Fix buffer overrun (or, well, actually a typo, 80 should be 0x80...).
[libav.git] / libavcodec / wmavoice.c
CommitLineData
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1/*
2 * Windows Media Audio Voice decoder.
3 * Copyright (c) 2009 Ronald S. Bultje
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
ba87f080 23 * @file
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24 * @brief Windows Media Audio Voice compatible decoder
25 * @author Ronald S. Bultje <rsbultje@gmail.com>
26 */
27
28#include <math.h>
29#include "avcodec.h"
30#include "get_bits.h"
31#include "put_bits.h"
32#include "wmavoice_data.h"
33#include "celp_math.h"
34#include "celp_filters.h"
35#include "acelp_vectors.h"
36#include "acelp_filters.h"
37#include "lsp.h"
38#include "libavutil/lzo.h"
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39#include "avfft.h"
40#include "fft.h"
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41
42#define MAX_BLOCKS 8 ///< maximum number of blocks per frame
43#define MAX_LSPS 16 ///< maximum filter order
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44#define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple
45 ///< of 16 for ASM input buffer alignment
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46#define MAX_FRAMES 3 ///< maximum number of frames per superframe
47#define MAX_FRAMESIZE 160 ///< maximum number of samples per frame
48#define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history
49#define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
50 ///< maximum number of samples per superframe
51#define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
52 ///< was split over two packets
53#define VLC_NBITS 6 ///< number of bits to read per VLC iteration
54
55/**
56 * Frame type VLC coding.
57 */
58static VLC frame_type_vlc;
59
60/**
61 * Adaptive codebook types.
62 */
63enum {
64 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed)
65 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
66 ///< we interpolate to get a per-sample pitch.
67 ///< Signal is generated using an asymmetric sinc
68 ///< window function
69 ///< @note see #wmavoice_ipol1_coeffs
70 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using
71 ///< a Hamming sinc window function
72 ///< @note see #wmavoice_ipol2_coeffs
73};
74
75/**
76 * Fixed codebook types.
77 */
78enum {
79 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence
80 ///< generated from a hardcoded (fixed) codebook
81 ///< with per-frame (low) gain values
82 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block
83 ///< gain values
84 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals,
85 ///< used in particular for low-bitrate streams
86 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
87 ///< combinations of either single pulses or
88 ///< pulse pairs
89};
90
91/**
92 * Description of frame types.
93 */
94static const struct frame_type_desc {
95 uint8_t n_blocks; ///< amount of blocks per frame (each block
96 ///< (contains 160/#n_blocks samples)
97 uint8_t log_n_blocks; ///< log2(#n_blocks)
98 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*)
99 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*)
100 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs
101 ///< (rather than just one single pulse)
102 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
103 uint16_t frame_size; ///< the amount of bits that make up the block
104 ///< data (per frame)
105} frame_descs[17] = {
106 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 },
107 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 },
108 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 },
109 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 },
110 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
111 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
112 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
113 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
114 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 },
115 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 },
116 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 },
117 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 },
118 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 },
119 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 },
120 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 },
121 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 },
122 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 }
123};
124
125/**
126 * WMA Voice decoding context.
127 */
128typedef struct {
129 /**
130 * @defgroup struct_global Global values
131 * Global values, specified in the stream header / extradata or used
132 * all over.
133 * @{
134 */
135 GetBitContext gb; ///< packet bitreader. During decoder init,
136 ///< it contains the extradata from the
137 ///< demuxer. During decoding, it contains
138 ///< packet data.
139 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
140
141 int spillover_bitsize; ///< number of bits used to specify
142 ///< #spillover_nbits in the packet header
143 ///< = ceil(log2(ctx->block_align << 3))
144 int history_nsamples; ///< number of samples in history for signal
145 ///< prediction (through ACB)
146
9a32573b 147 /* postfilter specific values */
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148 int do_apf; ///< whether to apply the averaged
149 ///< projection filter (APF)
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150 int denoise_strength; ///< strength of denoising in Wiener filter
151 ///< [0-11]
152 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
153 ///< Wiener filter coefficients (postfilter)
154 int dc_level; ///< Predicted amount of DC noise, based
155 ///< on which a DC removal filter is used
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156
157 int lsps; ///< number of LSPs per frame [10 or 16]
158 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
159 int lsp_def_mode; ///< defines different sets of LSP defaults
160 ///< [0, 1]
161 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
162 ///< per-frame (independent coding)
163 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
164 ///< per superframe (residual coding)
165
166 int min_pitch_val; ///< base value for pitch parsing code
167 int max_pitch_val; ///< max value + 1 for pitch parsing
168 int pitch_nbits; ///< number of bits used to specify the
169 ///< pitch value in the frame header
170 int block_pitch_nbits; ///< number of bits used to specify the
171 ///< first block's pitch value
172 int block_pitch_range; ///< range of the block pitch
173 int block_delta_pitch_nbits; ///< number of bits used to specify the
174 ///< delta pitch between this and the last
175 ///< block's pitch value, used in all but
176 ///< first block
177 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
178 ///< from -this to +this-1)
179 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
180 ///< conversion
181
182 /**
183 * @}
184 * @defgroup struct_packet Packet values
185 * Packet values, specified in the packet header or related to a packet.
186 * A packet is considered to be a single unit of data provided to this
187 * decoder by the demuxer.
188 * @{
189 */
190 int spillover_nbits; ///< number of bits of the previous packet's
191 ///< last superframe preceeding this
192 ///< packet's first full superframe (useful
193 ///< for re-synchronization also)
194 int has_residual_lsps; ///< if set, superframes contain one set of
195 ///< LSPs that cover all frames, encoded as
196 ///< independent and residual LSPs; if not
197 ///< set, each frame contains its own, fully
198 ///< independent, LSPs
199 int skip_bits_next; ///< number of bits to skip at the next call
200 ///< to #wmavoice_decode_packet() (since
201 ///< they're part of the previous superframe)
202
203 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
204 ///< cache for superframe data split over
205 ///< multiple packets
206 int sframe_cache_size; ///< set to >0 if we have data from an
207 ///< (incomplete) superframe from a previous
208 ///< packet that spilled over in the current
209 ///< packet; specifies the amount of bits in
210 ///< #sframe_cache
211 PutBitContext pb; ///< bitstream writer for #sframe_cache
212
213 /**
214 * @}
215 * @defgroup struct_frame 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 float gain_pred_err[6]; ///< cache for gain prediction
251 float excitation_history[MAX_SIGNAL_HISTORY];
252 ///< cache of the signal of previous
253 ///< superframes, used as a history for
254 ///< signal generation
255 float synth_history[MAX_LSPS]; ///< see #excitation_history
256 /**
257 * @}
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258 * @defgroup post_filter Postfilter values
259 * Varibales used for postfilter implementation, mostly history for
260 * smoothing and so on, and context variables for FFT/iFFT.
