1 /*
2 * Windows Media Audio Voice decoder.
3 * Copyright (c) 2009 Ronald S. Bultje
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * @brief Windows Media Audio Voice compatible decoder
25 * @author Ronald S. Bultje <rsbultje@gmail.com>
26 */
27
28 #include <math.h>
29
30 #include "libavutil/channel_layout.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/mem_internal.h"
33 #include "libavutil/thread.h"
34 #include "avcodec.h"
35 #include "codec_internal.h"
36 #include "internal.h"
37 #include "get_bits.h"
38 #include "put_bits.h"
39 #include "wmavoice_data.h"
40 #include "celp_filters.h"
41 #include "acelp_vectors.h"
42 #include "acelp_filters.h"
43 #include "lsp.h"
44 #include "dct.h"
45 #include "rdft.h"
46 #include "sinewin.h"
47
48 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame
49 #define MAX_LSPS 16 ///< maximum filter order
50 #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple
51 ///< of 16 for ASM input buffer alignment
52 #define MAX_FRAMES 3 ///< maximum number of frames per superframe
53 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame
54 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history
55 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
56 ///< maximum number of samples per superframe
57 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
58 ///< was split over two packets
59 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration
60
61 /**
62 * Frame type VLC coding.
63 */
64 static VLC frame_type_vlc;
65
66 /**
67 * Adaptive codebook types.
68 */
69 enum {
70 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed)
71 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
72 ///< we interpolate to get a per-sample pitch.
73 ///< Signal is generated using an asymmetric sinc
74 ///< window function
75 ///< @note see #wmavoice_ipol1_coeffs
76 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using
77 ///< a Hamming sinc window function
78 ///< @note see #wmavoice_ipol2_coeffs
79 };
80
81 /**
82 * Fixed codebook types.
83 */
84 enum {
85 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence
86 ///< generated from a hardcoded (fixed) codebook
87 ///< with per-frame (low) gain values
88 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block
89 ///< gain values
90 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals,
91 ///< used in particular for low-bitrate streams
92 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
93 ///< combinations of either single pulses or
94 ///< pulse pairs
95 };
96
97 /**
98 * Description of frame types.
99 */
100 static const struct frame_type_desc {
101 uint8_t n_blocks; ///< amount of blocks per frame (each block
102 ///< (contains 160/#n_blocks samples)
103 uint8_t log_n_blocks; ///< log2(#n_blocks)
104 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*)
105 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*)
106 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs
107 ///< (rather than just one single pulse)
108 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
109 } frame_descs[17] = {
110 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0 },
111 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0 },
112 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0 },
113 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 },
114 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 },
115 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0 },
116 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 },
117 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 },
118 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 },
119 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 },
120 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 },
121 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 },
122 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 },
123 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 },
124 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 },
125 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 },
126 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }
127 };
128
129 /**
130 * WMA Voice decoding context.
131 */
132 typedef struct WMAVoiceContext {
133 /**
134 * @name Global values specified in the stream header / extradata or used all over.
135 * @{
136 */
137 GetBitContext gb; ///< packet bitreader. During decoder init,
138 ///< it contains the extradata from the
139 ///< demuxer. During decoding, it contains
140 ///< packet data.
141 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
142
143 int spillover_bitsize; ///< number of bits used to specify
144 ///< #spillover_nbits in the packet header
145 ///< = ceil(log2(ctx->block_align << 3))
146 int history_nsamples; ///< number of samples in history for signal
147 ///< prediction (through ACB)
148
149 /* postfilter specific values */
150 int do_apf; ///< whether to apply the averaged
151 ///< projection filter (APF)
152 int denoise_strength; ///< strength of denoising in Wiener filter
153 ///< [0-11]
154 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
155 ///< Wiener filter coefficients (postfilter)
156 int dc_level; ///< Predicted amount of DC noise, based
157 ///< on which a DC removal filter is used
158
159 int lsps; ///< number of LSPs per frame [10 or 16]
160 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
161 int lsp_def_mode; ///< defines different sets of LSP defaults
162 ///< [0, 1]
163
164 int min_pitch_val; ///< base value for pitch parsing code
165 int max_pitch_val; ///< max value + 1 for pitch parsing
166 int pitch_nbits; ///< number of bits used to specify the
167 ///< pitch value in the frame header
168 int block_pitch_nbits; ///< number of bits used to specify the
169 ///< first block's pitch value
170 int block_pitch_range; ///< range of the block pitch
171 int block_delta_pitch_nbits; ///< number of bits used to specify the
172 ///< delta pitch between this and the last
173 ///< block's pitch value, used in all but
174 ///< first block
175 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
176 ///< from -this to +this-1)
177 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
178 ///< conversion
179
180 /**
181 * @}
182 *
183 * @name Packet values specified in the packet header or related to a packet.
184 *
185 * A packet is considered to be a single unit of data provided to this
186 * decoder by the demuxer.
187 * @{
188 */
189 int spillover_nbits; ///< number of bits of the previous packet's
190 ///< last superframe preceding this
191 ///< packet's first full superframe (useful
192 ///< for re-synchronization also)
193 int has_residual_lsps; ///< if set, superframes contain one set of
194 ///< LSPs that cover all frames, encoded as
195 ///< independent and residual LSPs; if not
196 ///< set, each frame contains its own, fully
197 ///< independent, LSPs
198 int skip_bits_next; ///< number of bits to skip at the next call
199 ///< to #wmavoice_decode_packet() (since
200 ///< they're part of the previous superframe)
201
202 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + AV_INPUT_BUFFER_PADDING_SIZE];
203 ///< cache for superframe data split over
204 ///< multiple packets
205 int sframe_cache_size; ///< set to >0 if we have data from an
206 ///< (incomplete) superframe from a previous
207 ///< packet that spilled over in the current
208 ///< packet; specifies the amount of bits in
209 ///< #sframe_cache
210 PutBitContext pb; ///< bitstream writer for #sframe_cache
211
212 /**
213 * @}
214 *
215 * @name Frame and superframe values
216 * Superframe and frame data - these can change from frame to frame,
217 * although some of them do in that case serve as a cache / history for
218 * the next frame or superframe.
219 * @{
220 */
221 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
222 ///< superframe
223 int last_pitch_val; ///< pitch value of the previous frame
224 int last_acb_type; ///< frame type [0-2] of the previous frame
225 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
226 ///< << 16) / #MAX_FRAMESIZE
227 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
228
229 int aw_idx_is_ext; ///< whether the AW index was encoded in
230 ///< 8 bits (instead of 6)
231 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
232 ///< can apply the pulse, relative to the
233 ///< value in aw_first_pulse_off. The exact
234 ///< position of the first AW-pulse is within
235 ///< [pulse_off, pulse_off + this], and
236 ///< depends on bitstream values; [16 or 24]
237 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
238 ///< that this number can be negative (in
239 ///< which case it basically means "zero")
240 int aw_first_pulse_off[2]; ///< index of first sample to which to
241 ///< apply AW-pulses, or -0xff if unset
242 int aw_next_pulse_off_cache; ///< the position (relative to start of the
243 ///< second block) at which pulses should
244 ///< start to be positioned, serves as a
245 ///< cache for pitch-adaptive window pulses
246 ///< between blocks
247
248 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
249 ///< only used for comfort noise in #pRNG()
250 int nb_superframes; ///< number of superframes in current packet
251 float gain_pred_err[6]; ///< cache for gain prediction
252 float excitation_history[MAX_SIGNAL_HISTORY];
253 ///< cache of the signal of previous
254 ///< superframes, used as a history for
255 ///< signal generation
256 float synth_history[MAX_LSPS]; ///< see #excitation_history
257 /**
258 * @}
259 *
260 * @name Postfilter values
261 *
262 * Variables used for postfilter implementation, mostly history for
263 * smoothing and so on, and context variables for FFT/iFFT.
264 * @{
265 */
266 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
267 ///< postfilter (for denoise filter)
268 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
269 ///< transform, part of postfilter)
270 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
271 ///< range
272 float postfilter_agc; ///< gain control memory, used in
273 ///< #adaptive_gain_control()
274 float dcf_mem[2]; ///< DC filter history
275 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
276 ///< zero filter output (i.e. excitation)
277 ///< by postfilter
278 float denoise_filter_cache[MAX_FRAMESIZE];
279 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
280 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
281 ///< aligned buffer for LPC tilting
282 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
283 ///< aligned buffer for denoise coefficients
284 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
285 ///< aligned buffer for postfilter speech
286 ///< synthesis
287 /**
288 * @}
289 */
290 } WMAVoiceContext;
291
292 /**
293 * Set up the variable bit mode (VBM) tree from container extradata.
