cf9884108c1152d424a115b94a724f91082d93fc
[libav.git] / libavcodec / wmavoice.c
1 /*
2 * Windows Media Audio Voice decoder.
3 * Copyright (c) 2009 Ronald S. Bultje
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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
34 #include "avcodec.h"
35 #include "bitstream.h"
36 #include "internal.h"
37 #include "put_bits.h"
38 #include "wmavoice_data.h"
39 #include "celp_filters.h"
40 #include "acelp_vectors.h"
41 #include "acelp_filters.h"
42 #include "lsp.h"
43 #include "dct.h"
44 #include "rdft.h"
45 #include "sinewin.h"
46
47 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame
48 #define MAX_LSPS 16 ///< maximum filter order
49 #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple
50 ///< of 16 for ASM input buffer alignment
51 #define MAX_FRAMES 3 ///< maximum number of frames per superframe
52 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame
53 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history
54 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
55 ///< maximum number of samples per superframe
56 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
57 ///< was split over two packets
58 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration
59
60 /**
61 * Frame type VLC coding.
62 */
63 static VLC frame_type_vlc;
64
65 /**
66 * Adaptive codebook types.
67 */
68 enum {
69 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed)
70 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
71 ///< we interpolate to get a per-sample pitch.
72 ///< Signal is generated using an asymmetric sinc
73 ///< window function
74 ///< @note see #wmavoice_ipol1_coeffs
75 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using
76 ///< a Hamming sinc window function
77 ///< @note see #wmavoice_ipol2_coeffs
78 };
79
80 /**
81 * Fixed codebook types.
82 */
83 enum {
84 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence
85 ///< generated from a hardcoded (fixed) codebook
86 ///< with per-frame (low) gain values
87 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block
88 ///< gain values
89 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals,
90 ///< used in particular for low-bitrate streams
91 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
92 ///< combinations of either single pulses or
93 ///< pulse pairs
94 };
95
96 /**
97 * Description of frame types.
98 */
99 static const struct frame_type_desc {
100 uint8_t n_blocks; ///< amount of blocks per frame (each block
101 ///< (contains 160/#n_blocks samples)
102 uint8_t log_n_blocks; ///< log2(#n_blocks)
103 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*)
104 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*)
105 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs
106 ///< (rather than just one single pulse)
107 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
108 uint16_t frame_size; ///< the amount of bits that make up the block
109 ///< data (per frame)
110 } frame_descs[17] = {
111 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 },
112 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 },
113 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 },
114 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 },
115 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
116 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
117 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
118 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
119 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 },
120 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 },
121 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 },
122 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 },
123 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 },
124 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 },
125 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 },
126 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 },
127 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 }
128 };
129
130 /**
131 * WMA Voice decoding context.
132 */
133 typedef struct WMAVoiceContext {
134 /**
135 * @name Global values specified in the stream header / extradata or used all over.
136 * @{
137 */
138 BitstreamContext bc; ///< packet bitreader. During decoder init,
139 ///< it contains the extradata from the
140 ///< demuxer. During decoding, it contains
141 ///< packet data.
142 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
143
144 int spillover_bitsize; ///< number of bits used to specify
145 ///< #spillover_nbits in the packet header
146 ///< = ceil(log2(ctx->block_align << 3))
147 int history_nsamples; ///< number of samples in history for signal
148 ///< prediction (through ACB)
149
150 /* postfilter specific values */
151 int do_apf; ///< whether to apply the averaged
152 ///< projection filter (APF)
153 int denoise_strength; ///< strength of denoising in Wiener filter
154 ///< [0-11]
155 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
156 ///< Wiener filter coefficients (postfilter)
157 int dc_level; ///< Predicted amount of DC noise, based
158 ///< on which a DC removal filter is used
159
160 int lsps; ///< number of LSPs per frame [10 or 16]
161 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
162 int lsp_def_mode; ///< defines different sets of LSP defaults
163 ///< [0, 1]
164 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
165 ///< per-frame (independent coding)
166 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
167 ///< per superframe (residual coding)
168
169 int min_pitch_val; ///< base value for pitch parsing code
170 int max_pitch_val; ///< max value + 1 for pitch parsing
171 int pitch_nbits; ///< number of bits used to specify the
172 ///< pitch value in the frame header
173 int block_pitch_nbits; ///< number of bits used to specify the
174 ///< first block's pitch value
175 int block_pitch_range; ///< range of the block pitch
176 int block_delta_pitch_nbits; ///< number of bits used to specify the
177 ///< delta pitch between this and the last
178 ///< block's pitch value, used in all but
179 ///< first block
180 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
181 ///< from -this to +this-1)
182 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
183 ///< conversion
184
185 /**
186 * @}
187 *
188 * @name Packet values specified in the packet header or related to a packet.
189 *
190 * A packet is considered to be a single unit of data provided to this
191 * decoder by the demuxer.
192 * @{
193 */
194 int spillover_nbits; ///< number of bits of the previous packet's
195 ///< last superframe preceding this
196 ///< packet's first full superframe (useful
197 ///< for re-synchronization also)
198 int has_residual_lsps; ///< if set, superframes contain one set of
199 ///< LSPs that cover all frames, encoded as
200 ///< independent and residual LSPs; if not
201 ///< set, each frame contains its own, fully
202 ///< independent, LSPs
203 int skip_bits_next; ///< number of bits to skip at the next call
204 ///< to #wmavoice_decode_packet() (since
205 ///< they're part of the previous superframe)
206
207 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + AV_INPUT_BUFFER_PADDING_SIZE];
208 ///< cache for superframe data split over
209 ///< multiple packets
210 int sframe_cache_size; ///< set to >0 if we have data from an
211 ///< (incomplete) superframe from a previous
212 ///< packet that spilled over in the current
213 ///< packet; specifies the amount of bits in
214 ///< #sframe_cache
215 PutBitContext pb; ///< bitstream writer for #sframe_cache
216
217 /**
218 * @}
219 *
220 * @name Frame and superframe values
221 * Superframe and frame data - these can change from frame to frame,
222 * although some of them do in that case serve as a cache / history for
223 * the next frame or superframe.
224 * @{
225 */
226 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
227 ///< superframe
228 int last_pitch_val; ///< pitch value of the previous frame
229 int last_acb_type; ///< frame type [0-2] of the previous frame
230 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
231 ///< << 16) / #MAX_FRAMESIZE
232 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
233
234 int aw_idx_is_ext; ///< whether the AW index was encoded in
235 ///< 8 bits (instead of 6)
236 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
237 ///< can apply the pulse, relative to the
238 ///< value in aw_first_pulse_off. The exact
239 ///< position of the first AW-pulse is within
240 ///< [pulse_off, pulse_off + this], and
241 ///< depends on bitstream values; [16 or 24]
242 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
243 ///< that this number can be negative (in
244 ///< which case it basically means "zero")
245 int aw_first_pulse_off[2]; ///< index of first sample to which to
246 ///< apply AW-pulses, or -0xff if unset
247 int aw_next_pulse_off_cache; ///< the position (relative to start of the
248 ///< second block) at which pulses should
249 ///< start to be positioned, serves as a
250 ///< cache for pitch-adaptive window pulses
251 ///< between blocks
252
253 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
254 ///< only used for comfort noise in #pRNG()
255 float gain_pred_err[6]; ///< cache for gain prediction
256 float excitation_history[MAX_SIGNAL_HISTORY];
257 ///< cache of the signal of previous
258 ///< superframes, used as a history for
259 ///< signal generation
260 float synth_history[MAX_LSPS]; ///< see #excitation_history
261 /**
262 * @}
263 *
264 * @name Postfilter values
265 *
266 * Variables used for postfilter implementation, mostly history for
267 * smoothing and so on, and context variables for FFT/iFFT.
268 * @{
269 */
270 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
271 ///< postfilter (for denoise filter)
272 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
273 ///< transform, part of postfilter)
274 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
275 ///< range
276 float postfilter_agc; ///< gain control memory, used in
277 ///< #adaptive_gain_control()
278 float dcf_mem[2]; ///< DC filter history
279 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
280 ///< zero filter output (i.e. excitation)
281 ///< by postfilter
282 float denoise_filter_cache[MAX_FRAMESIZE];
283 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
284 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
285 ///< aligned buffer for LPC tilting
286 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
287 ///< aligned buffer for denoise coefficients
288 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
289 ///< aligned buffer for postfilter speech
290 ///< synthesis
291 /**
292 * @}
293 */
294 } WMAVoiceContext;
295
296 /**
297 * Set up the variable bit mode (VBM) tree from container extradata.