261 * @{
262 */
263 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
264 ///< postfilter (for denoise filter)
265 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
266 ///< transform, part of postfilter)
267 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
268 ///< range
269 float postfilter_agc; ///< gain control memory, used in
270 ///< #adaptive_gain_control()
271 float dcf_mem[2]; ///< DC filter history
272 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
273 ///< zero filter output (i.e. excitation)
274 ///< by postfilter
275 float denoise_filter_cache[MAX_FRAMESIZE];
276 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
277 DECLARE_ALIGNED(16, float, tilted_lpcs_pf)[0x80];
278 ///< aligned buffer for LPC tilting
279 DECLARE_ALIGNED(16, float, denoise_coeffs_pf)[0x80];
280 ///< aligned buffer for denoise coefficients
593a8614 281 DECLARE_ALIGNED(16, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
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282 ///< aligned buffer for postfilter speech
283 ///< synthesis
284 /**
285 * @}
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286 */
287} WMAVoiceContext;
288
289/**
290 * Sets up the variable bit mode (VBM) tree from container extradata.
291 * @param gb bit I/O context.
292 * The bit context (s->gb) should be loaded with byte 23-46 of the
293 * container extradata (i.e. the ones containing the VBM tree).
294 * @param vbm_tree pointer to array to which the decoded VBM tree will be
295 * written.
296 * @return 0 on success, <0 on error.
297 */
298static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
299{
300 static const uint8_t bits[] = {
301 2, 2, 2, 4, 4, 4,
302 6, 6, 6, 8, 8, 8,
303 10, 10, 10, 12, 12, 12,
304 14, 14, 14, 14
305 };
306 static const uint16_t codes[] = {
307 0x0000, 0x0001, 0x0002, // 00/01/10
308 0x000c, 0x000d, 0x000e, // 11+00/01/10
309 0x003c, 0x003d, 0x003e, // 1111+00/01/10
310 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
311 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
312 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
313 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
314 };
315 int cntr[8], n, res;
316
317 memset(vbm_tree, 0xff, sizeof(vbm_tree));
318 memset(cntr, 0, sizeof(cntr));
319 for (n = 0; n < 17; n++) {
320 res = get_bits(gb, 3);
321 if (cntr[res] > 3) // should be >= 3 + (res == 7))
322 return -1;
323 vbm_tree[res * 3 + cntr[res]++] = n;
324 }
325 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
326 bits, 1, 1, codes, 2, 2, 132);
327 return 0;
328}
329
330/**
331 * Set up decoder with parameters from demuxer (extradata etc.).
332 */
333static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
334{
335 int n, flags, pitch_range, lsp16_flag;
336 WMAVoiceContext *s = ctx->priv_data;
337
338 /**
339 * Extradata layout:
340 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
341 * - byte 19-22: flags field (annoyingly in LE; see below for known
342 * values),
343 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
344 * rest is 0).
345 */
346 if (ctx->extradata_size != 46) {
347 av_log(ctx, AV_LOG_ERROR,
348 "Invalid extradata size %d (should be 46)\n",
349 ctx->extradata_size);
350 return -1;
351 }
352 flags = AV_RL32(ctx->extradata + 18);
353 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
354 s->do_apf = flags & 0x1;
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355 if (s->do_apf) {
356 ff_rdft_init(&s->rdft, 7, DFT_R2C);
357 ff_rdft_init(&s->irdft, 7, IDFT_C2R);
358 ff_dct_init(&s->dct, 6, DCT_I);
359 ff_dct_init(&s->dst, 6, DST_I);
360
361 ff_sine_window_init(s->cos, 256);
362 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
363 for (n = 0; n < 255; n++) {
364 s->sin[n] = -s->sin[510 - n];
365 s->cos[510 - n] = s->cos[n];
366 }
367 }
368 s->denoise_strength = (flags >> 2) & 0xF;
369 if (s->denoise_strength >= 12) {
370 av_log(ctx, AV_LOG_ERROR,
371 "Invalid denoise filter strength %d (max=11)\n",
372 s->denoise_strength);
373 return -1;
374 }
375 s->denoise_tilt_corr = !!(flags & 0x40);
376 s->dc_level = (flags >> 7) & 0xF;
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377 s->lsp_q_mode = !!(flags & 0x2000);
378 s->lsp_def_mode = !!(flags & 0x4000);
379 lsp16_flag = flags & 0x1000;
380 if (lsp16_flag) {
381 s->lsps = 16;
382 s->frame_lsp_bitsize = 34;
383 s->sframe_lsp_bitsize = 60;
384 } else {
385 s->lsps = 10;
386 s->frame_lsp_bitsize = 24;
387 s->sframe_lsp_bitsize = 48;
388 }
389 for (n = 0; n < s->lsps; n++)
390 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
391
392 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
393 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
394 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
395 return -1;
396 }
397
398 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
399 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
400 pitch_range = s->max_pitch_val - s->min_pitch_val;
401 s->pitch_nbits = av_ceil_log2(pitch_range);
402 s->last_pitch_val = 40;
403 s->last_acb_type = ACB_TYPE_NONE;
404 s->history_nsamples = s->max_pitch_val + 8;
405
406 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
407 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
408 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
409
410 av_log(ctx, AV_LOG_ERROR,
411 "Unsupported samplerate %d (min=%d, max=%d)\n",
412 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
413
414 return -1;
415 }
416
417 s->block_conv_table[0] = s->min_pitch_val;
418 s->block_conv_table[1] = (pitch_range * 25) >> 6;
419 s->block_conv_table[2] = (pitch_range * 44) >> 6;
420 s->block_conv_table[3] = s->max_pitch_val - 1;
421 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
422 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
423 s->block_pitch_range = s->block_conv_table[2] +
424 s->block_conv_table[3] + 1 +
425 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
426 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
427
428 ctx->sample_fmt = SAMPLE_FMT_FLT;
429
430 return 0;
431}
432
433/**
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434 * @defgroup postfilter Postfilter functions
435 * Postfilter functions (gain control, wiener denoise filter, DC filter,
436 * kalman smoothening, plus surrounding code to wrap it)
437 * @{
438 */
439/**
440 * Adaptive gain control (as used in postfilter).