294 * @param gb bit I/O context.
295 * The bit context (s->gb) should be loaded with byte 23-46 of the
296 * container extradata (i.e. the ones containing the VBM tree).
297 * @param vbm_tree pointer to array to which the decoded VBM tree will be
298 * written.
299 * @return 0 on success, <0 on error.
300 */
decode_vbmtree(GetBitContext * gb,int8_t vbm_tree[25])301 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
302 {
303 int cntr[8] = { 0 }, n, res;
304
305 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
306 for (n = 0; n < 17; n++) {
307 res = get_bits(gb, 3);
308 if (cntr[res] > 3) // should be >= 3 + (res == 7))
309 return -1;
310 vbm_tree[res * 3 + cntr[res]++] = n;
311 }
312 return 0;
313 }
314
wmavoice_init_static_data(void)315 static av_cold void wmavoice_init_static_data(void)
316 {
317 static const uint8_t bits[] = {
318 2, 2, 2, 4, 4, 4,
319 6, 6, 6, 8, 8, 8,
320 10, 10, 10, 12, 12, 12,
321 14, 14, 14, 14
322 };
323 static const uint16_t codes[] = {
324 0x0000, 0x0001, 0x0002, // 00/01/10
325 0x000c, 0x000d, 0x000e, // 11+00/01/10
326 0x003c, 0x003d, 0x003e, // 1111+00/01/10
327 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
328 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
329 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
330 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
331 };
332
333 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
334 bits, 1, 1, codes, 2, 2, 132);
335 }
336
wmavoice_flush(AVCodecContext * ctx)337 static av_cold void wmavoice_flush(AVCodecContext *ctx)
338 {
339 WMAVoiceContext *s = ctx->priv_data;
340 int n;
341
342 s->postfilter_agc = 0;
343 s->sframe_cache_size = 0;
344 s->skip_bits_next = 0;
345 for (n = 0; n < s->lsps; n++)
346 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
347 memset(s->excitation_history, 0,
348 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
349 memset(s->synth_history, 0,
350 sizeof(*s->synth_history) * MAX_LSPS);
351 memset(s->gain_pred_err, 0,
352 sizeof(s->gain_pred_err));
353
354 if (s->do_apf) {
355 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
356 sizeof(*s->synth_filter_out_buf) * s->lsps);
357 memset(s->dcf_mem, 0,
358 sizeof(*s->dcf_mem) * 2);
359 memset(s->zero_exc_pf, 0,
360 sizeof(*s->zero_exc_pf) * s->history_nsamples);
361 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
362 }
363 }
364
365 /**
366 * Set up decoder with parameters from demuxer (extradata etc.).
367 */
wmavoice_decode_init(AVCodecContext * ctx)368 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
369 {
370 static AVOnce init_static_once = AV_ONCE_INIT;
371 int n, flags, pitch_range, lsp16_flag, ret;
372 WMAVoiceContext *s = ctx->priv_data;
373
374 ff_thread_once(&init_static_once, wmavoice_init_static_data);
375
376 /**
377 * Extradata layout:
378 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
379 * - byte 19-22: flags field (annoyingly in LE; see below for known
380 * values),
381 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
382 * rest is 0).
383 */
384 if (ctx->extradata_size != 46) {
385 av_log(ctx, AV_LOG_ERROR,
386 "Invalid extradata size %d (should be 46)\n",
387 ctx->extradata_size);
388 return AVERROR_INVALIDDATA;
389 }
390 if (ctx->block_align <= 0 || ctx->block_align > (1<<22)) {
391 av_log(ctx, AV_LOG_ERROR, "Invalid block alignment %d.\n", ctx->block_align);
392 return AVERROR_INVALIDDATA;
393 }
394
395 flags = AV_RL32(ctx->extradata + 18);
396 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
397 s->do_apf = flags & 0x1;
398 if (s->do_apf) {
399 if ((ret = ff_rdft_init(&s->rdft, 7, DFT_R2C)) < 0 ||
400 (ret = ff_rdft_init(&s->irdft, 7, IDFT_C2R)) < 0 ||
401 (ret = ff_dct_init (&s->dct, 6, DCT_I)) < 0 ||
402 (ret = ff_dct_init (&s->dst, 6, DST_I)) < 0)
403 return ret;
404
405 ff_sine_window_init(s->cos, 256);
406 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
407 for (n = 0; n < 255; n++) {
408 s->sin[n] = -s->sin[510 - n];
409 s->cos[510 - n] = s->cos[n];
410 }
411 }
412 s->denoise_strength = (flags >> 2) & 0xF;
413 if (s->denoise_strength >= 12) {
414 av_log(ctx, AV_LOG_ERROR,
415 "Invalid denoise filter strength %d (max=11)\n",
416 s->denoise_strength);
417 return AVERROR_INVALIDDATA;
418 }
419 s->denoise_tilt_corr = !!(flags & 0x40);
420 s->dc_level = (flags >> 7) & 0xF;
421 s->lsp_q_mode = !!(flags & 0x2000);
422 s->lsp_def_mode = !!(flags & 0x4000);
423 lsp16_flag = flags & 0x1000;
424 if (lsp16_flag) {
425 s->lsps = 16;
426 } else {
427 s->lsps = 10;
428 }
429 for (n = 0; n < s->lsps; n++)
430 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
431
432 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
433 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
434 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
435 return AVERROR_INVALIDDATA;
436 }
437
438 if (ctx->sample_rate >= INT_MAX / (256 * 37))
439 return AVERROR_INVALIDDATA;
440
441 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
442 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
443 pitch_range = s->max_pitch_val - s->min_pitch_val;
444 if (pitch_range <= 0) {
445 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
446 return AVERROR_INVALIDDATA;
447 }
448 s->pitch_nbits = av_ceil_log2(pitch_range);
449 s->last_pitch_val = 40;
450 s->last_acb_type = ACB_TYPE_NONE;
451 s->history_nsamples = s->max_pitch_val + 8;
452
453 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
454 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
455 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
456
457 av_log(ctx, AV_LOG_ERROR,
458 "Unsupported samplerate %d (min=%d, max=%d)\n",
459 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
460
461 return AVERROR(ENOSYS);
462 }
463
464 s->block_conv_table[0] = s->min_pitch_val;
465 s->block_conv_table[1] = (pitch_range * 25) >> 6;
466 s->block_conv_table[2] = (pitch_range * 44) >> 6;
467 s->block_conv_table[3] = s->max_pitch_val - 1;
468 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
469 if (s->block_delta_pitch_hrange <= 0) {
470 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
471 return AVERROR_INVALIDDATA;
472 }
473 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
474 s->block_pitch_range = s->block_conv_table[2] +
475 s->block_conv_table[3] + 1 +
476 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
477 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
478
479 av_channel_layout_uninit(&ctx->ch_layout);
480 ctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
481 ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
482
483 return 0;
484 }
485
486 /**
487 * @name Postfilter functions
488 * Postfilter functions (gain control, wiener denoise filter, DC filter,
489 * kalman smoothening, plus surrounding code to wrap it)
490 * @{
491 */
492 /**
493 * Adaptive gain control (as used in postfilter).
494 *
495 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
496 * that the energy here is calculated using sum(abs(...)), whereas the
497 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
498 *
499 * @param out output buffer for filtered samples
500 * @param in input buffer containing the samples as they are after the
501 * postfilter steps so far
502 * @param speech_synth input buffer containing speech synth before postfilter
503 * @param size input buffer size
504 * @param alpha exponential filter factor
505 * @param gain_mem pointer to filter memory (single float)
506 */
adaptive_gain_control(float * out,const float * in,const float * speech_synth,int size,float alpha,float * gain_mem)507 static void adaptive_gain_control(float *out, const float *in,
508 const float *speech_synth,
509 int size, float alpha, float *gain_mem)
510 {
511 int i;
512 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
513 float mem = *gain_mem;
514
515 for (i = 0; i < size; i++) {
516 speech_energy += fabsf(speech_synth[i]);
517 postfilter_energy += fabsf(in[i]);
518 }
519 gain_scale_factor = postfilter_energy == 0.0 ? 0.0 :
520 (1.0 - alpha) * speech_energy / postfilter_energy;
521
522 for (i = 0; i < size; i++) {
523 mem = alpha * mem + gain_scale_factor;
524 out[i] = in[i] * mem;
525 }
526
527 *gain_mem = mem;
528 }
529
530 /**
531 * Kalman smoothing function.
532 *
533 * This function looks back pitch +/- 3 samples back into history to find
534 * the best fitting curve (that one giving the optimal gain of the two
535 * signals, i.e. the highest dot product between the two), and then
536 * uses that signal history to smoothen the output of the speech synthesis
537 * filter.