298 * @param bc bit I/O context.
299 * The bit context (s->bc) should be loaded with byte 23-46 of the
300 * container extradata (i.e. the ones containing the VBM tree).
301 * @param vbm_tree pointer to array to which the decoded VBM tree will be
302 * written.
303 * @return 0 on success, <0 on error.
304 */
305 static av_cold int decode_vbmtree(BitstreamContext *bc, int8_t vbm_tree[25])
306 {
307 int cntr[8] = { 0 }, n, res;
308
309 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
310 for (n = 0; n < 17; n++) {
311 res = bitstream_read(bc, 3);
312 if (cntr[res] > 3) // should be >= 3 + (res == 7))
313 return -1;
314 vbm_tree[res * 3 + cntr[res]++] = n;
315 }
316 return 0;
317 }
318
319 static av_cold void wmavoice_init_static_data(AVCodec *codec)
320 {
321 static const uint8_t bits[] = {
322 2, 2, 2, 4, 4, 4,
323 6, 6, 6, 8, 8, 8,
324 10, 10, 10, 12, 12, 12,
325 14, 14, 14, 14
326 };
327 static const uint16_t codes[] = {
328 0x0000, 0x0001, 0x0002, // 00/01/10
329 0x000c, 0x000d, 0x000e, // 11+00/01/10
330 0x003c, 0x003d, 0x003e, // 1111+00/01/10
331 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
332 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
333 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
334 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
335 };
336
337 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
338 bits, 1, 1, codes, 2, 2, 132);
339 }
340
341 /**
342 * Set up decoder with parameters from demuxer (extradata etc.).
343 */
344 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
345 {
346 int n, flags, pitch_range, lsp16_flag;
347 WMAVoiceContext *s = ctx->priv_data;
348
349 /**
350 * Extradata layout:
351 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
352 * - byte 19-22: flags field (annoyingly in LE; see below for known
353 * values),
354 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
355 * rest is 0).
356 */
357 if (ctx->extradata_size != 46) {
358 av_log(ctx, AV_LOG_ERROR,
359 "Invalid extradata size %d (should be 46)\n",
360 ctx->extradata_size);
361 return AVERROR_INVALIDDATA;
362 }
363 flags = AV_RL32(ctx->extradata + 18);
364 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
365 s->do_apf = flags & 0x1;
366 if (s->do_apf) {
367 ff_rdft_init(&s->rdft, 7, DFT_R2C);
368 ff_rdft_init(&s->irdft, 7, IDFT_C2R);
369 ff_dct_init(&s->dct, 6, DCT_I);
370 ff_dct_init(&s->dst, 6, DST_I);
371
372 ff_sine_window_init(s->cos, 256);
373 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
374 for (n = 0; n < 255; n++) {
375 s->sin[n] = -s->sin[510 - n];
376 s->cos[510 - n] = s->cos[n];
377 }
378 }
379 s->denoise_strength = (flags >> 2) & 0xF;
380 if (s->denoise_strength >= 12) {
381 av_log(ctx, AV_LOG_ERROR,
382 "Invalid denoise filter strength %d (max=11)\n",
383 s->denoise_strength);
384 return AVERROR_INVALIDDATA;
385 }
386 s->denoise_tilt_corr = !!(flags & 0x40);
387 s->dc_level = (flags >> 7) & 0xF;
388 s->lsp_q_mode = !!(flags & 0x2000);
389 s->lsp_def_mode = !!(flags & 0x4000);
390 lsp16_flag = flags & 0x1000;
391 if (lsp16_flag) {
392 s->lsps = 16;
393 s->frame_lsp_bitsize = 34;
394 s->sframe_lsp_bitsize = 60;
395 } else {
396 s->lsps = 10;
397 s->frame_lsp_bitsize = 24;
398 s->sframe_lsp_bitsize = 48;
399 }
400 for (n = 0; n < s->lsps; n++)
401 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
402
403 bitstream_init8(&s->bc, ctx->extradata + 22, ctx->extradata_size - 22);
404 if (decode_vbmtree(&s->bc, s->vbm_tree) < 0) {
405 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
406 return AVERROR_INVALIDDATA;
407 }
408
409 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
410 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
411 pitch_range = s->max_pitch_val - s->min_pitch_val;
412 if (pitch_range <= 0) {
413 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
414 return AVERROR_INVALIDDATA;
415 }
416 s->pitch_nbits = av_ceil_log2(pitch_range);
417 s->last_pitch_val = 40;
418 s->last_acb_type = ACB_TYPE_NONE;
419 s->history_nsamples = s->max_pitch_val + 8;
420
421 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
422 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
423 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
424
425 av_log(ctx, AV_LOG_ERROR,
426 "Unsupported samplerate %d (min=%d, max=%d)\n",
427 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
428
429 return AVERROR(ENOSYS);
430 }
431
432 s->block_conv_table[0] = s->min_pitch_val;
433 s->block_conv_table[1] = (pitch_range * 25) >> 6;
434 s->block_conv_table[2] = (pitch_range * 44) >> 6;
435 s->block_conv_table[3] = s->max_pitch_val - 1;
436 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
437 if (s->block_delta_pitch_hrange <= 0) {
438 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
439 return AVERROR_INVALIDDATA;
440 }
441 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
442 s->block_pitch_range = s->block_conv_table[2] +
443 s->block_conv_table[3] + 1 +
444 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
445 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
446
447 ctx->channels = 1;
448 ctx->channel_layout = AV_CH_LAYOUT_MONO;
449 ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
450
451 return 0;
452 }
453
454 /**
455 * @name Postfilter functions
456 * Postfilter functions (gain control, wiener denoise filter, DC filter,
457 * kalman smoothening, plus surrounding code to wrap it)
458 * @{
459 */
460 /**
461 * Adaptive gain control (as used in postfilter).
462 *
463 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
464 * that the energy here is calculated using sum(abs(...)), whereas the
465 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
466 *
467 * @param out output buffer for filtered samples
468 * @param in input buffer containing the samples as they are after the
469 * postfilter steps so far
470 * @param speech_synth input buffer containing speech synth before postfilter
471 * @param size input buffer size
472 * @param alpha exponential filter factor
473 * @param gain_mem pointer to filter memory (single float)
474 */
475 static void adaptive_gain_control(float *out, const float *in,
476 const float *speech_synth,
477 int size, float alpha, float *gain_mem)
478 {
479 int i;
480 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
481 float mem = *gain_mem;
482
483 for (i = 0; i < size; i++) {
484 speech_energy += fabsf(speech_synth[i]);
485 postfilter_energy += fabsf(in[i]);
486 }
487 gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
488
489 for (i = 0; i < size; i++) {
490 mem = alpha * mem + gain_scale_factor;
491 out[i] = in[i] * mem;
492 }
493
494 *gain_mem = mem;
495 }
496
497 /**
498 * Kalman smoothing function.
499 *
500 * This function looks back pitch +/- 3 samples back into history to find
501 * the best fitting curve (that one giving the optimal gain of the two
502 * signals, i.e. the highest dot product between the two), and then
503 * uses that signal history to smoothen the output of the speech synthesis
504 * filter.
505 *
506 * @param s WMA Voice decoding context
507 * @param pitch pitch of the speech signal
508 * @param in input speech signal
509 * @param out output pointer for smoothened signal
510 * @param size input/output buffer size
511 *
512 * @returns -1 if no smoothening took place, e.g. because no optimal
513 * fit could be found, or 0 on success.