441 *
442 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
443 * that the energy here is calculated using sum(abs(...)), whereas the
444 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
445 *
446 * @param out output buffer for filtered samples
447 * @param in input buffer containing the samples as they are after the
448 * postfilter steps so far
449 * @param speech_synth input buffer containing speech synth before postfilter
450 * @param size input buffer size
451 * @param alpha exponential filter factor
452 * @param gain_mem pointer to filter memory (single float)
453 */
454static void adaptive_gain_control(float *out, const float *in,
455 const float *speech_synth,
456 int size, float alpha, float *gain_mem)
457{
458 int i;
459 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
460 float mem = *gain_mem;
461
462 for (i = 0; i < size; i++) {
463 speech_energy += fabsf(speech_synth[i]);
464 postfilter_energy += fabsf(in[i]);
465 }
466 gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
467
468 for (i = 0; i < size; i++) {
469 mem = alpha * mem + gain_scale_factor;
470 out[i] = in[i] * mem;
471 }
472
473 *gain_mem = mem;
474}
475
476/**
477 * Kalman smoothing function.
478 *
479 * This function looks back pitch +/- 3 samples back into history to find
480 * the best fitting curve (that one giving the optimal gain of the two
481 * signals, i.e. the highest dot product between the two), and then
482 * uses that signal history to smoothen the output of the speech synthesis
483 * filter.
484 *
485 * @param s WMA Voice decoding context
486 * @param pitch pitch of the speech signal
487 * @param in input speech signal
488 * @param out output pointer for smoothened signal
489 * @param size input/output buffer size
490 *
491 * @returns -1 if no smoothening took place, e.g. because no optimal
492 * fit could be found, or 0 on success.
493 */
494static int kalman_smoothen(WMAVoiceContext *s, int pitch,
495 const float *in, float *out, int size)
496{
497 int n;
498 float optimal_gain = 0, dot;
499 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
500 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
501 *best_hist_ptr;
502
503 /* find best fitting point in history */
504 do {
505 dot = ff_dot_productf(in, ptr, size);
506 if (dot > optimal_gain) {
507 optimal_gain = dot;
508 best_hist_ptr = ptr;
509 }
510 } while (--ptr >= end);
511
512 if (optimal_gain <= 0)
513 return -1;
514 dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size);
515 if (dot <= 0) // would be 1.0
516 return -1;
517
518 if (optimal_gain <= dot) {
519 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
520 } else
521 dot = 0.625;
522
523 /* actual smoothing */
524 for (n = 0; n < size; n++)
525 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
526
527 return 0;
528}
529
530/**
531 * Get the tilt factor of a formant filter from its transfer function
532 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
533 * but somehow (??) it does a speech synthesis filter in the
534 * middle, which is missing here
535 *
536 * @param lpcs LPC coefficients
537 * @param n_lpcs Size of LPC buffer
538 * @returns the tilt factor
539 */
540static float tilt_factor(const float *lpcs, int n_lpcs)
541{
542 float rh0, rh1;
543
544 rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs);
545 rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1);
546
547 return rh1 / rh0;
548}
549
550/**
551 * Derive denoise filter coefficients (in real domain) from the LPCs.
552 */
553static void calc_input_response(WMAVoiceContext *s, float *lpcs,
554 int fcb_type, float *coeffs, int remainder)
555{
556 float last_coeff, min = 15.0, max = -15.0;
557 float irange, angle_mul, gain_mul, range, sq;
558 int n, idx;
559
560 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
561 ff_rdft_calc(&s->rdft, lpcs);
562#define log_range(var, assign) do { \
563 float tmp = log10f(assign); var = tmp; \
564 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
565 } while (0)
566 log_range(last_coeff, lpcs[1] * lpcs[1]);
567 for (n = 1; n < 64; n++)
568 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
569 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
570 log_range(lpcs[0], lpcs[0] * lpcs[0]);
571#undef log_range
572 range = max - min;
573 lpcs[64] = last_coeff;
574
575 /* Now, use this spectrum to pick out these frequencies with higher
576 * (relative) power/energy (which we then take to be "not noise"),
577 * and set up a table (still in lpc[]) of (relative) gains per frequency.
578 * These frequencies will be maintained, while others ("noise") will be
579 * decreased in the filter output. */
580 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
581 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
582 (5.0 / 14.7));
583 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
584 for (n = 0; n <= 64; n++) {
585 float pow;
586
587 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
588 pow = wmavoice_denoise_power_table[s->denoise_strength][idx];
589 lpcs[n] = angle_mul * pow;
590
591 /* 70.57 =~ 1/log10(1.0331663) */
592 idx = (pow * gain_mul - 0.0295) * 70.570526123;
593 if (idx > 127) { // fallback if index falls outside table range
594 coeffs[n] = wmavoice_energy_table[127] *
595 powf(1.0331663, idx - 127);
596 } else
597 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
598 }
599
600 /* calculate the Hilbert transform of the gains, which we do (since this
601 * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
602 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
603 * "moment" of the LPCs in this filter. */
604 ff_dct_calc(&s->dct, lpcs);
605 ff_dct_calc(&s->dst, lpcs);
606
607 /* Split out the coefficient indexes into phase/magnitude pairs */
608 idx = 255 + av_clip(lpcs[64], -255, 255);
609 coeffs[0] = coeffs[0] * s->cos[idx];
610 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
611 last_coeff = coeffs[64] * s->cos[idx];
612 for (n = 63;; n--) {
613 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
614 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
615 coeffs[n * 2] = coeffs[n] * s->cos[idx];
616
617 if (!--n) break;
618
619 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
620 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
621 coeffs[n * 2] = coeffs[n] * s->cos[idx];
622 }
623 coeffs[1] = last_coeff;
624
625 /* move into real domain */
626 ff_rdft_calc(&s->irdft, coeffs);
627
628 /* tilt correction and normalize scale */
629 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
630 if (s->denoise_tilt_corr) {
631 float tilt_mem = 0;
632
633 coeffs[remainder - 1] = 0;
634 ff_tilt_compensation(&tilt_mem,
635 -1.8 * tilt_factor(coeffs, remainder - 1),
636 coeffs, remainder);
637 }
638 sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder));
639 for (n = 0; n < remainder; n++)
640 coeffs[n] *= sq;
641}
642
643/**
644 * This function applies a Wiener filter on the (noisy) speech signal as
645 * a means to denoise it.