538 *
539 * @param s WMA Voice decoding context
540 * @param pitch pitch of the speech signal
541 * @param in input speech signal
542 * @param out output pointer for smoothened signal
543 * @param size input/output buffer size
544 *
545 * @returns -1 if no smoothening took place, e.g. because no optimal
546 * fit could be found, or 0 on success.
547 */
kalman_smoothen(WMAVoiceContext * s,int pitch,const float * in,float * out,int size)548 static int kalman_smoothen(WMAVoiceContext *s, int pitch,
549 const float *in, float *out, int size)
550 {
551 int n;
552 float optimal_gain = 0, dot;
553 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
554 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
555 *best_hist_ptr = NULL;
556
557 /* find best fitting point in history */
558 do {
559 dot = avpriv_scalarproduct_float_c(in, ptr, size);
560 if (dot > optimal_gain) {
561 optimal_gain = dot;
562 best_hist_ptr = ptr;
563 }
564 } while (--ptr >= end);
565
566 if (optimal_gain <= 0)
567 return -1;
568 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
569 if (dot <= 0) // would be 1.0
570 return -1;
571
572 if (optimal_gain <= dot) {
573 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
574 } else
575 dot = 0.625;
576
577 /* actual smoothing */
578 for (n = 0; n < size; n++)
579 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
580
581 return 0;
582 }
583
584 /**
585 * Get the tilt factor of a formant filter from its transfer function
586 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
587 * but somehow (??) it does a speech synthesis filter in the
588 * middle, which is missing here
589 *
590 * @param lpcs LPC coefficients
591 * @param n_lpcs Size of LPC buffer
592 * @returns the tilt factor
593 */
tilt_factor(const float * lpcs,int n_lpcs)594 static float tilt_factor(const float *lpcs, int n_lpcs)
595 {
596 float rh0, rh1;
597
598 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
599 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
600
601 return rh1 / rh0;
602 }
603
604 /**
605 * Derive denoise filter coefficients (in real domain) from the LPCs.
606 */
calc_input_response(WMAVoiceContext * s,float * lpcs,int fcb_type,float * coeffs,int remainder)607 static void calc_input_response(WMAVoiceContext *s, float *lpcs,
608 int fcb_type, float *coeffs, int remainder)
609 {
610 float last_coeff, min = 15.0, max = -15.0;
611 float irange, angle_mul, gain_mul, range, sq;
612 int n, idx;
613
614 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
615 s->rdft.rdft_calc(&s->rdft, lpcs);
616 #define log_range(var, assign) do { \
617 float tmp = log10f(assign); var = tmp; \
618 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
619 } while (0)
620 log_range(last_coeff, lpcs[1] * lpcs[1]);
621 for (n = 1; n < 64; n++)
622 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
623 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
624 log_range(lpcs[0], lpcs[0] * lpcs[0]);
625 #undef log_range
626 range = max - min;
627 lpcs[64] = last_coeff;
628
629 /* Now, use this spectrum to pick out these frequencies with higher
630 * (relative) power/energy (which we then take to be "not noise"),
631 * and set up a table (still in lpc[]) of (relative) gains per frequency.
632 * These frequencies will be maintained, while others ("noise") will be
633 * decreased in the filter output. */
634 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
635 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
636 (5.0 / 14.7));
637 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
638 for (n = 0; n <= 64; n++) {
639 float pwr;
640
641 idx = lrint((max - lpcs[n]) * irange - 1);
642 idx = FFMAX(0, idx);
643 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
644 lpcs[n] = angle_mul * pwr;
645
646 /* 70.57 =~ 1/log10(1.0331663) */
647 idx = av_clipf((pwr * gain_mul - 0.0295) * 70.570526123, 0, INT_MAX / 2);
648
649 if (idx > 127) { // fall back if index falls outside table range
650 coeffs[n] = wmavoice_energy_table[127] *
651 powf(1.0331663, idx - 127);
652 } else
653 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
654 }
655
656 /* calculate the Hilbert transform of the gains, which we do (since this
657 * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
658 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
659 * "moment" of the LPCs in this filter. */
660 s->dct.dct_calc(&s->dct, lpcs);
661 s->dst.dct_calc(&s->dst, lpcs);
662
663 /* Split out the coefficient indexes into phase/magnitude pairs */
664 idx = 255 + av_clip(lpcs[64], -255, 255);
665 coeffs[0] = coeffs[0] * s->cos[idx];
666 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
667 last_coeff = coeffs[64] * s->cos[idx];
668 for (n = 63;; n--) {
669 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
670 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
671 coeffs[n * 2] = coeffs[n] * s->cos[idx];
672
673 if (!--n) break;
674
675 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
676 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
677 coeffs[n * 2] = coeffs[n] * s->cos[idx];
678 }
679 coeffs[1] = last_coeff;
680
681 /* move into real domain */
682 s->irdft.rdft_calc(&s->irdft, coeffs);
683
684 /* tilt correction and normalize scale */
685 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
686 if (s->denoise_tilt_corr) {
687 float tilt_mem = 0;
688
689 coeffs[remainder - 1] = 0;
690 ff_tilt_compensation(&tilt_mem,
691 -1.8 * tilt_factor(coeffs, remainder - 1),
692 coeffs, remainder);
693 }
694 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
695 remainder));
696 for (n = 0; n < remainder; n++)
697 coeffs[n] *= sq;
698 }
699
700 /**
701 * This function applies a Wiener filter on the (noisy) speech signal as
702 * a means to denoise it.
703 *
704 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
705 * - using this power spectrum, calculate (for each frequency) the Wiener
706 * filter gain, which depends on the frequency power and desired level
707 * of noise subtraction (when set too high, this leads to artifacts)
708 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
709 * of 4-8kHz);
710 * - by doing a phase shift, calculate the Hilbert transform of this array
711 * of per-frequency filter-gains to get the filtering coefficients;
712 * - smoothen/normalize/de-tilt these filter coefficients as desired;
713 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
714 * to get the denoised speech signal;
715 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
716 * the frame boundary) are saved and applied to subsequent frames by an
717 * overlap-add method (otherwise you get clicking-artifacts).
718 *
719 * @param s WMA Voice decoding context
720 * @param fcb_type Frame (codebook) type
721 * @param synth_pf input: the noisy speech signal, output: denoised speech
722 * data; should be 16-byte aligned (for ASM purposes)
723 * @param size size of the speech data
724 * @param lpcs LPCs used to synthesize this frame's speech data
725 */
wiener_denoise(WMAVoiceContext * s,int fcb_type,float * synth_pf,int size,const float * lpcs)726 static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
727 float *synth_pf, int size,
728 const float *lpcs)
729 {
730 int remainder, lim, n;
731
732 if (fcb_type != FCB_TYPE_SILENCE) {
733 float *tilted_lpcs = s->tilted_lpcs_pf,
734 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
735
736 tilted_lpcs[0] = 1.0;
737 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
738 memset(&tilted_lpcs[s->lsps + 1], 0,
739 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
740 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
741 tilted_lpcs, s->lsps + 2);
742
743 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
744 * size is applied to the next frame. All input beyond this is zero,
745 * and thus all output beyond this will go towards zero, hence we can
746 * limit to min(size-1, 127-size) as a performance consideration. */
747 remainder = FFMIN(127 - size, size - 1);
748 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
749
750 /* apply coefficients (in frequency spectrum domain), i.e. complex
751 * number multiplication */
752 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
753 s->rdft.rdft_calc(&s->rdft, synth_pf);
754 s->rdft.rdft_calc(&s->rdft, coeffs);
755 synth_pf[0] *= coeffs[0];
756 synth_pf[1] *= coeffs[1];
757 for (n = 1; n < 64; n++) {
758 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
759 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
760 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
761 }
762 s->irdft.rdft_calc(&s->irdft, synth_pf);
763 }
764
765 /* merge filter output with the history of previous runs */
766 if (s->denoise_filter_cache_size) {
767 lim = FFMIN(s->denoise_filter_cache_size, size);
768 for (n = 0; n < lim; n++)
769 synth_pf[n] += s->denoise_filter_cache[n];
770 s->denoise_filter_cache_size -= lim;
771 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
772 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
773 }
774
775 /* move remainder of filter output into a cache for future runs */
776 if (fcb_type != FCB_TYPE_SILENCE) {
777 lim = FFMIN(remainder, s->denoise_filter_cache_size);
778 for (n = 0; n < lim; n++)
779 s->denoise_filter_cache[n] += synth_pf[size + n];
780 if (lim < remainder) {
781 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
782 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
783 s->denoise_filter_cache_size = remainder;
784 }
785 }
786 }
787
788 /**
789 * Averaging projection filter, the postfilter used in WMAVoice.