514 */
515 static int kalman_smoothen(WMAVoiceContext *s, int pitch,
516 const float *in, float *out, int size)
517 {
518 int n;
519 float optimal_gain = 0, dot;
520 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
521 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
522 *best_hist_ptr;
523
524 /* find best fitting point in history */
525 do {
526 dot = avpriv_scalarproduct_float_c(in, ptr, size);
527 if (dot > optimal_gain) {
528 optimal_gain = dot;
529 best_hist_ptr = ptr;
530 }
531 } while (--ptr >= end);
532
533 if (optimal_gain <= 0)
534 return -1;
535 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
536 if (dot <= 0) // would be 1.0
537 return -1;
538
539 if (optimal_gain <= dot) {
540 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
541 } else
542 dot = 0.625;
543
544 /* actual smoothing */
545 for (n = 0; n < size; n++)
546 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
547
548 return 0;
549 }
550
551 /**
552 * Get the tilt factor of a formant filter from its transfer function
553 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
554 * but somehow (??) it does a speech synthesis filter in the
555 * middle, which is missing here
556 *
557 * @param lpcs LPC coefficients
558 * @param n_lpcs Size of LPC buffer
559 * @returns the tilt factor
560 */
561 static float tilt_factor(const float *lpcs, int n_lpcs)
562 {
563 float rh0, rh1;
564
565 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
566 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
567
568 return rh1 / rh0;
569 }
570
571 /**
572 * Derive denoise filter coefficients (in real domain) from the LPCs.
573 */
574 static void calc_input_response(WMAVoiceContext *s, float *lpcs,
575 int fcb_type, float *coeffs, int remainder)
576 {
577 float last_coeff, min = 15.0, max = -15.0;
578 float irange, angle_mul, gain_mul, range, sq;
579 int n, idx;
580
581 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
582 s->rdft.rdft_calc(&s->rdft, lpcs);
583 #define log_range(var, assign) do { \
584 float tmp = log10f(assign); var = tmp; \
585 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
586 } while (0)
587 log_range(last_coeff, lpcs[1] * lpcs[1]);
588 for (n = 1; n < 64; n++)
589 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
590 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
591 log_range(lpcs[0], lpcs[0] * lpcs[0]);
592 #undef log_range
593 range = max - min;
594 lpcs[64] = last_coeff;
595
596 /* Now, use this spectrum to pick out these frequencies with higher
597 * (relative) power/energy (which we then take to be "not noise"),
598 * and set up a table (still in lpc[]) of (relative) gains per frequency.
599 * These frequencies will be maintained, while others ("noise") will be
600 * decreased in the filter output. */
601 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
602 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
603 (5.0 / 14.7));
604 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
605 for (n = 0; n <= 64; n++) {
606 float pwr;
607
608 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
609 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
610 lpcs[n] = angle_mul * pwr;
611
612 /* 70.57 =~ 1/log10(1.0331663) */
613 idx = (pwr * gain_mul - 0.0295) * 70.570526123;
614 if (idx > 127) { // fall back if index falls outside table range
615 coeffs[n] = wmavoice_energy_table[127] *
616 powf(1.0331663, idx - 127);
617 } else
618 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
619 }
620
621 /* calculate the Hilbert transform of the gains, which we do (since this
622 * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
623 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
624 * "moment" of the LPCs in this filter. */
625 s->dct.dct_calc(&s->dct, lpcs);
626 s->dst.dct_calc(&s->dst, lpcs);
627
628 /* Split out the coefficient indexes into phase/magnitude pairs */
629 idx = 255 + av_clip(lpcs[64], -255, 255);
630 coeffs[0] = coeffs[0] * s->cos[idx];
631 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
632 last_coeff = coeffs[64] * s->cos[idx];
633 for (n = 63;; n--) {
634 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
635 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
636 coeffs[n * 2] = coeffs[n] * s->cos[idx];
637
638 if (!--n) break;
639
640 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
641 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
642 coeffs[n * 2] = coeffs[n] * s->cos[idx];
643 }
644 coeffs[1] = last_coeff;
645
646 /* move into real domain */
647 s->irdft.rdft_calc(&s->irdft, coeffs);
648
649 /* tilt correction and normalize scale */
650 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
651 if (s->denoise_tilt_corr) {
652 float tilt_mem = 0;
653
654 coeffs[remainder - 1] = 0;
655 ff_tilt_compensation(&tilt_mem,
656 -1.8 * tilt_factor(coeffs, remainder - 1),
657 coeffs, remainder);
658 }
659 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
660 remainder));
661 for (n = 0; n < remainder; n++)
662 coeffs[n] *= sq;
663 }
664
665 /**
666 * This function applies a Wiener filter on the (noisy) speech signal as
667 * a means to denoise it.
668 *
669 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
670 * - using this power spectrum, calculate (for each frequency) the Wiener
671 * filter gain, which depends on the frequency power and desired level
672 * of noise subtraction (when set too high, this leads to artifacts)
673 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
674 * of 4-8kHz);
675 * - by doing a phase shift, calculate the Hilbert transform of this array
676 * of per-frequency filter-gains to get the filtering coefficients;
677 * - smoothen/normalize/de-tilt these filter coefficients as desired;
678 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
679 * to get the denoised speech signal;
680 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
681 * the frame boundary) are saved and applied to subsequent frames by an
682 * overlap-add method (otherwise you get clicking-artifacts).
683 *
684 * @param s WMA Voice decoding context
685 * @param fcb_type Frame (codebook) type
686 * @param synth_pf input: the noisy speech signal, output: denoised speech
687 * data; should be 16-byte aligned (for ASM purposes)
688 * @param size size of the speech data
689 * @param lpcs LPCs used to synthesize this frame's speech data
690 */
691 static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
692 float *synth_pf, int size,
693 const float *lpcs)
694 {
695 int remainder, lim, n;
696
697 if (fcb_type != FCB_TYPE_SILENCE) {
698 float *tilted_lpcs = s->tilted_lpcs_pf,
699 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
700
701 tilted_lpcs[0] = 1.0;
702 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
703 memset(&tilted_lpcs[s->lsps + 1], 0,
704 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
705 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
706 tilted_lpcs, s->lsps + 2);
707
708 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
709 * size is applied to the next frame. All input beyond this is zero,
710 * and thus all output beyond this will go towards zero, hence we can
711 * limit to min(size-1, 127-size) as a performance consideration. */
712 remainder = FFMIN(127 - size, size - 1);
713 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
714
715 /* apply coefficients (in frequency spectrum domain), i.e. complex
716 * number multiplication */
717 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
718 s->rdft.rdft_calc(&s->rdft, synth_pf);
719 s->rdft.rdft_calc(&s->rdft, coeffs);
720 synth_pf[0] *= coeffs[0];
721 synth_pf[1] *= coeffs[1];
722 for (n = 1; n < 64; n++) {
723 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
724 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
725 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
726 }
727 s->irdft.rdft_calc(&s->irdft, synth_pf);
728 }
729
730 /* merge filter output with the history of previous runs */
731 if (s->denoise_filter_cache_size) {
732 lim = FFMIN(s->denoise_filter_cache_size, size);
733 for (n = 0; n < lim; n++)
734 synth_pf[n] += s->denoise_filter_cache[n];
735 s->denoise_filter_cache_size -= lim;
736 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
737 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
738 }
739
740 /* move remainder of filter output into a cache for future runs */
741 if (fcb_type != FCB_TYPE_SILENCE) {
742 lim = FFMIN(remainder, s->denoise_filter_cache_size);
743 for (n = 0; n < lim; n++)
744 s->denoise_filter_cache[n] += synth_pf[size + n];
745 if (lim < remainder) {
746 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
747 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
748 s->denoise_filter_cache_size = remainder;
749 }
750 }
751 }
752
753 /**
754 * Averaging projection filter, the postfilter used in WMAVoice.