646 *
647 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
648 * - using this power spectrum, calculate (for each frequency) the Wiener
649 * filter gain, which depends on the frequency power and desired level
650 * of noise subtraction (when set too high, this leads to artifacts)
651 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
652 * of 4-8kHz);
653 * - by doing a phase shift, calculate the Hilbert transform of this array
654 * of per-frequency filter-gains to get the filtering coefficients;
655 * - smoothen/normalize/de-tilt these filter coefficients as desired;
656 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
657 * to get the denoised speech signal;
658 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
659 * the frame boundary) are saved and applied to subsequent frames by an
660 * overlap-add method (otherwise you get clicking-artifacts).
661 *
662 * @param s WMA Voice decoding context
663 * @param s fcb_type Frame (codebook) type
664 * @param synth_pf input: the noisy speech signal, output: denoised speech
665 * data; should be 16-byte aligned (for ASM purposes)
666 * @param size size of the speech data
667 * @param lpcs LPCs used to synthesize this frame's speech data
668 */
669static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
670 float *synth_pf, int size,
671 const float *lpcs)
672{
673 int remainder, lim, n;
674
675 if (fcb_type != FCB_TYPE_SILENCE) {
676 float *tilted_lpcs = s->tilted_lpcs_pf,
677 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
678
679 tilted_lpcs[0] = 1.0;
680 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
681 memset(&tilted_lpcs[s->lsps + 1], 0,
682 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
683 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
684 tilted_lpcs, s->lsps + 2);
685
686 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
687 * size is applied to the next frame. All input beyond this is zero,
688 * and thus all output beyond this will go towards zero, hence we can
689 * limit to min(size-1, 127-size) as a performance consideration. */
690 remainder = FFMIN(127 - size, size - 1);
691 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
692
693 /* apply coefficients (in frequency spectrum domain), i.e. complex
694 * number multiplication */
695 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
696 ff_rdft_calc(&s->rdft, synth_pf);
697 ff_rdft_calc(&s->rdft, coeffs);
698 synth_pf[0] *= coeffs[0];
699 synth_pf[1] *= coeffs[1];
700 for (n = 1; n < 128; n++) {
701 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
702 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
703 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
704 }
705 ff_rdft_calc(&s->irdft, synth_pf);
706 }
707
708 /* merge filter output with the history of previous runs */
709 if (s->denoise_filter_cache_size) {
710 lim = FFMIN(s->denoise_filter_cache_size, size);
711 for (n = 0; n < lim; n++)
712 synth_pf[n] += s->denoise_filter_cache[n];
713 s->denoise_filter_cache_size -= lim;
714 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
715 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
716 }
717
718 /* move remainder of filter output into a cache for future runs */
719 if (fcb_type != FCB_TYPE_SILENCE) {
720 lim = FFMIN(remainder, s->denoise_filter_cache_size);
721 for (n = 0; n < lim; n++)
722 s->denoise_filter_cache[n] += synth_pf[size + n];
723 if (lim < remainder) {
724 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
725 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
726 s->denoise_filter_cache_size = remainder;
727 }
728 }
729}
730
731/**
732 * Averaging projection filter, the postfilter used in WMAVoice.
733 *
734 * This uses the following steps:
735 * - A zero-synthesis filter (generate excitation from synth signal)
736 * - Kalman smoothing on excitation, based on pitch
737 * - Re-synthesized smoothened output
738 * - Iterative Wiener denoise filter
739 * - Adaptive gain filter
740 * - DC filter
741 *
742 * @param s WMAVoice decoding context
743 * @param synth Speech synthesis output (before postfilter)
744 * @param samples Output buffer for filtered samples
745 * @param size Buffer size of synth & samples
746 * @param lpcs Generated LPCs used for speech synthesis
747 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
748 * @param pitch Pitch of the input signal
749 */
750static void postfilter(WMAVoiceContext *s, const float *synth,
751 float *samples, int size,
752 const float *lpcs, float *zero_exc_pf,
753 int fcb_type, int pitch)
754{
755 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
756 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
757 *synth_filter_in = zero_exc_pf;
758
759 assert(size <= MAX_FRAMESIZE / 2);
760
761 /* generate excitation from input signal */
762 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
763
764 if (fcb_type >= FCB_TYPE_AW_PULSES &&
765 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
766 synth_filter_in = synth_filter_in_buf;
767
768 /* re-synthesize speech after smoothening, and keep history */
769 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
770 synth_filter_in, size, s->lsps);
771 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
772 sizeof(synth_pf[0]) * s->lsps);
773
774 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
775
776 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
777 &s->postfilter_agc);
778
779 if (s->dc_level > 8) {
780 /* remove ultra-low frequency DC noise / highpass filter;
781 * coefficients are identical to those used in SIPR decoding,
782 * and very closely resemble those used in AMR-NB decoding. */
783 ff_acelp_apply_order_2_transfer_function(samples, samples,
784 (const float[2]) { -1.99997, 1.0 },
785 (const float[2]) { -1.9330735188, 0.93589198496 },
786 0.93980580475, s->dcf_mem, size);
787 }
788}
789/**
790 * @}
791 */
792
793/**
fa65584f
RB
794 * Dequantize LSPs
795 * @param lsps output pointer to the array that will hold the LSPs
796 * @param num number of LSPs to be dequantized
797 * @param values quantized values, contains n_stages values
798 * @param sizes range (i.e. max value) of each quantized value
799 * @param n_stages number of dequantization runs
800 * @param table dequantization table to be used
801 * @param mul_q LSF multiplier
802 * @param base_q base (lowest) LSF values
803 */
804static void dequant_lsps(double *lsps, int num,
805 const uint16_t *values,
806 const uint16_t *sizes,
807 int n_stages, const uint8_t *table,
808 const double *mul_q,
809 const double *base_q)
810{
811 int n, m;
812
813 memset(lsps, 0, num * sizeof(*lsps));
814 for (n = 0; n < n_stages; n++) {
815 const uint8_t *t_off = &table[values[n] * num];
816 double base = base_q[n], mul = mul_q[n];
817
818 for (m = 0; m < num; m++)
819 lsps[m] += base + mul * t_off[m];
820
821 table += sizes[n] * num;
822 }
823}
824
825/**
826 * @defgroup lsp_dequant LSP dequantization routines
827 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
828 * @note we assume enough bits are available, caller should check.