790 *
791 * This uses the following steps:
792 * - A zero-synthesis filter (generate excitation from synth signal)
793 * - Kalman smoothing on excitation, based on pitch
794 * - Re-synthesized smoothened output
795 * - Iterative Wiener denoise filter
796 * - Adaptive gain filter
797 * - DC filter
798 *
799 * @param s WMAVoice decoding context
800 * @param synth Speech synthesis output (before postfilter)
801 * @param samples Output buffer for filtered samples
802 * @param size Buffer size of synth & samples
803 * @param lpcs Generated LPCs used for speech synthesis
804 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
805 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
806 * @param pitch Pitch of the input signal
807 */
postfilter(WMAVoiceContext * s,const float * synth,float * samples,int size,const float * lpcs,float * zero_exc_pf,int fcb_type,int pitch)808 static void postfilter(WMAVoiceContext *s, const float *synth,
809 float *samples, int size,
810 const float *lpcs, float *zero_exc_pf,
811 int fcb_type, int pitch)
812 {
813 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
814 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
815 *synth_filter_in = zero_exc_pf;
816
817 av_assert0(size <= MAX_FRAMESIZE / 2);
818
819 /* generate excitation from input signal */
820 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
821
822 if (fcb_type >= FCB_TYPE_AW_PULSES &&
823 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
824 synth_filter_in = synth_filter_in_buf;
825
826 /* re-synthesize speech after smoothening, and keep history */
827 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
828 synth_filter_in, size, s->lsps);
829 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
830 sizeof(synth_pf[0]) * s->lsps);
831
832 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
833
834 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
835 &s->postfilter_agc);
836
837 if (s->dc_level > 8) {
838 /* remove ultra-low frequency DC noise / highpass filter;
839 * coefficients are identical to those used in SIPR decoding,
840 * and very closely resemble those used in AMR-NB decoding. */
841 ff_acelp_apply_order_2_transfer_function(samples, samples,
842 (const float[2]) { -1.99997, 1.0 },
843 (const float[2]) { -1.9330735188, 0.93589198496 },
844 0.93980580475, s->dcf_mem, size);
845 }
846 }
847 /**
848 * @}
849 */
850
851 /**
852 * Dequantize LSPs
853 * @param lsps output pointer to the array that will hold the LSPs
854 * @param num number of LSPs to be dequantized
855 * @param values quantized values, contains n_stages values
856 * @param sizes range (i.e. max value) of each quantized value
857 * @param n_stages number of dequantization runs
858 * @param table dequantization table to be used
859 * @param mul_q LSF multiplier
860 * @param base_q base (lowest) LSF values
861 */
dequant_lsps(double * lsps,int num,const uint16_t * values,const uint16_t * sizes,int n_stages,const uint8_t * table,const double * mul_q,const double * base_q)862 static void dequant_lsps(double *lsps, int num,
863 const uint16_t *values,
864 const uint16_t *sizes,
865 int n_stages, const uint8_t *table,
866 const double *mul_q,
867 const double *base_q)
868 {
869 int n, m;
870
871 memset(lsps, 0, num * sizeof(*lsps));
872 for (n = 0; n < n_stages; n++) {
873 const uint8_t *t_off = &table[values[n] * num];
874 double base = base_q[n], mul = mul_q[n];
875
876 for (m = 0; m < num; m++)
877 lsps[m] += base + mul * t_off[m];
878
879 table += sizes[n] * num;
880 }
881 }
882
883 /**
884 * @name LSP dequantization routines
885 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
886 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
887 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
888 * @{
889 */
890 /**
891 * Parse 10 independently-coded LSPs.
892 */
dequant_lsp10i(GetBitContext * gb,double * lsps)893 static void dequant_lsp10i(GetBitContext *gb, double *lsps)
894 {
895 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
896 static const double mul_lsf[4] = {
897 5.2187144800e-3, 1.4626986422e-3,
898 9.6179549166e-4, 1.1325736225e-3
899 };
900 static const double base_lsf[4] = {
901 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
902 M_PI * -3.3486e-2, M_PI * -5.7408e-2
903 };
904 uint16_t v[4];
905
906 v[0] = get_bits(gb, 8);
907 v[1] = get_bits(gb, 6);
908 v[2] = get_bits(gb, 5);
909 v[3] = get_bits(gb, 5);
910
911 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
912 mul_lsf, base_lsf);
913 }
914
915 /**
916 * Parse 10 independently-coded LSPs, and then derive the tables to
917 * generate LSPs for the other frames from them (residual coding).
918 */
dequant_lsp10r(GetBitContext * gb,double * i_lsps,const double * old,double * a1,double * a2,int q_mode)919 static void dequant_lsp10r(GetBitContext *gb,
920 double *i_lsps, const double *old,
921 double *a1, double *a2, int q_mode)
922 {
923 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
924 static const double mul_lsf[3] = {
925 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
926 };
927 static const double base_lsf[3] = {
928 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
929 };
930 const float (*ipol_tab)[2][10] = q_mode ?
931 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
932 uint16_t interpol, v[3];
933 int n;
934
935 dequant_lsp10i(gb, i_lsps);
936
937 interpol = get_bits(gb, 5);
938 v[0] = get_bits(gb, 7);
939 v[1] = get_bits(gb, 6);
940 v[2] = get_bits(gb, 6);
941
942 for (n = 0; n < 10; n++) {
943 double delta = old[n] - i_lsps[n];
944 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
945 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
946 }
947
948 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
949 mul_lsf, base_lsf);
950 }
951
952 /**
953 * Parse 16 independently-coded LSPs.
954 */
dequant_lsp16i(GetBitContext * gb,double * lsps)955 static void dequant_lsp16i(GetBitContext *gb, double *lsps)
956 {
957 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
958 static const double mul_lsf[5] = {
959 3.3439586280e-3, 6.9908173703e-4,
960 3.3216608306e-3, 1.0334960326e-3,
961 3.1899104283e-3
962 };
963 static const double base_lsf[5] = {
964 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
965 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
966 M_PI * -1.29816e-1
967 };
968 uint16_t v[5];
969
970 v[0] = get_bits(gb, 8);
971 v[1] = get_bits(gb, 6);
972 v[2] = get_bits(gb, 7);
973 v[3] = get_bits(gb, 6);
974 v[4] = get_bits(gb, 7);
975
976 dequant_lsps( lsps, 5, v, vec_sizes, 2,
977 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
978 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
979 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
980 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
981 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
982 }
983
984 /**
985 * Parse 16 independently-coded LSPs, and then derive the tables to
986 * generate LSPs for the other frames from them (residual coding).
987 */
dequant_lsp16r(GetBitContext * gb,double * i_lsps,const double * old,double * a1,double * a2,int q_mode)988 static void dequant_lsp16r(GetBitContext *gb,
989 double *i_lsps, const double *old,
990 double *a1, double *a2, int q_mode)
991 {
992 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
993 static const double mul_lsf[3] = {
994 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
995 };
996 static const double base_lsf[3] = {
997 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
998 };
999 const float (*ipol_tab)[2][16] = q_mode ?
1000 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
1001 uint16_t interpol, v[3];
1002 int n;
1003
1004 dequant_lsp16i(gb, i_lsps);
1005
1006 interpol = get_bits(gb, 5);
1007 v[0] = get_bits(gb, 7);
1008 v[1] = get_bits(gb, 7);
1009 v[2] = get_bits(gb, 7);
1010
1011 for (n = 0; n < 16; n++) {
1012 double delta = old[n] - i_lsps[n];
1013 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
1014 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
1015 }
1016
1017 dequant_lsps( a2, 10, v, vec_sizes, 1,
1018 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
1019 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
1020 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
1021 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
1022 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
1023 }
1024
1025 /**
1026 * @}
1027 * @name Pitch-adaptive window coding functions
1028 * The next few functions are for pitch-adaptive window coding.
1029 * @{
1030 */
1031 /**
1032 * Parse the offset of the first pitch-adaptive window pulses, and
1033 * the distribution of pulses between the two blocks in this frame.