755 *
756 * This uses the following steps:
757 * - A zero-synthesis filter (generate excitation from synth signal)
758 * - Kalman smoothing on excitation, based on pitch
759 * - Re-synthesized smoothened output
760 * - Iterative Wiener denoise filter
761 * - Adaptive gain filter
762 * - DC filter
763 *
764 * @param s WMAVoice decoding context
765 * @param synth Speech synthesis output (before postfilter)
766 * @param samples Output buffer for filtered samples
767 * @param size Buffer size of synth & samples
768 * @param lpcs Generated LPCs used for speech synthesis
769 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
770 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
771 * @param pitch Pitch of the input signal
772 */
773 static void postfilter(WMAVoiceContext *s, const float *synth,
774 float *samples, int size,
775 const float *lpcs, float *zero_exc_pf,
776 int fcb_type, int pitch)
777 {
778 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
779 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
780 *synth_filter_in = zero_exc_pf;
781
782 assert(size <= MAX_FRAMESIZE / 2);
783
784 /* generate excitation from input signal */
785 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
786
787 if (fcb_type >= FCB_TYPE_AW_PULSES &&
788 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
789 synth_filter_in = synth_filter_in_buf;
790
791 /* re-synthesize speech after smoothening, and keep history */
792 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
793 synth_filter_in, size, s->lsps);
794 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
795 sizeof(synth_pf[0]) * s->lsps);
796
797 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
798
799 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
800 &s->postfilter_agc);
801
802 if (s->dc_level > 8) {
803 /* remove ultra-low frequency DC noise / highpass filter;
804 * coefficients are identical to those used in SIPR decoding,
805 * and very closely resemble those used in AMR-NB decoding. */
806 ff_acelp_apply_order_2_transfer_function(samples, samples,
807 (const float[2]) { -1.99997, 1.0 },
808 (const float[2]) { -1.9330735188, 0.93589198496 },
809 0.93980580475, s->dcf_mem, size);
810 }
811 }
812 /**
813 * @}
814 */
815
816 /**
817 * Dequantize LSPs
818 * @param lsps output pointer to the array that will hold the LSPs
819 * @param num number of LSPs to be dequantized
820 * @param values quantized values, contains n_stages values
821 * @param sizes range (i.e. max value) of each quantized value
822 * @param n_stages number of dequantization runs
823 * @param table dequantization table to be used
824 * @param mul_q LSF multiplier
825 * @param base_q base (lowest) LSF values
826 */
827 static void dequant_lsps(double *lsps, int num,
828 const uint16_t *values,
829 const uint16_t *sizes,
830 int n_stages, const uint8_t *table,
831 const double *mul_q,
832 const double *base_q)
833 {
834 int n, m;
835
836 memset(lsps, 0, num * sizeof(*lsps));
837 for (n = 0; n < n_stages; n++) {
838 const uint8_t *t_off = &table[values[n] * num];
839 double base = base_q[n], mul = mul_q[n];
840
841 for (m = 0; m < num; m++)
842 lsps[m] += base + mul * t_off[m];
843
844 table += sizes[n] * num;
845 }
846 }
847
848 /**
849 * @name LSP dequantization routines
850 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
851 * @note we assume enough bits are available, caller should check.
852 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
853 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
854 * @{
855 */
856 /**
857 * Parse 10 independently-coded LSPs.
858 */
859 static void dequant_lsp10i(BitstreamContext *bc, double *lsps)
860 {
861 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
862 static const double mul_lsf[4] = {
863 5.2187144800e-3, 1.4626986422e-3,
864 9.6179549166e-4, 1.1325736225e-3
865 };
866 static const double base_lsf[4] = {
867 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
868 M_PI * -3.3486e-2, M_PI * -5.7408e-2
869 };
870 uint16_t v[4];
871
872 v[0] = bitstream_read(bc, 8);
873 v[1] = bitstream_read(bc, 6);
874 v[2] = bitstream_read(bc, 5);
875 v[3] = bitstream_read(bc, 5);
876
877 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
878 mul_lsf, base_lsf);
879 }
880
881 /**
882 * Parse 10 independently-coded LSPs, and then derive the tables to
883 * generate LSPs for the other frames from them (residual coding).
884 */
885 static void dequant_lsp10r(BitstreamContext *bc,
886 double *i_lsps, const double *old,
887 double *a1, double *a2, int q_mode)
888 {
889 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
890 static const double mul_lsf[3] = {
891 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
892 };
893 static const double base_lsf[3] = {
894 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
895 };
896 const float (*ipol_tab)[2][10] = q_mode ?
897 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
898 uint16_t interpol, v[3];
899 int n;
900
901 dequant_lsp10i(bc, i_lsps);
902
903 interpol = bitstream_read(bc, 5);
904 v[0] = bitstream_read(bc, 7);
905 v[1] = bitstream_read(bc, 6);
906 v[2] = bitstream_read(bc, 6);
907
908 for (n = 0; n < 10; n++) {
909 double delta = old[n] - i_lsps[n];
910 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
911 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
912 }
913
914 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
915 mul_lsf, base_lsf);
916 }
917
918 /**
919 * Parse 16 independently-coded LSPs.
920 */
921 static void dequant_lsp16i(BitstreamContext *bc, double *lsps)
922 {
923 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
924 static const double mul_lsf[5] = {
925 3.3439586280e-3, 6.9908173703e-4,
926 3.3216608306e-3, 1.0334960326e-3,
927 3.1899104283e-3
928 };
929 static const double base_lsf[5] = {
930 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
931 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
932 M_PI * -1.29816e-1
933 };
934 uint16_t v[5];
935
936 v[0] = bitstream_read(bc, 8);
937 v[1] = bitstream_read(bc, 6);
938 v[2] = bitstream_read(bc, 7);
939 v[3] = bitstream_read(bc, 6);
940 v[4] = bitstream_read(bc, 7);
941
942 dequant_lsps( lsps, 5, v, vec_sizes, 2,
943 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
944 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
945 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
946 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
947 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
948 }
949
950 /**
951 * Parse 16 independently-coded LSPs, and then derive the tables to
952 * generate LSPs for the other frames from them (residual coding).
953 */
954 static void dequant_lsp16r(BitstreamContext *bc,
955 double *i_lsps, const double *old,
956 double *a1, double *a2, int q_mode)
957 {
958 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
959 static const double mul_lsf[3] = {
960 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
961 };
962 static const double base_lsf[3] = {
963 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
964 };
965 const float (*ipol_tab)[2][16] = q_mode ?
966 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
967 uint16_t interpol, v[3];
968 int n;
969
970 dequant_lsp16i(bc, i_lsps);
971
972 interpol = bitstream_read(bc, 5);
973 v[0] = bitstream_read(bc, 7);
974 v[1] = bitstream_read(bc, 7);
975 v[2] = bitstream_read(bc, 7);
976
977 for (n = 0; n < 16; n++) {
978 double delta = old[n] - i_lsps[n];
979 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
980 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
981 }
982
983 dequant_lsps( a2, 10, v, vec_sizes, 1,
984 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
985 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
986 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
987 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
988 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
989 }
990
991 /**
992 * @}
993 * @name Pitch-adaptive window coding functions
994 * The next few functions are for pitch-adaptive window coding.
995 * @{
996 */
997 /**
998 * Parse the offset of the first pitch-adaptive window pulses, and
999 * the distribution of pulses between the two blocks in this frame.
1000 * @param s WMA Voice decoding context private data
1001 * @param bc bit I/O context
1002 * @param pitch pitch for each block in this frame
1003 */
1004 static void aw_parse_coords(WMAVoiceContext *s, BitstreamContext *bc,
1005 const int *pitch)
1006 {
1007 static const int16_t start_offset[94] = {
1008 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1009 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1010 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1011 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1012 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1013 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1014 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1015 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1016 };
1017 int bits, offset;
1018
1019 /* position of pulse */
1020 s->aw_idx_is_ext = 0;
1021 if ((bits = bitstream_read(bc, 6)) >= 54) {
1022 s->aw_idx_is_ext = 1;
1023 bits += (bits - 54) * 3 + bitstream_read(bc, 2);
1024 }
1025
1026 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1027 * the distribution of the pulses in each block contained in this frame. */
1028 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1029 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1030 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1031 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1032 offset += s->aw_n_pulses[0] * pitch[0];
1033 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1034 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1035
1036 /* if continuing from a position before the block, reset position to
1037 * start of block (when corrected for the range over which it can be
1038 * spread in aw_pulse_set1()). */
1039 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1040 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1041 s->aw_first_pulse_off[1] -= pitch[1];
1042 if (start_offset[bits] < 0)
1043 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1044 s->aw_first_pulse_off[0] -= pitch[0];
1045 }
1046 }
1047
1048 /**
1049 * Apply second set of pitch-adaptive window pulses.