829 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
830 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
831 * @{
832 */
833/**
834 * Parse 10 independently-coded LSPs.
835 */
836static void dequant_lsp10i(GetBitContext *gb, double *lsps)
837{
838 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
839 static const double mul_lsf[4] = {
840 5.2187144800e-3, 1.4626986422e-3,
841 9.6179549166e-4, 1.1325736225e-3
842 };
843 static const double base_lsf[4] = {
844 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
845 M_PI * -3.3486e-2, M_PI * -5.7408e-2
846 };
847 uint16_t v[4];
848
849 v[0] = get_bits(gb, 8);
850 v[1] = get_bits(gb, 6);
851 v[2] = get_bits(gb, 5);
852 v[3] = get_bits(gb, 5);
853
854 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
855 mul_lsf, base_lsf);
856}
857
858/**
859 * Parse 10 independently-coded LSPs, and then derive the tables to
860 * generate LSPs for the other frames from them (residual coding).
861 */
862static void dequant_lsp10r(GetBitContext *gb,
863 double *i_lsps, const double *old,
864 double *a1, double *a2, int q_mode)
865{
866 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
867 static const double mul_lsf[3] = {
868 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
869 };
870 static const double base_lsf[3] = {
871 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
872 };
873 const float (*ipol_tab)[2][10] = q_mode ?
874 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
875 uint16_t interpol, v[3];
876 int n;
877
878 dequant_lsp10i(gb, i_lsps);
879
880 interpol = get_bits(gb, 5);
881 v[0] = get_bits(gb, 7);
882 v[1] = get_bits(gb, 6);
883 v[2] = get_bits(gb, 6);
884
885 for (n = 0; n < 10; n++) {
886 double delta = old[n] - i_lsps[n];
887 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
888 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
889 }
890
891 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
892 mul_lsf, base_lsf);
893}
894
895/**
896 * Parse 16 independently-coded LSPs.
897 */
898static void dequant_lsp16i(GetBitContext *gb, double *lsps)
899{
900 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
901 static const double mul_lsf[5] = {
902 3.3439586280e-3, 6.9908173703e-4,
903 3.3216608306e-3, 1.0334960326e-3,
904 3.1899104283e-3
905 };
906 static const double base_lsf[5] = {
907 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
908 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
909 M_PI * -1.29816e-1
910 };
911 uint16_t v[5];
912
913 v[0] = get_bits(gb, 8);
914 v[1] = get_bits(gb, 6);
915 v[2] = get_bits(gb, 7);
916 v[3] = get_bits(gb, 6);
917 v[4] = get_bits(gb, 7);
918
919 dequant_lsps( lsps, 5, v, vec_sizes, 2,
920 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
921 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
922 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
923 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
924 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
925}
926
927/**
928 * Parse 16 independently-coded LSPs, and then derive the tables to
929 * generate LSPs for the other frames from them (residual coding).
930 */
931static void dequant_lsp16r(GetBitContext *gb,
932 double *i_lsps, const double *old,
933 double *a1, double *a2, int q_mode)
934{
935 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
936 static const double mul_lsf[3] = {
937 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
938 };
939 static const double base_lsf[3] = {
940 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
941 };
942 const float (*ipol_tab)[2][16] = q_mode ?
943 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
944 uint16_t interpol, v[3];
945 int n;
946
947 dequant_lsp16i(gb, i_lsps);
948
949 interpol = get_bits(gb, 5);
950 v[0] = get_bits(gb, 7);
951 v[1] = get_bits(gb, 7);
952 v[2] = get_bits(gb, 7);
953
954 for (n = 0; n < 16; n++) {
955 double delta = old[n] - i_lsps[n];
956 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
957 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
958 }
959
960 dequant_lsps( a2, 10, v, vec_sizes, 1,
961 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
962 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
963 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
964 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
965 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
966}
967
968/**
969 * @}
970 * @defgroup aw Pitch-adaptive window coding functions
971 * The next few functions are for pitch-adaptive window coding.
972 * @{
973 */
974/**
975 * Parse the offset of the first pitch-adaptive window pulses, and
976 * the distribution of pulses between the two blocks in this frame.
977 * @param s WMA Voice decoding context private data
978 * @param gb bit I/O context
979 * @param pitch pitch for each block in this frame
980 */
981static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
982 const int *pitch)
983{
984 static const int16_t start_offset[94] = {
985 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
986 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
987 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
988 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
989 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
990 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
991 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
992 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
993 };
994 int bits, offset;
995
996 /* position of pulse */
997 s->aw_idx_is_ext = 0;
998 if ((bits = get_bits(gb, 6)) >= 54) {
999 s->aw_idx_is_ext = 1;
1000 bits += (bits - 54) * 3 + get_bits(gb, 2);
1001 }
1002
1003 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1004 * the distribution of the pulses in each block contained in this frame. */
1005 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1006 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1007 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1008 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1009 offset += s->aw_n_pulses[0] * pitch[0];
1010 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1011 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1012
1013 /* if continuing from a position before the block, reset position to
1014 * start of block (when corrected for the range over which it can be
1015 * spread in aw_pulse_set1()). */
1016 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1017 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1018 s->aw_first_pulse_off[1] -= pitch[1];
1019 if (start_offset[bits] < 0)
1020 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1021 s->aw_first_pulse_off[0] -= pitch[0];
1022 }
1023}
1024
1025/**
1026 * Apply second set of pitch-adaptive window pulses.