1034 * @param s WMA Voice decoding context private data
1035 * @param gb bit I/O context
1036 * @param pitch pitch for each block in this frame
1037 */
aw_parse_coords(WMAVoiceContext * s,GetBitContext * gb,const int * pitch)1038 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
1039 const int *pitch)
1040 {
1041 static const int16_t start_offset[94] = {
1042 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1043 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1044 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1045 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1046 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1047 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1048 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1049 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1050 };
1051 int bits, offset;
1052
1053 /* position of pulse */
1054 s->aw_idx_is_ext = 0;
1055 if ((bits = get_bits(gb, 6)) >= 54) {
1056 s->aw_idx_is_ext = 1;
1057 bits += (bits - 54) * 3 + get_bits(gb, 2);
1058 }
1059
1060 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1061 * the distribution of the pulses in each block contained in this frame. */
1062 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1063 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1064 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1065 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1066 offset += s->aw_n_pulses[0] * pitch[0];
1067 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1068 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1069
1070 /* if continuing from a position before the block, reset position to
1071 * start of block (when corrected for the range over which it can be
1072 * spread in aw_pulse_set1()). */
1073 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1074 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1075 s->aw_first_pulse_off[1] -= pitch[1];
1076 if (start_offset[bits] < 0)
1077 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1078 s->aw_first_pulse_off[0] -= pitch[0];
1079 }
1080 }
1081
1082 /**
1083 * Apply second set of pitch-adaptive window pulses.
1084 * @param s WMA Voice decoding context private data
1085 * @param gb bit I/O context
1086 * @param block_idx block index in frame [0, 1]
1087 * @param fcb structure containing fixed codebook vector info
1088 * @return -1 on error, 0 otherwise
1089 */
aw_pulse_set2(WMAVoiceContext * s,GetBitContext * gb,int block_idx,AMRFixed * fcb)1090 static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1091 int block_idx, AMRFixed *fcb)
1092 {
1093 uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
1094 uint16_t *use_mask = use_mask_mem + 2;
1095 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1096 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1097 * of idx are the position of the bit within a particular item in the
1098 * array (0 being the most significant bit, and 15 being the least
1099 * significant bit), and the remainder (>> 4) is the index in the
1100 * use_mask[]-array. This is faster and uses less memory than using a
1101 * 80-byte/80-int array. */
1102 int pulse_off = s->aw_first_pulse_off[block_idx],
1103 pulse_start, n, idx, range, aidx, start_off = 0;
1104
1105 /* set offset of first pulse to within this block */
1106 if (s->aw_n_pulses[block_idx] > 0)
1107 while (pulse_off + s->aw_pulse_range < 1)
1108 pulse_off += fcb->pitch_lag;
1109
1110 /* find range per pulse */
1111 if (s->aw_n_pulses[0] > 0) {
1112 if (block_idx == 0) {
1113 range = 32;
1114 } else /* block_idx = 1 */ {
1115 range = 8;
1116 if (s->aw_n_pulses[block_idx] > 0)
1117 pulse_off = s->aw_next_pulse_off_cache;
1118 }
1119 } else
1120 range = 16;
1121 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1122
1123 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1124 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1125 * we exclude that range from being pulsed again in this function. */
1126 memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
1127 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
1128 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1129 if (s->aw_n_pulses[block_idx] > 0)
1130 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1131 int excl_range = s->aw_pulse_range; // always 16 or 24
1132 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1133 int first_sh = 16 - (idx & 15);
1134 *use_mask_ptr++ &= 0xFFFFu << first_sh;
1135 excl_range -= first_sh;
1136 if (excl_range >= 16) {
1137 *use_mask_ptr++ = 0;
1138 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1139 } else
1140 *use_mask_ptr &= 0xFFFF >> excl_range;
1141 }
1142
1143 /* find the 'aidx'th offset that is not excluded */
1144 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1145 for (n = 0; n <= aidx; pulse_start++) {
1146 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1147 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1148 if (use_mask[0]) idx = 0x0F;
1149 else if (use_mask[1]) idx = 0x1F;
1150 else if (use_mask[2]) idx = 0x2F;
1151 else if (use_mask[3]) idx = 0x3F;
1152 else if (use_mask[4]) idx = 0x4F;
1153 else return -1;
1154 idx -= av_log2_16bit(use_mask[idx >> 4]);
1155 }
1156 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1157 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1158 n++;
1159 start_off = idx;
1160 }
1161 }
1162
1163 fcb->x[fcb->n] = start_off;
1164 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1165 fcb->n++;
1166
1167 /* set offset for next block, relative to start of that block */
1168 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1169 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1170 return 0;
1171 }
1172
1173 /**
1174 * Apply first set of pitch-adaptive window pulses.
1175 * @param s WMA Voice decoding context private data
1176 * @param gb bit I/O context
1177 * @param block_idx block index in frame [0, 1]
1178 * @param fcb storage location for fixed codebook pulse info
1179 */
aw_pulse_set1(WMAVoiceContext * s,GetBitContext * gb,int block_idx,AMRFixed * fcb)1180 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1181 int block_idx, AMRFixed *fcb)
1182 {
1183 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1184 float v;
1185
1186 if (s->aw_n_pulses[block_idx] > 0) {
1187 int n, v_mask, i_mask, sh, n_pulses;
1188
1189 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1190 n_pulses = 3;
1191 v_mask = 8;
1192 i_mask = 7;
1193 sh = 4;
1194 } else { // 4 pulses, 1:sign + 2:index each
1195 n_pulses = 4;
1196 v_mask = 4;
1197 i_mask = 3;
1198 sh = 3;
1199 }
1200
1201 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1202 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1203 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1204 s->aw_first_pulse_off[block_idx];
1205 while (fcb->x[fcb->n] < 0)
1206 fcb->x[fcb->n] += fcb->pitch_lag;
1207 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1208 fcb->n++;
1209 }
1210 } else {
1211 int num2 = (val & 0x1FF) >> 1, delta, idx;
1212
1213 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1214 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1215 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1216 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1217 v = (val & 0x200) ? -1.0 : 1.0;
1218
1219 fcb->no_repeat_mask |= 3 << fcb->n;
1220 fcb->x[fcb->n] = idx - delta;
1221 fcb->y[fcb->n] = v;
1222 fcb->x[fcb->n + 1] = idx;
1223 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1224 fcb->n += 2;
1225 }
1226 }
1227
1228 /**
1229 * @}
1230 *
1231 * Generate a random number from frame_cntr and block_idx, which will live
1232 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1233 * table of size 1000 of which you want to read block_size entries).
1234 *
1235 * @param frame_cntr current frame number
1236 * @param block_num current block index
1237 * @param block_size amount of entries we want to read from a table
1238 * that has 1000 entries
1239 * @return a (non-)random number in the [0, 1000 - block_size] range.
1240 */
pRNG(int frame_cntr,int block_num,int block_size)1241 static int pRNG(int frame_cntr, int block_num, int block_size)
1242 {
1243 /* array to simplify the calculation of z:
1244 * y = (x % 9) * 5 + 6;
1245 * z = (49995 * x) / y;
1246 * Since y only has 9 values, we can remove the division by using a
1247 * LUT and using FASTDIV-style divisions. For each of the 9 values
1248 * of y, we can rewrite z as:
1249 * z = x * (49995 / y) + x * ((49995 % y) / y)
1250 * In this table, each col represents one possible value of y, the
1251 * first number is 49995 / y, and the second is the FASTDIV variant
1252 * of 49995 % y / y. */
1253 static const unsigned int div_tbl[9][2] = {
1254 { 8332, 3 * 715827883U }, // y = 6
1255 { 4545, 0 * 390451573U }, // y = 11
1256 { 3124, 11 * 268435456U }, // y = 16
1257 { 2380, 15 * 204522253U }, // y = 21
1258 { 1922, 23 * 165191050U }, // y = 26
1259 { 1612, 23 * 138547333U }, // y = 31
1260 { 1388, 27 * 119304648U }, // y = 36
1261 { 1219, 16 * 104755300U }, // y = 41
1262 { 1086, 39 * 93368855U } // y = 46
1263 };
1264 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1265 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1266 // so this is effectively a modulo (%)
1267 y = x - 9 * MULH(477218589, x); // x % 9
1268 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1269 // z = x * 49995 / (y * 5 + 6)
1270 return z % (1000 - block_size);
1271 }
1272
1273 /**
1274 * Parse hardcoded signal for a single block.
1275 * @note see #synth_block().
1276 */
synth_block_hardcoded(WMAVoiceContext * s,GetBitContext * gb,int block_idx,int size,const struct frame_type_desc * frame_desc,float * excitation)1277 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1278 int block_idx, int size,
1279 const struct frame_type_desc *frame_desc,
1280 float *excitation)
1281 {
1282 float gain;
1283 int n, r_idx;
1284
1285 av_assert0(size <= MAX_FRAMESIZE);
1286
1287 /* Set the offset from which we start reading wmavoice_std_codebook */
1288 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1289 r_idx = pRNG(s->frame_cntr, block_idx, size);
1290 gain = s->silence_gain;
1291 } else /* FCB_TYPE_HARDCODED */ {
1292 r_idx = get_bits(gb, 8);
1293 gain = wmavoice_gain_universal[get_bits(gb, 6)];
1294 }
1295
1296 /* Clear gain prediction parameters */
1297 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1298
1299 /* Apply gain to hardcoded codebook and use that as excitation signal */
1300 for (n = 0; n < size; n++)
1301 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1302 }
1303
1304 /**
1305 * Parse FCB/ACB signal for a single block.