1050 * @param s WMA Voice decoding context private data
1051 * @param bc bit I/O context
1052 * @param block_idx block index in frame [0, 1]
1053 * @param fcb structure containing fixed codebook vector info
1054 * @return -1 on error, 0 otherwise
1055 */
1056 static int aw_pulse_set2(WMAVoiceContext *s, BitstreamContext *bc,
1057 int block_idx, AMRFixed *fcb)
1058 {
1059 uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
1060 uint16_t *use_mask = use_mask_mem + 2;
1061 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1062 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1063 * of idx are the position of the bit within a particular item in the
1064 * array (0 being the most significant bit, and 15 being the least
1065 * significant bit), and the remainder (>> 4) is the index in the
1066 * use_mask[]-array. This is faster and uses less memory than using a
1067 * 80-byte/80-int array. */
1068 int pulse_off = s->aw_first_pulse_off[block_idx],
1069 pulse_start, n, idx, range, aidx, start_off = 0;
1070
1071 /* set offset of first pulse to within this block */
1072 if (s->aw_n_pulses[block_idx] > 0)
1073 while (pulse_off + s->aw_pulse_range < 1)
1074 pulse_off += fcb->pitch_lag;
1075
1076 /* find range per pulse */
1077 if (s->aw_n_pulses[0] > 0) {
1078 if (block_idx == 0) {
1079 range = 32;
1080 } else /* block_idx = 1 */ {
1081 range = 8;
1082 if (s->aw_n_pulses[block_idx] > 0)
1083 pulse_off = s->aw_next_pulse_off_cache;
1084 }
1085 } else
1086 range = 16;
1087 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1088
1089 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1090 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1091 * we exclude that range from being pulsed again in this function. */
1092 memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
1093 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
1094 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1095 if (s->aw_n_pulses[block_idx] > 0)
1096 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1097 int excl_range = s->aw_pulse_range; // always 16 or 24
1098 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1099 int first_sh = 16 - (idx & 15);
1100 *use_mask_ptr++ &= 0xFFFFu << first_sh;
1101 excl_range -= first_sh;
1102 if (excl_range >= 16) {
1103 *use_mask_ptr++ = 0;
1104 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1105 } else
1106 *use_mask_ptr &= 0xFFFF >> excl_range;
1107 }
1108
1109 /* find the 'aidx'th offset that is not excluded */
1110 aidx = bitstream_read(bc, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1111 for (n = 0; n <= aidx; pulse_start++) {
1112 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1113 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1114 if (use_mask[0]) idx = 0x0F;
1115 else if (use_mask[1]) idx = 0x1F;
1116 else if (use_mask[2]) idx = 0x2F;
1117 else if (use_mask[3]) idx = 0x3F;
1118 else if (use_mask[4]) idx = 0x4F;
1119 else return -1;
1120 idx -= av_log2_16bit(use_mask[idx >> 4]);
1121 }
1122 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1123 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1124 n++;
1125 start_off = idx;
1126 }
1127 }
1128
1129 fcb->x[fcb->n] = start_off;
1130 fcb->y[fcb->n] = bitstream_read_bit(bc) ? -1.0 : 1.0;
1131 fcb->n++;
1132
1133 /* set offset for next block, relative to start of that block */
1134 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1135 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1136 return 0;
1137 }
1138
1139 /**
1140 * Apply first set of pitch-adaptive window pulses.
1141 * @param s WMA Voice decoding context private data
1142 * @param bc bit I/O context
1143 * @param block_idx block index in frame [0, 1]
1144 * @param fcb storage location for fixed codebook pulse info
1145 */
1146 static void aw_pulse_set1(WMAVoiceContext *s, BitstreamContext *bc,
1147 int block_idx, AMRFixed *fcb)
1148 {
1149 int val = bitstream_read(bc, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1150 float v;
1151
1152 if (s->aw_n_pulses[block_idx] > 0) {
1153 int n, v_mask, i_mask, sh, n_pulses;
1154
1155 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1156 n_pulses = 3;
1157 v_mask = 8;
1158 i_mask = 7;
1159 sh = 4;
1160 } else { // 4 pulses, 1:sign + 2:index each
1161 n_pulses = 4;
1162 v_mask = 4;
1163 i_mask = 3;
1164 sh = 3;
1165 }
1166
1167 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1168 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1169 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1170 s->aw_first_pulse_off[block_idx];
1171 while (fcb->x[fcb->n] < 0)
1172 fcb->x[fcb->n] += fcb->pitch_lag;
1173 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1174 fcb->n++;
1175 }
1176 } else {
1177 int num2 = (val & 0x1FF) >> 1, delta, idx;
1178
1179 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1180 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1181 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1182 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1183 v = (val & 0x200) ? -1.0 : 1.0;
1184
1185 fcb->no_repeat_mask |= 3 << fcb->n;
1186 fcb->x[fcb->n] = idx - delta;
1187 fcb->y[fcb->n] = v;
1188 fcb->x[fcb->n + 1] = idx;
1189 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1190 fcb->n += 2;
1191 }
1192 }
1193
1194 /**
1195 * @}
1196 *
1197 * Generate a random number from frame_cntr and block_idx, which will live
1198 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1199 * table of size 1000 of which you want to read block_size entries).
1200 *
1201 * @param frame_cntr current frame number
1202 * @param block_num current block index
1203 * @param block_size amount of entries we want to read from a table
1204 * that has 1000 entries
1205 * @return a (non-)random number in the [0, 1000 - block_size] range.
1206 */
1207 static int pRNG(int frame_cntr, int block_num, int block_size)
1208 {
1209 /* array to simplify the calculation of z:
1210 * y = (x % 9) * 5 + 6;
1211 * z = (49995 * x) / y;
1212 * Since y only has 9 values, we can remove the division by using a
1213 * LUT and using FASTDIV-style divisions. For each of the 9 values
1214 * of y, we can rewrite z as:
1215 * z = x * (49995 / y) + x * ((49995 % y) / y)
1216 * In this table, each col represents one possible value of y, the
1217 * first number is 49995 / y, and the second is the FASTDIV variant
1218 * of 49995 % y / y. */
1219 static const unsigned int div_tbl[9][2] = {
1220 { 8332, 3 * 715827883U }, // y = 6
1221 { 4545, 0 * 390451573U }, // y = 11
1222 { 3124, 11 * 268435456U }, // y = 16
1223 { 2380, 15 * 204522253U }, // y = 21
1224 { 1922, 23 * 165191050U }, // y = 26
1225 { 1612, 23 * 138547333U }, // y = 31
1226 { 1388, 27 * 119304648U }, // y = 36
1227 { 1219, 16 * 104755300U }, // y = 41
1228 { 1086, 39 * 93368855U } // y = 46
1229 };
1230 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1231 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1232 // so this is effectively a modulo (%)
1233 y = x - 9 * MULH(477218589, x); // x % 9
1234 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1235 // z = x * 49995 / (y * 5 + 6)
1236 return z % (1000 - block_size);
1237 }
1238
1239 /**
1240 * Parse hardcoded signal for a single block.
1241 * @note see #synth_block().
1242 */
1243 static void synth_block_hardcoded(WMAVoiceContext *s, BitstreamContext *bc,
1244 int block_idx, int size,
1245 const struct frame_type_desc *frame_desc,
1246 float *excitation)
1247 {
1248 float gain;
1249 int n, r_idx;
1250
1251 assert(size <= MAX_FRAMESIZE);
1252
1253 /* Set the offset from which we start reading wmavoice_std_codebook */
1254 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1255 r_idx = pRNG(s->frame_cntr, block_idx, size);
1256 gain = s->silence_gain;
1257 } else /* FCB_TYPE_HARDCODED */ {
1258 r_idx = bitstream_read(bc, 8);
1259 gain = wmavoice_gain_universal[bitstream_read(bc, 6)];
1260 }
1261
1262 /* Clear gain prediction parameters */
1263 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1264
1265 /* Apply gain to hardcoded codebook and use that as excitation signal */
1266 for (n = 0; n < size; n++)
1267 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1268 }
1269
1270 /**
1271 * Parse FCB/ACB signal for a single block.
1272 * @note see #synth_block().
1273 */
1274 static void synth_block_fcb_acb(WMAVoiceContext *s, BitstreamContext *bc,
1275 int block_idx, int size,
1276 int block_pitch_sh2,
1277 const struct frame_type_desc *frame_desc,
1278 float *excitation)
1279 {
1280 static const float gain_coeff[6] = {
1281 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1282 };
1283 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1284 int n, idx, gain_weight;
1285 AMRFixed fcb;
1286
1287 assert(size <= MAX_FRAMESIZE / 2);
1288 memset(pulses, 0, sizeof(*pulses) * size);
1289
1290 fcb.pitch_lag = block_pitch_sh2 >> 2;
1291 fcb.pitch_fac = 1.0;
1292 fcb.no_repeat_mask = 0;
1293 fcb.n = 0;
1294
1295 /* For the other frame types, this is where we apply the innovation
1296 * (fixed) codebook pulses of the speech signal. */
1297 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1298 aw_pulse_set1(s, bc, block_idx, &fcb);
1299 if (aw_pulse_set2(s, bc, block_idx, &fcb)) {
1300 /* Conceal the block with silence and return.