1027 * @param s WMA Voice decoding context private data
1028 * @param gb bit I/O context
1029 * @param block_idx block index in frame [0, 1]
1030 * @param fcb structure containing fixed codebook vector info
1031 */
1032static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1033 int block_idx, AMRFixed *fcb)
1034{
1035 uint16_t use_mask[7]; // only 5 are used, rest is padding
1036 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1037 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1038 * of idx are the position of the bit within a particular item in the
1039 * array (0 being the most significant bit, and 15 being the least
1040 * significant bit), and the remainder (>> 4) is the index in the
1041 * use_mask[]-array. This is faster and uses less memory than using a
1042 * 80-byte/80-int array. */
1043 int pulse_off = s->aw_first_pulse_off[block_idx],
1044 pulse_start, n, idx, range, aidx, start_off = 0;
1045
1046 /* set offset of first pulse to within this block */
1047 if (s->aw_n_pulses[block_idx] > 0)
1048 while (pulse_off + s->aw_pulse_range < 1)
1049 pulse_off += fcb->pitch_lag;
1050
1051 /* find range per pulse */
1052 if (s->aw_n_pulses[0] > 0) {
1053 if (block_idx == 0) {
1054 range = 32;
1055 } else /* block_idx = 1 */ {
1056 range = 8;
1057 if (s->aw_n_pulses[block_idx] > 0)
1058 pulse_off = s->aw_next_pulse_off_cache;
1059 }
1060 } else
1061 range = 16;
1062 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1063
1064 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1065 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1066 * we exclude that range from being pulsed again in this function. */
1067 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
1068 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1069 if (s->aw_n_pulses[block_idx] > 0)
1070 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1071 int excl_range = s->aw_pulse_range; // always 16 or 24
1072 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1073 int first_sh = 16 - (idx & 15);
1074 *use_mask_ptr++ &= 0xFFFF << first_sh;
1075 excl_range -= first_sh;
1076 if (excl_range >= 16) {
1077 *use_mask_ptr++ = 0;
1078 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1079 } else
1080 *use_mask_ptr &= 0xFFFF >> excl_range;
1081 }
1082
1083 /* find the 'aidx'th offset that is not excluded */
1084 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1085 for (n = 0; n <= aidx; pulse_start++) {
1086 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1087 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1088 if (use_mask[0]) idx = 0x0F;
1089 else if (use_mask[1]) idx = 0x1F;
1090 else if (use_mask[2]) idx = 0x2F;
1091 else if (use_mask[3]) idx = 0x3F;
1092 else if (use_mask[4]) idx = 0x4F;
1093 else return;
1094 idx -= av_log2_16bit(use_mask[idx >> 4]);
1095 }
1096 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1097 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1098 n++;
1099 start_off = idx;
1100 }
1101 }
1102
1103 fcb->x[fcb->n] = start_off;
1104 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1105 fcb->n++;
1106
1107 /* set offset for next block, relative to start of that block */
1108 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1109 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1110}
1111
1112/**
1113 * Apply first set of pitch-adaptive window pulses.
1114 * @param s WMA Voice decoding context private data
1115 * @param gb bit I/O context
1116 * @param block_idx block index in frame [0, 1]
1117 * @param fcb storage location for fixed codebook pulse info
1118 */
1119static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1120 int block_idx, AMRFixed *fcb)
1121{
1122 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1123 float v;
1124
1125 if (s->aw_n_pulses[block_idx] > 0) {
1126 int n, v_mask, i_mask, sh, n_pulses;
1127
1128 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1129 n_pulses = 3;
1130 v_mask = 8;
1131 i_mask = 7;
1132 sh = 4;
1133 } else { // 4 pulses, 1:sign + 2:index each
1134 n_pulses = 4;
1135 v_mask = 4;
1136 i_mask = 3;
1137 sh = 3;
1138 }
1139
1140 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1141 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1142 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1143 s->aw_first_pulse_off[block_idx];
1144 while (fcb->x[fcb->n] < 0)
1145 fcb->x[fcb->n] += fcb->pitch_lag;
1146 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1147 fcb->n++;
1148 }
1149 } else {
1150 int num2 = (val & 0x1FF) >> 1, delta, idx;
1151
1152 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1153 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1154 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1155 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1156 v = (val & 0x200) ? -1.0 : 1.0;
1157
1158 fcb->no_repeat_mask |= 3 << fcb->n;
1159 fcb->x[fcb->n] = idx - delta;
1160 fcb->y[fcb->n] = v;
1161 fcb->x[fcb->n + 1] = idx;
1162 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1163 fcb->n += 2;
1164 }
1165}
1166
1167/**
1168 * @}
1169 *
1170 * Generate a random number from frame_cntr and block_idx, which will lief
1171 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1172 * table of size 1000 of which you want to read block_size entries).
1173 *
1174 * @param frame_cntr current frame number
1175 * @param block_num current block index
1176 * @param block_size amount of entries we want to read from a table
1177 * that has 1000 entries
32e543f8 1178 * @return a (non-)random number in the [0, 1000 - block_size] range.
fa65584f
RB
1179 */
1180static int pRNG(int frame_cntr, int block_num, int block_size)
1181{
1182 /* array to simplify the calculation of z:
1183 * y = (x % 9) * 5 + 6;
1184 * z = (49995 * x) / y;
1185 * Since y only has 9 values, we can remove the division by using a
1186 * LUT and using FASTDIV-style divisions. For each of the 9 values
1187 * of y, we can rewrite z as:
1188 * z = x * (49995 / y) + x * ((49995 % y) / y)
1189 * In this table, each col represents one possible value of y, the
1190 * first number is 49995 / y, and the second is the FASTDIV variant
1191 * of 49995 % y / y. */
1192 static const unsigned int div_tbl[9][2] = {
1193 { 8332, 3 * 715827883U }, // y = 6
1194 { 4545, 0 * 390451573U }, // y = 11
1195 { 3124, 11 * 268435456U }, // y = 16
1196 { 2380, 15 * 204522253U }, // y = 21
1197 { 1922, 23 * 165191050U }, // y = 26
1198 { 1612, 23 * 138547333U }, // y = 31
1199 { 1388, 27 * 119304648U }, // y = 36
1200 { 1219, 16 * 104755300U }, // y = 41
1201 { 1086, 39 * 93368855U } // y = 46
1202 };
1203 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1204 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1205 // so this is effectively a modulo (%)
1206 y = x - 9 * MULH(477218589, x); // x % 9
1207 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1208 // z = x * 49995 / (y * 5 + 6)
1209 return z % (1000 - block_size);
1210}
1211
1212/**
1213 * Parse hardcoded signal for a single block.
1214 * @note see #synth_block().