1306 * @note see #synth_block().
1307 */
synth_block_fcb_acb(WMAVoiceContext * s,GetBitContext * gb,int block_idx,int size,int block_pitch_sh2,const struct frame_type_desc * frame_desc,float * excitation)1308 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1309 int block_idx, int size,
1310 int block_pitch_sh2,
1311 const struct frame_type_desc *frame_desc,
1312 float *excitation)
1313 {
1314 static const float gain_coeff[6] = {
1315 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1316 };
1317 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1318 int n, idx, gain_weight;
1319 AMRFixed fcb;
1320
1321 av_assert0(size <= MAX_FRAMESIZE / 2);
1322 memset(pulses, 0, sizeof(*pulses) * size);
1323
1324 fcb.pitch_lag = block_pitch_sh2 >> 2;
1325 fcb.pitch_fac = 1.0;
1326 fcb.no_repeat_mask = 0;
1327 fcb.n = 0;
1328
1329 /* For the other frame types, this is where we apply the innovation
1330 * (fixed) codebook pulses of the speech signal. */
1331 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1332 aw_pulse_set1(s, gb, block_idx, &fcb);
1333 if (aw_pulse_set2(s, gb, block_idx, &fcb)) {
1334 /* Conceal the block with silence and return.
1335 * Skip the correct amount of bits to read the next
1336 * block from the correct offset. */
1337 int r_idx = pRNG(s->frame_cntr, block_idx, size);
1338
1339 for (n = 0; n < size; n++)
1340 excitation[n] =
1341 wmavoice_std_codebook[r_idx + n] * s->silence_gain;
1342 skip_bits(gb, 7 + 1);
1343 return;
1344 }
1345 } else /* FCB_TYPE_EXC_PULSES */ {
1346 int offset_nbits = 5 - frame_desc->log_n_blocks;
1347
1348 fcb.no_repeat_mask = -1;
1349 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1350 * (instead of double) for a subset of pulses */
1351 for (n = 0; n < 5; n++) {
1352 float sign;
1353 int pos1, pos2;
1354
1355 sign = get_bits1(gb) ? 1.0 : -1.0;
1356 pos1 = get_bits(gb, offset_nbits);
1357 fcb.x[fcb.n] = n + 5 * pos1;
1358 fcb.y[fcb.n++] = sign;
1359 if (n < frame_desc->dbl_pulses) {
1360 pos2 = get_bits(gb, offset_nbits);
1361 fcb.x[fcb.n] = n + 5 * pos2;
1362 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1363 }
1364 }
1365 }
1366 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1367
1368 /* Calculate gain for adaptive & fixed codebook signal.
1369 * see ff_amr_set_fixed_gain(). */
1370 idx = get_bits(gb, 7);
1371 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
1372 gain_coeff, 6) -
1373 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1374 acb_gain = wmavoice_gain_codebook_acb[idx];
1375 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1376 -2.9957322736 /* log(0.05) */,
1377 1.6094379124 /* log(5.0) */);
1378
1379 gain_weight = 8 >> frame_desc->log_n_blocks;
1380 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1381 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1382 for (n = 0; n < gain_weight; n++)
1383 s->gain_pred_err[n] = pred_err;
1384
1385 /* Calculation of adaptive codebook */
1386 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1387 int len;
1388 for (n = 0; n < size; n += len) {
1389 int next_idx_sh16;
1390 int abs_idx = block_idx * size + n;
1391 int pitch_sh16 = (s->last_pitch_val << 16) +
1392 s->pitch_diff_sh16 * abs_idx;
1393 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1394 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1395 idx = idx_sh16 >> 16;
1396 if (s->pitch_diff_sh16) {
1397 if (s->pitch_diff_sh16 > 0) {
1398 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1399 } else
1400 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1401 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1402 1, size - n);
1403 } else
1404 len = size;
1405
1406 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1407 wmavoice_ipol1_coeffs, 17,
1408 idx, 9, len);
1409 }
1410 } else /* ACB_TYPE_HAMMING */ {
1411 int block_pitch = block_pitch_sh2 >> 2;
1412 idx = block_pitch_sh2 & 3;
1413 if (idx) {
1414 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1415 wmavoice_ipol2_coeffs, 4,
1416 idx, 8, size);
1417 } else
1418 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1419 sizeof(float) * size);
1420 }
1421
1422 /* Interpolate ACB/FCB and use as excitation signal */
1423 ff_weighted_vector_sumf(excitation, excitation, pulses,
1424 acb_gain, fcb_gain, size);
1425 }
1426
1427 /**
1428 * Parse data in a single block.
1429 *
1430 * @param s WMA Voice decoding context private data
1431 * @param gb bit I/O context
1432 * @param block_idx index of the to-be-read block
1433 * @param size amount of samples to be read in this block
1434 * @param block_pitch_sh2 pitch for this block << 2
1435 * @param lsps LSPs for (the end of) this frame
1436 * @param prev_lsps LSPs for the last frame
1437 * @param frame_desc frame type descriptor
1438 * @param excitation target memory for the ACB+FCB interpolated signal
1439 * @param synth target memory for the speech synthesis filter output
1440 * @return 0 on success, <0 on error.
1441 */
synth_block(WMAVoiceContext * s,GetBitContext * gb,int block_idx,int size,int block_pitch_sh2,const double * lsps,const double * prev_lsps,const struct frame_type_desc * frame_desc,float * excitation,float * synth)1442 static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1443 int block_idx, int size,
1444 int block_pitch_sh2,
1445 const double *lsps, const double *prev_lsps,
1446 const struct frame_type_desc *frame_desc,
1447 float *excitation, float *synth)
1448 {
1449 double i_lsps[MAX_LSPS];
1450 float lpcs[MAX_LSPS];
1451 float fac;
1452 int n;
1453
1454 if (frame_desc->acb_type == ACB_TYPE_NONE)
1455 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1456 else
1457 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1458 frame_desc, excitation);
1459
1460 /* convert interpolated LSPs to LPCs */
1461 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1462 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1463 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1464 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1465
1466 /* Speech synthesis */
1467 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1468 }
1469
1470 /**
1471 * Synthesize output samples for a single frame.
1472 *
1473 * @param ctx WMA Voice decoder context
1474 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1475 * @param frame_idx Frame number within superframe [0-2]
1476 * @param samples pointer to output sample buffer, has space for at least 160
1477 * samples
1478 * @param lsps LSP array
1479 * @param prev_lsps array of previous frame's LSPs
1480 * @param excitation target buffer for excitation signal
1481 * @param synth target buffer for synthesized speech data
1482 * @return 0 on success, <0 on error.