1301 * Skip the correct amount of bits to read the next
1302 * block from the correct offset. */
1303 int r_idx = pRNG(s->frame_cntr, block_idx, size);
1304
1305 for (n = 0; n < size; n++)
1306 excitation[n] =
1307 wmavoice_std_codebook[r_idx + n] * s->silence_gain;
1308 bitstream_skip(bc, 7 + 1);
1309 return;
1310 }
1311 } else /* FCB_TYPE_EXC_PULSES */ {
1312 int offset_nbits = 5 - frame_desc->log_n_blocks;
1313
1314 fcb.no_repeat_mask = -1;
1315 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1316 * (instead of double) for a subset of pulses */
1317 for (n = 0; n < 5; n++) {
1318 float sign;
1319 int pos1, pos2;
1320
1321 sign = bitstream_read_bit(bc) ? 1.0 : -1.0;
1322 pos1 = bitstream_read(bc, offset_nbits);
1323 fcb.x[fcb.n] = n + 5 * pos1;
1324 fcb.y[fcb.n++] = sign;
1325 if (n < frame_desc->dbl_pulses) {
1326 pos2 = bitstream_read(bc, offset_nbits);
1327 fcb.x[fcb.n] = n + 5 * pos2;
1328 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1329 }
1330 }
1331 }
1332 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1333
1334 /* Calculate gain for adaptive & fixed codebook signal.
1335 * see ff_amr_set_fixed_gain(). */
1336 idx = bitstream_read(bc, 7);
1337 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
1338 gain_coeff, 6) -
1339 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1340 acb_gain = wmavoice_gain_codebook_acb[idx];
1341 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1342 -2.9957322736 /* log(0.05) */,
1343 1.6094379124 /* log(5.0) */);
1344
1345 gain_weight = 8 >> frame_desc->log_n_blocks;
1346 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1347 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1348 for (n = 0; n < gain_weight; n++)
1349 s->gain_pred_err[n] = pred_err;
1350
1351 /* Calculation of adaptive codebook */
1352 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1353 int len;
1354 for (n = 0; n < size; n += len) {
1355 int next_idx_sh16;
1356 int abs_idx = block_idx * size + n;
1357 int pitch_sh16 = (s->last_pitch_val << 16) +
1358 s->pitch_diff_sh16 * abs_idx;
1359 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1360 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1361 idx = idx_sh16 >> 16;
1362 if (s->pitch_diff_sh16) {
1363 if (s->pitch_diff_sh16 > 0) {
1364 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1365 } else
1366 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1367 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1368 1, size - n);
1369 } else
1370 len = size;
1371
1372 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1373 wmavoice_ipol1_coeffs, 17,
1374 idx, 9, len);
1375 }
1376 } else /* ACB_TYPE_HAMMING */ {
1377 int block_pitch = block_pitch_sh2 >> 2;
1378 idx = block_pitch_sh2 & 3;
1379 if (idx) {
1380 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1381 wmavoice_ipol2_coeffs, 4,
1382 idx, 8, size);
1383 } else
1384 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1385 sizeof(float) * size);
1386 }
1387
1388 /* Interpolate ACB/FCB and use as excitation signal */
1389 ff_weighted_vector_sumf(excitation, excitation, pulses,
1390 acb_gain, fcb_gain, size);
1391 }
1392
1393 /**
1394 * Parse data in a single block.
1395 * @note we assume enough bits are available, caller should check.
1396 *
1397 * @param s WMA Voice decoding context private data
1398 * @param bc bit I/O context
1399 * @param block_idx index of the to-be-read block
1400 * @param size amount of samples to be read in this block
1401 * @param block_pitch_sh2 pitch for this block << 2
1402 * @param lsps LSPs for (the end of) this frame
1403 * @param prev_lsps LSPs for the last frame
1404 * @param frame_desc frame type descriptor
1405 * @param excitation target memory for the ACB+FCB interpolated signal
1406 * @param synth target memory for the speech synthesis filter output
1407 * @return 0 on success, <0 on error.
1408 */
1409 static void synth_block(WMAVoiceContext *s, BitstreamContext *bc,
1410 int block_idx, int size,
1411 int block_pitch_sh2,
1412 const double *lsps, const double *prev_lsps,
1413 const struct frame_type_desc *frame_desc,
1414 float *excitation, float *synth)
1415 {
1416 double i_lsps[MAX_LSPS];
1417 float lpcs[MAX_LSPS];
1418 float fac;
1419 int n;
1420
1421 if (frame_desc->acb_type == ACB_TYPE_NONE)
1422 synth_block_hardcoded(s, bc, block_idx, size, frame_desc, excitation);
1423 else
1424 synth_block_fcb_acb(s, bc, block_idx, size, block_pitch_sh2,
1425 frame_desc, excitation);
1426
1427 /* convert interpolated LSPs to LPCs */
1428 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1429 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1430 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1431 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1432
1433 /* Speech synthesis */
1434 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1435 }
1436
1437 /**
1438 * Synthesize output samples for a single frame.
1439 * @note we assume enough bits are available, caller should check.
1440 *
1441 * @param ctx WMA Voice decoder context
1442 * @param bc bit I/O context (s->bc or one for cross-packet superframes)
1443 * @param frame_idx Frame number within superframe [0-2]
1444 * @param samples pointer to output sample buffer, has space for at least 160
1445 * samples
1446 * @param lsps LSP array
1447 * @param prev_lsps array of previous frame's LSPs
1448 * @param excitation target buffer for excitation signal
1449 * @param synth target buffer for synthesized speech data
1450 * @return 0 on success, <0 on error.
1451 */
1452 static int synth_frame(AVCodecContext *ctx, BitstreamContext *bc,
1453 int frame_idx, float *samples,
1454 const double *lsps, const double *prev_lsps,
1455 float *excitation, float *synth)
1456 {
1457 WMAVoiceContext *s = ctx->priv_data;
1458 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
1459 int pitch[MAX_BLOCKS], last_block_pitch;
1460
1461 /* Parse frame type ("frame header"), see frame_descs */
1462 int bd_idx = s->vbm_tree[bitstream_read_vlc(bc, frame_type_vlc.table, 6, 3)], block_nsamples;
1463
1464 if (bd_idx < 0) {
1465 av_log(ctx, AV_LOG_ERROR,
1466 "Invalid frame type VLC code, skipping\n");
1467 return AVERROR_INVALIDDATA;
1468 }
1469
1470 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1471
1472 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1473 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1474 /* Pitch is provided per frame, which is interpreted as the pitch of
1475 * the last sample of the last block of this frame. We can interpolate
1476 * the pitch of other blocks (and even pitch-per-sample) by gradually
1477 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1478 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1479 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1480 cur_pitch_val = s->min_pitch_val + bitstream_read(bc, s->pitch_nbits);
1481 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1482 if (s->last_acb_type == ACB_TYPE_NONE ||
1483 20 * abs(cur_pitch_val - s->last_pitch_val) >
1484 (cur_pitch_val + s->last_pitch_val))
1485 s->last_pitch_val = cur_pitch_val;
1486
1487 /* pitch per block */
1488 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1489 int fac = n * 2 + 1;
1490
1491 pitch[n] = (MUL16(fac, cur_pitch_val) +
1492 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1493 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1494 }
1495
1496 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1497 s->pitch_diff_sh16 =
1498 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1499 }
1500
1501 /* Global gain (if silence) and pitch-adaptive window coordinates */
1502 switch (frame_descs[bd_idx].fcb_type) {
1503 case FCB_TYPE_SILENCE:
1504 s->silence_gain = wmavoice_gain_silence[bitstream_read(bc, 8)];
1505 break;
1506 case FCB_TYPE_AW_PULSES:
1507 aw_parse_coords(s, bc, pitch);
1508 break;
1509 }
1510
1511 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1512 int bl_pitch_sh2;
1513
1514 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1515 switch (frame_descs[bd_idx].acb_type) {
1516 case ACB_TYPE_HAMMING: {
1517 /* Pitch is given per block. Per-block pitches are encoded as an
1518 * absolute value for the first block, and then delta values
1519 * relative to this value) for all subsequent blocks. The scale of
1520 * this pitch value is semi-logarithmic compared to its use in the
1521 * decoder, so we convert it to normal scale also. */
1522 int block_pitch,
1523 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1524 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1525 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1526
1527 if (n == 0) {