1215 */
1216static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1217 int block_idx, int size,
1218 const struct frame_type_desc *frame_desc,
1219 float *excitation)
1220{
1221 float gain;
1222 int n, r_idx;
1223
1224 assert(size <= MAX_FRAMESIZE);
1225
1226 /* Set the offset from which we start reading wmavoice_std_codebook */
1227 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1228 r_idx = pRNG(s->frame_cntr, block_idx, size);
1229 gain = s->silence_gain;
1230 } else /* FCB_TYPE_HARDCODED */ {
1231 r_idx = get_bits(gb, 8);
1232 gain = wmavoice_gain_universal[get_bits(gb, 6)];
1233 }
1234
1235 /* Clear gain prediction parameters */
1236 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1237
1238 /* Apply gain to hardcoded codebook and use that as excitation signal */
1239 for (n = 0; n < size; n++)
1240 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1241}
1242
1243/**
1244 * Parse FCB/ACB signal for a single block.
1245 * @note see #synth_block().
1246 */
1247static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1248 int block_idx, int size,
1249 int block_pitch_sh2,
1250 const struct frame_type_desc *frame_desc,
1251 float *excitation)
1252{
1253 static const float gain_coeff[6] = {
1254 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1255 };
1256 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1257 int n, idx, gain_weight;
1258 AMRFixed fcb;
1259
1260 assert(size <= MAX_FRAMESIZE / 2);
1261 memset(pulses, 0, sizeof(*pulses) * size);
1262
1263 fcb.pitch_lag = block_pitch_sh2 >> 2;
1264 fcb.pitch_fac = 1.0;
1265 fcb.no_repeat_mask = 0;
1266 fcb.n = 0;
1267
1268 /* For the other frame types, this is where we apply the innovation
1269 * (fixed) codebook pulses of the speech signal. */
1270 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1271 aw_pulse_set1(s, gb, block_idx, &fcb);
1272 aw_pulse_set2(s, gb, block_idx, &fcb);
1273 } else /* FCB_TYPE_EXC_PULSES */ {
1274 int offset_nbits = 5 - frame_desc->log_n_blocks;
1275
1276 fcb.no_repeat_mask = -1;
1277 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1278 * (instead of double) for a subset of pulses */
1279 for (n = 0; n < 5; n++) {
1280 float sign;
1281 int pos1, pos2;
1282
1283 sign = get_bits1(gb) ? 1.0 : -1.0;
1284 pos1 = get_bits(gb, offset_nbits);
1285 fcb.x[fcb.n] = n + 5 * pos1;
1286 fcb.y[fcb.n++] = sign;
1287 if (n < frame_desc->dbl_pulses) {
1288 pos2 = get_bits(gb, offset_nbits);
1289 fcb.x[fcb.n] = n + 5 * pos2;
1290 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1291 }
1292 }
1293 }
1294 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1295
1296 /* Calculate gain for adaptive & fixed codebook signal.
1297 * see ff_amr_set_fixed_gain(). */
1298 idx = get_bits(gb, 7);
1299 fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) -
1300 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1301 acb_gain = wmavoice_gain_codebook_acb[idx];
1302 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1303 -2.9957322736 /* log(0.05) */,
1304 1.6094379124 /* log(5.0) */);
1305
1306 gain_weight = 8 >> frame_desc->log_n_blocks;
1307 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1308 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1309 for (n = 0; n < gain_weight; n++)
1310 s->gain_pred_err[n] = pred_err;
1311
1312 /* Calculation of adaptive codebook */
1313 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1314 int len;
1315 for (n = 0; n < size; n += len) {
1316 int next_idx_sh16;
1317 int abs_idx = block_idx * size + n;
1318 int pitch_sh16 = (s->last_pitch_val << 16) +
1319 s->pitch_diff_sh16 * abs_idx;
1320 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1321 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1322 idx = idx_sh16 >> 16;
1323 if (s->pitch_diff_sh16) {
1324 if (s->pitch_diff_sh16 > 0) {
1325 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1326 } else
1327 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1328 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1329 1, size - n);
1330 } else
1331 len = size;
1332
1333 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1334 wmavoice_ipol1_coeffs, 17,
1335 idx, 9, len);
1336 }
1337 } else /* ACB_TYPE_HAMMING */ {
1338 int block_pitch = block_pitch_sh2 >> 2;
1339 idx = block_pitch_sh2 & 3;
1340 if (idx) {
1341 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1342 wmavoice_ipol2_coeffs, 4,
1343 idx, 8, size);
1344 } else
1345 av_memcpy_backptr(excitation, sizeof(float) * block_pitch,
1346 sizeof(float) * size);
1347 }
1348
1349 /* Interpolate ACB/FCB and use as excitation signal */
1350 ff_weighted_vector_sumf(excitation, excitation, pulses,
1351 acb_gain, fcb_gain, size);
1352}
1353
1354/**
1355 * Parse data in a single block.
1356 * @note we assume enough bits are available, caller should check.
1357 *
1358 * @param s WMA Voice decoding context private data
1359 * @param gb bit I/O context
1360 * @param block_idx index of the to-be-read block
1361 * @param size amount of samples to be read in this block
1362 * @param block_pitch_sh2 pitch for this block << 2
1363 * @param lsps LSPs for (the end of) this frame
1364 * @param prev_lsps LSPs for the last frame
1365 * @param frame_desc frame type descriptor
1366 * @param excitation target memory for the ACB+FCB interpolated signal
1367 * @param synth target memory for the speech synthesis filter output
1368 * @return 0 on success, <0 on error.
1369 */
1370static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1371 int block_idx, int size,
1372 int block_pitch_sh2,
1373 const double *lsps, const double *prev_lsps,
1374 const struct frame_type_desc *frame_desc,
1375 float *excitation, float *synth)
1376{
1377 double i_lsps[MAX_LSPS];
1378 float lpcs[MAX_LSPS];
1379 float fac;
1380 int n;
1381
1382 if (frame_desc->acb_type == ACB_TYPE_NONE)
1383 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1384 else
1385 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1386 frame_desc, excitation);
1387
1388 /* convert interpolated LSPs to LPCs */
1389 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1390 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1391 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1392 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1393
1394 /* Speech synthesis */
1395 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1396}
1397
1398/**
1399 * Synthesize output samples for a single frame.
1400 * @note we assume enough bits are available, caller should check.
1401 *
1402 * @param ctx WMA Voice decoder context
1403 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
9a32573b 1404 * @param frame_idx Frame number within superframe [0-2]
fa65584f
RB
1405 * @param samples pointer to output sample buffer, has space for at least 160
1406 * samples
1407 * @param lsps LSP array
1408 * @param prev_lsps array of previous frame's LSPs
1409 * @param excitation target buffer for excitation signal
1410 * @param synth target buffer for synthesized speech data
1411 * @return 0 on success, <0 on error.
