 | /* size == ctx->block_align is used to indicate whether we are dealing with |
| 1966 | * a new packet or a packet of which we already read the packet header |
| 1967 | * previously. */ |
| 1968 | if (size == ctx->block_align) { // new packet header |
| 1969 | if ((res = parse_packet_header(s)) < 0) |
| 1970 | return res; |
| 1971 | |
| 1972 | /* If the packet header specifies a s->spillover_nbits, then we want |
| 1973 | * to push out all data of the previous packet (+ spillover) before |
| 1974 | * continuing to parse new superframes in the current packet. */ |
| 1975 | if (s->spillover_nbits > 0) { |
| 1976 | if (s->sframe_cache_size > 0) { |
| 1977 | int cnt = get_bits_count(gb); |
| 1978 | copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); |
| 1979 | flush_put_bits(&s->pb); |
| 1980 | s->sframe_cache_size += s->spillover_nbits; |
| 1981 | if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 && |
| 1982 | *got_frame_ptr) { |
| 1983 | cnt += s->spillover_nbits; |
| 1984 | s->skip_bits_next = cnt & 7; |
| 1985 | return cnt >> 3; |
| 1986 | } else |
| 1987 | skip_bits_long (gb, s->spillover_nbits - cnt + |
| 1988 | get_bits_count(gb)); // resync |
| 1989 | } else |
| 1990 | skip_bits_long(gb, s->spillover_nbits); // resync |
| 1991 | } |
| 1992 | } else if (s->skip_bits_next) |
| 1993 | skip_bits(gb, s->skip_bits_next); |
| 1994 | |
| 1995 | /* Try parsing superframes in current packet */ |
| 1996 | s->sframe_cache_size = 0; |
| 1997 | s->skip_bits_next = 0; |
| 1998 | pos = get_bits_left(gb); |
| 1999 | if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { |
| 2000 | return res; |
| 2001 | } else if (*got_frame_ptr) { |
| 2002 | int cnt = get_bits_count(gb); |
| 2003 | s->skip_bits_next = cnt & 7; |
| 2004 | return cnt >> 3; |
| 2005 | } else if ((s->sframe_cache_size = pos) > 0) { |
| 2006 | /* rewind bit reader to start of last (incomplete) superframe... */ |
| 2007 | init_get_bits(gb, avpkt->data, size << 3); |
| 2008 | skip_bits_long(gb, (size << 3) - pos); |
| 2009 | av_assert1(get_bits_left(gb) == pos); |
| 2010 | |
| 2011 | /* ...and cache it for spillover in next packet */ |
| 2012 | init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); |
| 2013 | copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); |
| 2014 | // FIXME bad - just copy bytes as whole and add use the |
| 2015 | // skip_bits_next field |
| 2016 | } |
| 2017 | |
| 2018 | return size; |
| 2019 | } |
| 2020 | |
| 2021 | static av_cold int wmavoice_decode_end(AVCodecContext *ctx) |
| 2022 | { |
| 2023 | WMAVoiceContext *s = ctx->priv_data; |
| 2024 | |
| 2025 | if (s->do_apf) { |
| 2026 | ff_rdft_end(&s->rdft); |
| 2027 | ff_rdft_end(&s->irdft); |
| 2028 | ff_dct_end(&s->dct); |
| 2029 | ff_dct_end(&s->dst); |
| 2030 | } |
| 2031 | |
| 2032 | return 0; |
| 2033 | } |
| 2034 | |
| 2035 | static av_cold void wmavoice_flush(AVCodecContext *ctx) |
| 2036 | { |
| 2037 | WMAVoiceContext *s = ctx->priv_data; |
| 2038 | int n; |
| 2039 | |
| 2040 | s->postfilter_agc = 0; |
| 2041 | s->sframe_cache_size = 0; |
| 2042 | s->skip_bits_next = 0; |
| 2043 | for (n = 0; n < s->lsps; n++) |
| 2044 | s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); |
| 2045 | memset(s->excitation_history, 0, |
| 2046 | sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); |
| 2047 | memset(s->synth_history, 0, |
| 2048 | sizeof(*s->synth_history) * MAX_LSPS); |
| 2049 | memset(s->gain_pred_err, 0, |
| 2050 | sizeof(s->gain_pred_err)); |
| 2051 | |
| 2052 | if (s->do_apf) { |
| 2053 | memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, |
| 2054 | sizeof(*s->synth_filter_out_buf) * s->lsps); |
| 2055 | memset(s->dcf_mem, 0, |
| 2056 | sizeof(*s->dcf_mem) * 2); |
| 2057 | memset(s->zero_exc_pf, 0, |
| 2058 | sizeof(*s->zero_exc_pf) * s->history_nsamples); |
| 2059 | memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); |
| 2060 | } |
| 2061 | } |
| 2062 | |
| 2063 | AVCodec ff_wmavoice_decoder = { |
| 2064 | .name = "wmavoice", |
| 2065 | .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), |
| 2066 | .type = AVMEDIA_TYPE_AUDIO, |
| 2067 | .id = AV_CODEC_ID_WMAVOICE, |
| 2068 | .priv_data_size = sizeof(WMAVoiceContext), |
| 2069 | .init = wmavoice_decode_init, |
| 2070 | .init_static_data = wmavoice_init_static_data, |
| 2071 | .close = wmavoice_decode_end, |
| 2072 | .decode = wmavoice_decode_packet, |
| 2073 | .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, |
| 2074 | .flush = wmavoice_flush, |
| 2075 | }; |