1528 block_pitch = bitstream_read(bc, s->block_pitch_nbits);
1529 } else
1530 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1531 bitstream_read(bc, s->block_delta_pitch_nbits);
1532 /* Convert last_ so that any next delta is within _range */
1533 last_block_pitch = av_clip(block_pitch,
1534 s->block_delta_pitch_hrange,
1535 s->block_pitch_range -
1536 s->block_delta_pitch_hrange);
1537
1538 /* Convert semi-log-style scale back to normal scale */
1539 if (block_pitch < t1) {
1540 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1541 } else {
1542 block_pitch -= t1;
1543 if (block_pitch < t2) {
1544 bl_pitch_sh2 =
1545 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1546 } else {
1547 block_pitch -= t2;
1548 if (block_pitch < t3) {
1549 bl_pitch_sh2 =
1550 (s->block_conv_table[2] + block_pitch) << 2;
1551 } else
1552 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1553 }
1554 }
1555 pitch[n] = bl_pitch_sh2 >> 2;
1556 break;
1557 }
1558
1559 case ACB_TYPE_ASYMMETRIC: {
1560 bl_pitch_sh2 = pitch[n] << 2;
1561 break;
1562 }
1563
1564 default: // ACB_TYPE_NONE has no pitch
1565 bl_pitch_sh2 = 0;
1566 break;
1567 }
1568
1569 synth_block(s, bc, n, block_nsamples, bl_pitch_sh2,
1570 lsps, prev_lsps, &frame_descs[bd_idx],
1571 &excitation[n * block_nsamples],
1572 &synth[n * block_nsamples]);
1573 }
1574
1575 /* Averaging projection filter, if applicable. Else, just copy samples
1576 * from synthesis buffer */
1577 if (s->do_apf) {
1578 double i_lsps[MAX_LSPS];
1579 float lpcs[MAX_LSPS];
1580
1581 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1582 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1583 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1584 postfilter(s, synth, samples, 80, lpcs,
1585 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1586 frame_descs[bd_idx].fcb_type, pitch[0]);
1587
1588 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1589 i_lsps[n] = cos(lsps[n]);
1590 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1591 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1592 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1593 frame_descs[bd_idx].fcb_type, pitch[0]);
1594 } else
1595 memcpy(samples, synth, 160 * sizeof(synth[0]));
1596
1597 /* Cache values for next frame */
1598 s->frame_cntr++;
1599 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1600 s->last_acb_type = frame_descs[bd_idx].acb_type;
1601 switch (frame_descs[bd_idx].acb_type) {
1602 case ACB_TYPE_NONE:
1603 s->last_pitch_val = 0;
1604 break;
1605 case ACB_TYPE_ASYMMETRIC:
1606 s->last_pitch_val = cur_pitch_val;
1607 break;
1608 case ACB_TYPE_HAMMING:
1609 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1610 break;
1611 }
1612
1613 return 0;
1614 }
1615
1616 /**
1617 * Ensure minimum value for first item, maximum value for last value,
1618 * proper spacing between each value and proper ordering.
1619 *
1620 * @param lsps array of LSPs
1621 * @param num size of LSP array
1622 *
1623 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1624 * useful to put in a generic location later on. Parts are also
1625 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1626 * which is in float.
1627 */
1628 static void stabilize_lsps(double *lsps, int num)
1629 {
1630 int n, m, l;
1631
1632 /* set minimum value for first, maximum value for last and minimum
1633 * spacing between LSF values.
1634 * Very similar to ff_set_min_dist_lsf(), but in double. */
1635 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1636 for (n = 1; n < num; n++)
1637 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1638 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1639
1640 /* reorder (looks like one-time / non-recursed bubblesort).
1641 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1642 for (n = 1; n < num; n++) {
1643 if (lsps[n] < lsps[n - 1]) {
1644 for (m = 1; m < num; m++) {
1645 double tmp = lsps[m];
1646 for (l = m - 1; l >= 0; l--) {
1647 if (lsps[l] <= tmp) break;
1648 lsps[l + 1] = lsps[l];
1649 }
1650 lsps[l + 1] = tmp;
1651 }
1652 break;
1653 }
1654 }
1655 }
1656
1657 /**
1658 * Test if there's enough bits to read 1 superframe.
1659 *
1660 * @param orig_bc bit I/O context used for reading. This function
1661 * does not modify the state of the bitreader; it
1662 * only uses it to copy the current stream position
1663 * @param s WMA Voice decoding context private data
1664 * @return < 0 on error, 1 on not enough bits or 0 if OK.
1665 */
1666 static int check_bits_for_superframe(BitstreamContext *orig_bc,
1667 WMAVoiceContext *s)
1668 {
1669 BitstreamContext s_bc, *bc = &s_bc;
1670 int n, need_bits, bd_idx;
1671 const struct frame_type_desc *frame_desc;
1672
1673 /* initialize a copy */
1674 *bc = *orig_bc;
1675
1676 /* superframe header */
1677 if (bitstream_bits_left(bc) < 14)
1678 return 1;
1679 if (!bitstream_read_bit(bc))
1680 return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe
1681 if (bitstream_read_bit(bc)) bitstream_skip(bc, 12); // number of samples in superframe
1682 if (s->has_residual_lsps) { // residual LSPs (for all frames)
1683 if (bitstream_bits_left(bc) < s->sframe_lsp_bitsize)
1684 return 1;
1685 bitstream_skip(bc, s->sframe_lsp_bitsize);
1686 }
1687
1688 /* frames */
1689 for (n = 0; n < MAX_FRAMES; n++) {
1690 int aw_idx_is_ext = 0;
1691
1692 if (!s->has_residual_lsps) { // independent LSPs (per-frame)
1693 if (bitstream_bits_left(bc) < s->frame_lsp_bitsize)
1694 return 1;
1695 bitstream_skip(bc, s->frame_lsp_bitsize);
1696 }
1697 bd_idx = s->vbm_tree[bitstream_read_vlc(bc, 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 (bitstream_bits_left(bc) < s->pitch_nbits)
1703 return 1;
1704 bitstream_skip(bc, s->pitch_nbits);
1705 }
1706 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1707 bitstream_skip(bc, 8);
1708 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1709 int tmp = bitstream_read(bc, 6);
1710 if (tmp >= 0x36) {
1711 bitstream_skip(bc, 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 (bitstream_bits_left(bc) < need_bits)
1726 return 1;
1727 bitstream_skip(bc, 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->bc.
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 BitstreamContext *bc = &s->bc, s_bc;
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 bc = &s_bc;
1770 bitstream_init(bc, s->sframe_cache, s->sframe_cache_size);
1771 s->sframe_cache_size = 0;
1772 }
1773
1774 if ((res = check_bits_for_superframe(bc, 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 (!bitstream_read_bit(bc)) {
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 (bitstream_read_bit(bc)) {
1791 if ((n_samples = bitstream_read(bc, 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(bc, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1807 } else /* s->lsps == 16 */
1808 dequant_lsp16r(bc, 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 av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n");
1823 return res;
1824 }
1825 frame->nb_samples = n_samples;
1826 samples = (float *)frame->data[0];
1827
1828 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1829 for (n = 0; n < 3; n++) {
1830 if (!s->has_residual_lsps) {
1831 int m;
1832
1833 if (s->lsps == 10) {
1834 dequant_lsp10i(bc, lsps[n]);
1835 } else /* s->lsps == 16 */
1836 dequant_lsp16i(bc, lsps[n]);
1837
1838 for (m = 0; m < s->lsps; m++)
1839 lsps[n][m] += mean_lsf[m];
1840 stabilize_lsps(lsps[n], s->lsps);
1841 }
1842
1843 if ((res = synth_frame(ctx, bc, n,
1844 &samples[n * MAX_FRAMESIZE],
1845 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1846 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1847 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
1848 *got_frame_ptr = 0;
1849 return res;
1850 }
1851 }
1852
1853 /* Statistics? FIXME - we don't check for length, a slight overrun
1854 * will be caught by internal buffer padding, and anything else
1855 * will be skipped, not read. */
1856 if (bitstream_read_bit(bc)) {
1857 res = bitstream_read(bc, 4);
1858 bitstream_skip(bc, 10 * (res + 1));
1859 }
1860
1861 *got_frame_ptr = 1;
1862
1863 /* Update history */
1864 memcpy(s->prev_lsps, lsps[2],
1865 s->lsps * sizeof(*s->prev_lsps));
1866 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1867 s->lsps * sizeof(*synth));
1868 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1869 s->history_nsamples * sizeof(*excitation));
1870 if (s->do_apf)
1871 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1872 s->history_nsamples * sizeof(*s->zero_exc_pf));
1873
1874 return 0;
1875 }
1876
1877 /**
1878 * Parse the packet header at the start of each packet (input data to this
1879 * decoder).