1483 */
synth_frame(AVCodecContext * ctx,GetBitContext * gb,int frame_idx,float * samples,const double * lsps,const double * prev_lsps,float * excitation,float * synth)1484 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1485 float *samples,
1486 const double *lsps, const double *prev_lsps,
1487 float *excitation, float *synth)
1488 {
1489 WMAVoiceContext *s = ctx->priv_data;
1490 int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val);
1491 int pitch[MAX_BLOCKS], av_uninit(last_block_pitch);
1492
1493 /* Parse frame type ("frame header"), see frame_descs */
1494 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
1495
1496 if (bd_idx < 0) {
1497 av_log(ctx, AV_LOG_ERROR,
1498 "Invalid frame type VLC code, skipping\n");
1499 return AVERROR_INVALIDDATA;
1500 }
1501
1502 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1503
1504 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1505 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1506 /* Pitch is provided per frame, which is interpreted as the pitch of
1507 * the last sample of the last block of this frame. We can interpolate
1508 * the pitch of other blocks (and even pitch-per-sample) by gradually
1509 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1510 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1511 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1512 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1513 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1514 if (s->last_acb_type == ACB_TYPE_NONE ||
1515 20 * abs(cur_pitch_val - s->last_pitch_val) >
1516 (cur_pitch_val + s->last_pitch_val))
1517 s->last_pitch_val = cur_pitch_val;
1518
1519 /* pitch per block */
1520 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1521 int fac = n * 2 + 1;
1522
1523 pitch[n] = (MUL16(fac, cur_pitch_val) +
1524 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1525 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1526 }
1527
1528 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1529 s->pitch_diff_sh16 =
1530 (cur_pitch_val - s->last_pitch_val) * (1 << 16) / MAX_FRAMESIZE;
1531 }
1532
1533 /* Global gain (if silence) and pitch-adaptive window coordinates */
1534 switch (frame_descs[bd_idx].fcb_type) {
1535 case FCB_TYPE_SILENCE:
1536 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1537 break;
1538 case FCB_TYPE_AW_PULSES:
1539 aw_parse_coords(s, gb, pitch);
1540 break;
1541 }
1542
1543 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1544 int bl_pitch_sh2;
1545
1546 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1547 switch (frame_descs[bd_idx].acb_type) {
1548 case ACB_TYPE_HAMMING: {
1549 /* Pitch is given per block. Per-block pitches are encoded as an
1550 * absolute value for the first block, and then delta values
1551 * relative to this value) for all subsequent blocks. The scale of
1552 * this pitch value is semi-logarithmic compared to its use in the
1553 * decoder, so we convert it to normal scale also. */
1554 int block_pitch,
1555 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1556 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1557 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1558
1559 if (n == 0) {
1560 block_pitch = get_bits(gb, s->block_pitch_nbits);
1561 } else
1562 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1563 get_bits(gb, s->block_delta_pitch_nbits);
1564 /* Convert last_ so that any next delta is within _range */
1565 last_block_pitch = av_clip(block_pitch,
1566 s->block_delta_pitch_hrange,
1567 s->block_pitch_range -
1568 s->block_delta_pitch_hrange);
1569
1570 /* Convert semi-log-style scale back to normal scale */
1571 if (block_pitch < t1) {
1572 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1573 } else {
1574 block_pitch -= t1;
1575 if (block_pitch < t2) {
1576 bl_pitch_sh2 =
1577 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1578 } else {
1579 block_pitch -= t2;
1580 if (block_pitch < t3) {
1581 bl_pitch_sh2 =
1582 (s->block_conv_table[2] + block_pitch) << 2;
1583 } else
1584 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1585 }
1586 }
1587 pitch[n] = bl_pitch_sh2 >> 2;
1588 break;
1589 }
1590
1591 case ACB_TYPE_ASYMMETRIC: {
1592 bl_pitch_sh2 = pitch[n] << 2;
1593 break;
1594 }
1595
1596 default: // ACB_TYPE_NONE has no pitch
1597 bl_pitch_sh2 = 0;
1598 break;
1599 }
1600
1601 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1602 lsps, prev_lsps, &frame_descs[bd_idx],
1603 &excitation[n * block_nsamples],
1604 &synth[n * block_nsamples]);
1605 }
1606
1607 /* Averaging projection filter, if applicable. Else, just copy samples
1608 * from synthesis buffer */
1609 if (s->do_apf) {
1610 double i_lsps[MAX_LSPS];
1611 float lpcs[MAX_LSPS];
1612
1613 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1614 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1615 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1616 postfilter(s, synth, samples, 80, lpcs,
1617 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1618 frame_descs[bd_idx].fcb_type, pitch[0]);
1619
1620 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1621 i_lsps[n] = cos(lsps[n]);
1622 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1623 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1624 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1625 frame_descs[bd_idx].fcb_type, pitch[0]);
1626 } else
1627 memcpy(samples, synth, 160 * sizeof(synth[0]));
1628
1629 /* Cache values for next frame */
1630 s->frame_cntr++;
1631 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1632 s->last_acb_type = frame_descs[bd_idx].acb_type;
1633 switch (frame_descs[bd_idx].acb_type) {
1634 case ACB_TYPE_NONE:
1635 s->last_pitch_val = 0;
1636 break;
1637 case ACB_TYPE_ASYMMETRIC:
1638 s->last_pitch_val = cur_pitch_val;
1639 break;
1640 case ACB_TYPE_HAMMING:
1641 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1642 break;
1643 }
1644
1645 return 0;
1646 }
1647
1648 /**
1649 * Ensure minimum value for first item, maximum value for last value,
1650 * proper spacing between each value and proper ordering.
1651 *
1652 * @param lsps array of LSPs
1653 * @param num size of LSP array
1654 *
1655 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1656 * useful to put in a generic location later on. Parts are also
1657 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1658 * which is in float.
1659 */
stabilize_lsps(double * lsps,int num)1660 static void stabilize_lsps(double *lsps, int num)
1661 {
1662 int n, m, l;
1663
1664 /* set minimum value for first, maximum value for last and minimum
1665 * spacing between LSF values.
1666 * Very similar to ff_set_min_dist_lsf(), but in double. */
1667 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1668 for (n = 1; n < num; n++)
1669 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1670 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1671
1672 /* reorder (looks like one-time / non-recursed bubblesort).
1673 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1674 for (n = 1; n < num; n++) {
1675 if (lsps[n] < lsps[n - 1]) {
1676 for (m = 1; m < num; m++) {
1677 double tmp = lsps[m];
1678 for (l = m - 1; l >= 0; l--) {
1679 if (lsps[l] <= tmp) break;
1680 lsps[l + 1] = lsps[l];
1681 }
1682 lsps[l + 1] = tmp;
1683 }
1684 break;
1685 }
1686 }
1687 }
1688
1689 /**
1690 * Synthesize output samples for a single superframe. If we have any data
1691 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1692 * in s->gb.
1693 *
1694 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1695 * to give a total of 480 samples per frame. See #synth_frame() for frame
1696 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1697 * (if these are globally specified for all frames (residually); they can
1698 * also be specified individually per-frame. See the s->has_residual_lsps
1699 * option), and can specify the number of samples encoded in this superframe
1700 * (if less than 480), usually used to prevent blanks at track boundaries.
1701 *
1702 * @param ctx WMA Voice decoder context
1703 * @return 0 on success, <0 on error or 1 if there was not enough data to
1704 * fully parse the superframe
1705 */
synth_superframe(AVCodecContext * ctx,AVFrame * frame,int * got_frame_ptr)1706 static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
1707 int *got_frame_ptr)
1708 {
1709 WMAVoiceContext *s = ctx->priv_data;
1710 GetBitContext *gb = &s->gb, s_gb;
1711 int n, res, n_samples = MAX_SFRAMESIZE;
1712 double lsps[MAX_FRAMES][MAX_LSPS];
1713 const double *mean_lsf = s->lsps == 16 ?
1714 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1715 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1716 float synth[MAX_LSPS + MAX_SFRAMESIZE];
1717 float *samples;
1718
1719 memcpy(synth, s->synth_history,
1720 s->lsps * sizeof(*synth));
1721 memcpy(excitation, s->excitation_history,
1722 s->history_nsamples * sizeof(*excitation));
1723
1724 if (s->sframe_cache_size > 0) {
1725 gb = &s_gb;
1726 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1727 s->sframe_cache_size = 0;
1728 }
1729
1730 /* First bit is speech/music bit, it differentiates between WMAVoice
1731 * speech samples (the actual codec) and WMAVoice music samples, which
1732 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1733 * the wild yet. */
1734 if (!get_bits1(gb)) {
1735 avpriv_request_sample(ctx, "WMAPro-in-WMAVoice");
1736 return AVERROR_PATCHWELCOME;
1737 }
1738
1739 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1740 if (get_bits1(gb)) {
1741 if ((n_samples = get_bits(gb, 12)) > MAX_SFRAMESIZE) {
1742 av_log(ctx, AV_LOG_ERROR,
1743 "Superframe encodes > %d samples (%d), not allowed\n",
1744 MAX_SFRAMESIZE, n_samples);
1745 return AVERROR_INVALIDDATA;
1746 }
1747 }
1748
1749 /* Parse LSPs, if global for the superframe (can also be per-frame). */
1750 if (s->has_residual_lsps) {
1751 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1752
1753 for (n = 0; n < s->lsps; n++)
1754 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1755
1756 if (s->lsps == 10) {
1757 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1758 } else /* s->lsps == 16 */
1759 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1760
1761 for (n = 0; n < s->lsps; n++) {
1762 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
1763 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1764 lsps[2][n] += mean_lsf[n];
1765 }
1766 for (n = 0; n < 3; n++)
1767 stabilize_lsps(lsps[n], s->lsps);
1768 }
1769
1770 /* synth_superframe can run multiple times per packet
1771 * free potential previous frame */
1772 av_frame_unref(frame);
1773
1774 /* get output buffer */
1775 frame->nb_samples = MAX_SFRAMESIZE;
1776 if ((res = ff_get_buffer(ctx, frame, 0)) < 0)
1777 return res;
1778 frame->nb_samples = n_samples;
1779 samples = (float *)frame->data[0];
1780
1781 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1782 for (n = 0; n < 3; n++) {
1783 if (!s->has_residual_lsps) {
1784 int m;
1785
1786 if (s->lsps == 10) {
1787 dequant_lsp10i(gb, lsps[n]);
1788 } else /* s->lsps == 16 */
1789 dequant_lsp16i(gb, lsps[n]);
1790
1791 for (m = 0; m < s->lsps; m++)
1792 lsps[n][m] += mean_lsf[m];
1793 stabilize_lsps(lsps[n], s->lsps);
1794 }
1795
1796 if ((res = synth_frame(ctx, gb, n,
1797 &samples[n * MAX_FRAMESIZE],
1798 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1799 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1800 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
1801 *got_frame_ptr = 0;
1802 return res;
1803 }
1804 }
1805
1806 /* Statistics? FIXME - we don't check for length, a slight overrun
1807 * will be caught by internal buffer padding, and anything else
1808 * will be skipped, not read. */
1809 if (get_bits1(gb)) {
1810 res = get_bits(gb, 4);
1811 skip_bits(gb, 10 * (res + 1));
1812 }
1813
1814 if (get_bits_left(gb) < 0) {
1815 wmavoice_flush(ctx);
1816 return AVERROR_INVALIDDATA;
1817 }
1818
1819 *got_frame_ptr = 1;
1820
1821 /* Update history */
1822 memcpy(s->prev_lsps, lsps[2],
1823 s->lsps * sizeof(*s->prev_lsps));
1824 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1825 s->lsps * sizeof(*synth));
1826 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1827 s->history_nsamples * sizeof(*excitation));
1828 if (s->do_apf)
1829 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1830 s->history_nsamples * sizeof(*s->zero_exc_pf));
1831
1832 return 0;
1833 }
1834
1835 /**
1836 * Parse the packet header at the start of each packet (input data to this
1837 * decoder).