1880 *
1881 * @param s WMA Voice decoding context private data
1882 * @return 1 if not enough bits were available, or 0 on success.
1883 */
1884 static int parse_packet_header(WMAVoiceContext *s)
1885 {
1886 BitstreamContext *bc = &s->bc;
1887 unsigned int res;
1888
1889 if (bitstream_bits_left(bc) < 11)
1890 return 1;
1891 bitstream_skip(bc, 4); // packet sequence number
1892 s->has_residual_lsps = bitstream_read_bit(bc);
1893 do {
1894 res = bitstream_read(bc, 6); // number of superframes per packet
1895 // (minus first one if there is spillover)
1896 if (bitstream_bits_left(bc) < 6 * (res == 0x3F) + s->spillover_bitsize)
1897 return 1;
1898 } while (res == 0x3F);
1899 s->spillover_nbits = bitstream_read(bc, s->spillover_bitsize);
1900
1901 return 0;
1902 }
1903
1904 /**
1905 * Copy (unaligned) bits from bc/data/size to pb.
1906 *
1907 * @param pb target buffer to copy bits into
1908 * @param data source buffer to copy bits from
1909 * @param size size of the source data, in bytes
1910 * @param bc bit I/O context specifying the current position in the source.
1911 * data. This function might use this to align the bit position to
1912 * a whole-byte boundary before calling #avpriv_copy_bits() on aligned
1913 * source data
1914 * @param nbits the amount of bits to copy from source to target
1915 *
1916 * @note after calling this function, the current position in the input bit
1917 * I/O context is undefined.
1918 */
1919 static void copy_bits(PutBitContext *pb,
1920 const uint8_t *data, int size,
1921 BitstreamContext *bc, int nbits)
1922 {
1923 int rmn_bytes, rmn_bits;
1924
1925 rmn_bits = rmn_bytes = bitstream_bits_left(bc);
1926 if (rmn_bits < nbits)
1927 return;
1928 if (nbits > pb->size_in_bits - put_bits_count(pb))
1929 return;
1930 rmn_bits &= 7; rmn_bytes >>= 3;
1931 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1932 put_bits(pb, rmn_bits, bitstream_read(bc, rmn_bits));
1933 avpriv_copy_bits(pb, data + size - rmn_bytes,
1934 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1935 }
1936
1937 /**
1938 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1939 * and we expect that the demuxer / application provides it to us as such
1940 * (else you'll probably get garbage as output). Every packet has a size of
1941 * ctx->block_align bytes, starts with a packet header (see
1942 * #parse_packet_header()), and then a series of superframes. Superframe
1943 * boundaries may exceed packets, i.e. superframes can split data over
1944 * multiple (two) packets.
1945 *
1946 * For more information about frames, see #synth_superframe().
1947 */
1948 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1949 int *got_frame_ptr, AVPacket *avpkt)
1950 {
1951 WMAVoiceContext *s = ctx->priv_data;
1952 BitstreamContext *bc = &s->bc;
1953 int size, res, pos;
1954
1955 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1956 * header at each ctx->block_align bytes. However, Libav's ASF demuxer
1957 * feeds us ASF packets, which may concatenate multiple "codec" packets
1958 * in a single "muxer" packet, so we artificially emulate that by
1959 * capping the packet size at ctx->block_align. */
1960 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1961 if (!size) {
1962 *got_frame_ptr = 0;
1963 return 0;
1964 }
1965 bitstream_init8(&s->bc, avpkt->data, size);
1966
1967 /* size == ctx->block_align is used to indicate whether we are dealing with
1968 * a new packet or a packet of which we already read the packet header
1969 * previously. */
1970 if (size == ctx->block_align) { // new packet header
1971 if ((res = parse_packet_header(s)) < 0)
1972 return res;
1973
1974 /* If the packet header specifies a s->spillover_nbits, then we want
1975 * to push out all data of the previous packet (+ spillover) before
1976 * continuing to parse new superframes in the current packet. */
1977 if (s->spillover_nbits > 0) {
1978 if (s->sframe_cache_size > 0) {
1979 int cnt = bitstream_tell(bc);
1980 copy_bits(&s->pb, avpkt->data, size, bc, s->spillover_nbits);
1981 flush_put_bits(&s->pb);
1982 s->sframe_cache_size += s->spillover_nbits;
1983 if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 &&
1984 *got_frame_ptr) {
1985 cnt += s->spillover_nbits;
1986 s->skip_bits_next = cnt & 7;
1987 return cnt >> 3;
1988 } else
1989 bitstream_skip (bc, s->spillover_nbits - cnt +
1990 bitstream_tell(bc)); // resync
1991 } else
1992 bitstream_skip(bc, s->spillover_nbits); // resync
1993 }
1994 } else if (s->skip_bits_next)
1995 bitstream_skip(bc, s->skip_bits_next);
1996
1997 /* Try parsing superframes in current packet */
1998 s->sframe_cache_size = 0;
1999 s->skip_bits_next = 0;
2000 pos = bitstream_bits_left(bc);
2001 if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
2002 return res;
2003 } else if (*got_frame_ptr) {
2004 int cnt = bitstream_tell(bc);
2005 s->skip_bits_next = cnt & 7;
2006 return cnt >> 3;
2007 } else if ((s->sframe_cache_size = pos) > 0) {
2008 /* rewind bit reader to start of last (incomplete) superframe... */
2009 bitstream_init8(bc, avpkt->data, size);
2010 bitstream_skip(bc, (size << 3) - pos);
2011 assert(bitstream_bits_left(bc) == pos);
2012
2013 /* ...and cache it for spillover in next packet */
2014 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
2015 copy_bits(&s->pb, avpkt->data, size, bc, s->sframe_cache_size);
2016 // FIXME bad - just copy bytes as whole and add use the
2017 // skip_bits_next field
2018 }
2019
2020 return size;
2021 }
2022
2023 static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
2024 {
2025 WMAVoiceContext *s = ctx->priv_data;
2026
2027 if (s->do_apf) {
2028 ff_rdft_end(&s->rdft);
2029 ff_rdft_end(&s->irdft);
2030 ff_dct_end(&s->dct);
2031 ff_dct_end(&s->dst);
2032 }
2033
2034 return 0;
2035 }
2036
2037 static av_cold void wmavoice_flush(AVCodecContext *ctx)
2038 {
2039 WMAVoiceContext *s = ctx->priv_data;
2040 int n;
2041
2042 s->postfilter_agc = 0;
2043 s->sframe_cache_size = 0;
2044 s->skip_bits_next = 0;
2045 for (n = 0; n < s->lsps; n++)
2046 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2047 memset(s->excitation_history, 0,
2048 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2049 memset(s->synth_history, 0,
2050 sizeof(*s->synth_history) * MAX_LSPS);
2051 memset(s->gain_pred_err, 0,
2052 sizeof(s->gain_pred_err));
2053
2054 if (s->do_apf) {
2055 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2056 sizeof(*s->synth_filter_out_buf) * s->lsps);
2057 memset(s->dcf_mem, 0,
2058 sizeof(*s->dcf_mem) * 2);
2059 memset(s->zero_exc_pf, 0,
2060 sizeof(*s->zero_exc_pf) * s->history_nsamples);
2061 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2062 }
2063 }
2064
2065 AVCodec ff_wmavoice_decoder = {
2066 .name = "wmavoice",
2067 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
2068 .type = AVMEDIA_TYPE_AUDIO,
2069 .id = AV_CODEC_ID_WMAVOICE,
2070 .priv_data_size = sizeof(WMAVoiceContext),
2071 .init = wmavoice_decode_init,
2072 .init_static_data = wmavoice_init_static_data,
2073 .close = wmavoice_decode_end,
2074 .decode = wmavoice_decode_packet,
2075 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
2076 .flush = wmavoice_flush,
2077 };