1838 *
1839 * @param s WMA Voice decoding context private data
1840 * @return <0 on error, nb_superframes on success.
1841 */
parse_packet_header(WMAVoiceContext * s)1842 static int parse_packet_header(WMAVoiceContext *s)
1843 {
1844 GetBitContext *gb = &s->gb;
1845 unsigned int res, n_superframes = 0;
1846
1847 skip_bits(gb, 4); // packet sequence number
1848 s->has_residual_lsps = get_bits1(gb);
1849 do {
1850 if (get_bits_left(gb) < 6 + s->spillover_bitsize)
1851 return AVERROR_INVALIDDATA;
1852
1853 res = get_bits(gb, 6); // number of superframes per packet
1854 // (minus first one if there is spillover)
1855 n_superframes += res;
1856 } while (res == 0x3F);
1857 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
1858
1859 return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA;
1860 }
1861
1862 /**
1863 * Copy (unaligned) bits from gb/data/size to pb.
1864 *
1865 * @param pb target buffer to copy bits into
1866 * @param data source buffer to copy bits from
1867 * @param size size of the source data, in bytes
1868 * @param gb bit I/O context specifying the current position in the source.
1869 * data. This function might use this to align the bit position to
1870 * a whole-byte boundary before calling #ff_copy_bits() on aligned
1871 * source data
1872 * @param nbits the amount of bits to copy from source to target
1873 *
1874 * @note after calling this function, the current position in the input bit
1875 * I/O context is undefined.
1876 */
copy_bits(PutBitContext * pb,const uint8_t * data,int size,GetBitContext * gb,int nbits)1877 static void copy_bits(PutBitContext *pb,
1878 const uint8_t *data, int size,
1879 GetBitContext *gb, int nbits)
1880 {
1881 int rmn_bytes, rmn_bits;
1882
1883 rmn_bits = rmn_bytes = get_bits_left(gb);
1884 if (rmn_bits < nbits)
1885 return;
1886 if (nbits > put_bits_left(pb))
1887 return;
1888 rmn_bits &= 7; rmn_bytes >>= 3;
1889 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1890 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1891 ff_copy_bits(pb, data + size - rmn_bytes,
1892 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1893 }
1894
1895 /**
1896 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1897 * and we expect that the demuxer / application provides it to us as such
1898 * (else you'll probably get garbage as output). Every packet has a size of
1899 * ctx->block_align bytes, starts with a packet header (see
1900 * #parse_packet_header()), and then a series of superframes. Superframe
1901 * boundaries may exceed packets, i.e. superframes can split data over
1902 * multiple (two) packets.
1903 *
1904 * For more information about frames, see #synth_superframe().
1905 */
wmavoice_decode_packet(AVCodecContext * ctx,AVFrame * frame,int * got_frame_ptr,AVPacket * avpkt)1906 static int wmavoice_decode_packet(AVCodecContext *ctx, AVFrame *frame,
1907 int *got_frame_ptr, AVPacket *avpkt)
1908 {
1909 WMAVoiceContext *s = ctx->priv_data;
1910 GetBitContext *gb = &s->gb;
1911 int size, res, pos;
1912
1913 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1914 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1915 * feeds us ASF packets, which may concatenate multiple "codec" packets
1916 * in a single "muxer" packet, so we artificially emulate that by
1917 * capping the packet size at ctx->block_align. */
1918 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1919 init_get_bits8(&s->gb, avpkt->data, size);
1920
1921 /* size == ctx->block_align is used to indicate whether we are dealing with
1922 * a new packet or a packet of which we already read the packet header
1923 * previously. */
1924 if (!(size % ctx->block_align)) { // new packet header
1925 if (!size) {
1926 s->spillover_nbits = 0;
1927 s->nb_superframes = 0;
1928 } else {
1929 if ((res = parse_packet_header(s)) < 0)
1930 return res;
1931 s->nb_superframes = res;
1932 }
1933
1934 /* If the packet header specifies a s->spillover_nbits, then we want
1935 * to push out all data of the previous packet (+ spillover) before
1936 * continuing to parse new superframes in the current packet. */
1937 if (s->sframe_cache_size > 0) {
1938 int cnt = get_bits_count(gb);
1939 if (cnt + s->spillover_nbits > avpkt->size * 8) {
1940 s->spillover_nbits = avpkt->size * 8 - cnt;
1941 }
1942 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1943 flush_put_bits(&s->pb);
1944 s->sframe_cache_size += s->spillover_nbits;
1945 if ((res = synth_superframe(ctx, frame, got_frame_ptr)) == 0 &&
1946 *got_frame_ptr) {
1947 cnt += s->spillover_nbits;
1948 s->skip_bits_next = cnt & 7;
1949 res = cnt >> 3;
1950 return res;
1951 } else
1952 skip_bits_long (gb, s->spillover_nbits - cnt +
1953 get_bits_count(gb)); // resync
1954 } else if (s->spillover_nbits) {
1955 skip_bits_long(gb, s->spillover_nbits); // resync
1956 }
1957 } else if (s->skip_bits_next)
1958 skip_bits(gb, s->skip_bits_next);
1959
1960 /* Try parsing superframes in current packet */
1961 s->sframe_cache_size = 0;
1962 s->skip_bits_next = 0;
1963 pos = get_bits_left(gb);
1964 if (s->nb_superframes-- == 0) {
1965 *got_frame_ptr = 0;
1966 return size;
1967 } else if (s->nb_superframes > 0) {
1968 if ((res = synth_superframe(ctx, frame, got_frame_ptr)) < 0) {
1969 return res;
1970 } else if (*got_frame_ptr) {
1971 int cnt = get_bits_count(gb);
1972 s->skip_bits_next = cnt & 7;
1973 res = cnt >> 3;
1974 return res;
1975 }
1976 } else if ((s->sframe_cache_size = pos) > 0) {
1977 /* ... cache it for spillover in next packet */
1978 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
1979 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
1980 // FIXME bad - just copy bytes as whole and add use the
1981 // skip_bits_next field
1982 }
1983
1984 return size;
1985 }
1986
wmavoice_decode_end(AVCodecContext * ctx)1987 static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
1988 {
1989 WMAVoiceContext *s = ctx->priv_data;
1990
1991 if (s->do_apf) {
1992 ff_rdft_end(&s->rdft);
1993 ff_rdft_end(&s->irdft);
1994 ff_dct_end(&s->dct);
1995 ff_dct_end(&s->dst);
1996 }
1997
1998 return 0;
1999 }
2000
2001 const FFCodec ff_wmavoice_decoder = {
2002 .p.name = "wmavoice",
2003 .p.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
2004 .p.type = AVMEDIA_TYPE_AUDIO,
2005 .p.id = AV_CODEC_ID_WMAVOICE,
2006 .priv_data_size = sizeof(WMAVoiceContext),
2007 .init = wmavoice_decode_init,
2008 .close = wmavoice_decode_end,
2009 FF_CODEC_DECODE_CB(wmavoice_decode_packet),
2010 .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
2011 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
2012 .flush = wmavoice_flush,
2013 };
2014