1412 */
9a32573b 1413static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
fa65584f
RB
1414 float *samples,
1415 const double *lsps, const double *prev_lsps,
1416 float *excitation, float *synth)
1417{
1418 WMAVoiceContext *s = ctx->priv_data;
1419 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
1420 int pitch[MAX_BLOCKS], last_block_pitch;
1421
1422 /* Parse frame type ("frame header"), see frame_descs */
1423 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)],
1424 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1425
1426 if (bd_idx < 0) {
1427 av_log(ctx, AV_LOG_ERROR,
1428 "Invalid frame type VLC code, skipping\n");
1429 return -1;
1430 }
1431
1432 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1433 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1434 /* Pitch is provided per frame, which is interpreted as the pitch of
1435 * the last sample of the last block of this frame. We can interpolate
1436 * the pitch of other blocks (and even pitch-per-sample) by gradually
1437 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1438 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1439 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1440 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1441 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1442 if (s->last_acb_type == ACB_TYPE_NONE ||
1443 20 * abs(cur_pitch_val - s->last_pitch_val) >
1444 (cur_pitch_val + s->last_pitch_val))
1445 s->last_pitch_val = cur_pitch_val;
1446
1447 /* pitch per block */
1448 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1449 int fac = n * 2 + 1;
1450
1451 pitch[n] = (MUL16(fac, cur_pitch_val) +
1452 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1453 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1454 }
1455
1456 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1457 s->pitch_diff_sh16 =
1458 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1459 }
1460
1461 /* Global gain (if silence) and pitch-adaptive window coordinates */
1462 switch (frame_descs[bd_idx].fcb_type) {
1463 case FCB_TYPE_SILENCE:
1464 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1465 break;
1466 case FCB_TYPE_AW_PULSES:
1467 aw_parse_coords(s, gb, pitch);
1468 break;
1469 }
1470
1471 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1472 int bl_pitch_sh2;
1473
1474 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1475 switch (frame_descs[bd_idx].acb_type) {
1476 case ACB_TYPE_HAMMING: {
1477 /* Pitch is given per block. Per-block pitches are encoded as an
1478 * absolute value for the first block, and then delta values
1479 * relative to this value) for all subsequent blocks. The scale of
1480 * this pitch value is semi-logaritmic compared to its use in the
1481 * decoder, so we convert it to normal scale also. */
1482 int block_pitch,
1483 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1484 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1485 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1486
1487 if (n == 0) {
1488 block_pitch = get_bits(gb, s->block_pitch_nbits);
1489 } else
1490 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1491 get_bits(gb, s->block_delta_pitch_nbits);
1492 /* Convert last_ so that any next delta is within _range */
1493 last_block_pitch = av_clip(block_pitch,
1494 s->block_delta_pitch_hrange,
1495 s->block_pitch_range -
1496 s->block_delta_pitch_hrange);
1497
1498 /* Convert semi-log-style scale back to normal scale */
1499 if (block_pitch < t1) {
1500 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1501 } else {
1502 block_pitch -= t1;
1503 if (block_pitch < t2) {
1504 bl_pitch_sh2 =
1505 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1506 } else {
1507 block_pitch -= t2;
1508 if (block_pitch < t3) {
1509 bl_pitch_sh2 =
1510 (s->block_conv_table[2] + block_pitch) << 2;
1511 } else
1512 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1513 }
1514 }
1515 pitch[n] = bl_pitch_sh2 >> 2;
1516 break;
1517 }
1518
1519 case ACB_TYPE_ASYMMETRIC: {
1520 bl_pitch_sh2 = pitch[n] << 2;
1521 break;
1522 }
1523
1524 default: // ACB_TYPE_NONE has no pitch
1525 bl_pitch_sh2 = 0;
1526 break;
1527 }
1528
1529 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1530 lsps, prev_lsps, &frame_descs[bd_idx],
1531 &excitation[n * block_nsamples],
1532 &synth[n * block_nsamples]);
1533 }
1534
1535 /* Averaging projection filter, if applicable. Else, just copy samples
1536 * from synthesis buffer */
1537 if (s->do_apf) {
9a32573b
RB
1538 double i_lsps[MAX_LSPS];
1539 float lpcs[MAX_LSPS];
1540
1541 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1542 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1543 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1544 postfilter(s, synth, samples, 80, lpcs,
1545 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1546 frame_descs[bd_idx].fcb_type, pitch[0]);
1547
1548 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1549 i_lsps[n] = cos(lsps[n]);
1550 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1551 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1552 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1553 frame_descs[bd_idx].fcb_type, pitch[0]);
1554 } else
b1078e9f 1555 memcpy(samples, synth, 160 * sizeof(synth[0]));
fa65584f
RB
1556
1557 /* Cache values for next frame */
1558 s->frame_cntr++;
1559 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1560 s->last_acb_type = frame_descs[bd_idx].acb_type;
1561 switch (frame_descs[bd_idx].acb_type) {
1562 case ACB_TYPE_NONE:
1563 s->last_pitch_val = 0;
1564 break;
1565 case ACB_TYPE_ASYMMETRIC:
1566 s->last_pitch_val = cur_pitch_val;
1567 break;
1568 case ACB_TYPE_HAMMING:
1569 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1570 break;
1571 }
1572
1573 return 0;
1574}
1575
1576/**
1577 * Ensure minimum value for first item, maximum value for last value,
1578 * proper spacing between each value and proper ordering.
1579 *
1580 * @param lsps array of LSPs
1581 * @param num size of LSP array
1582 *
1583 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1584 * useful to put in a generic location later on. Parts are also
1585 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1586 * which is in float.
1587 */
1588static void stabilize_lsps(double *lsps, int num)
1589{
1590 int n, m, l;
1591
1592 /* set minimum value for first, maximum value for last and minimum
1593 * spacing between LSF values.
1594 * Very similar to ff_set_min_dist_lsf(), but in double. */
1595 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1596 for (n = 1; n < num; n++)
1597 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1598 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1599
1600 /* reorder (looks like one-time / non-recursed bubblesort).
1601 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1602 for (n = 1; n < num; n++) {
1603 if (lsps[n] < lsps[n - 1]) {
1604 for (m = 1; m < num; m++) {
1605 double tmp = lsps[m];
1606 for (l = m - 1; l >= 0; l--) {
1607 if (lsps[l] <= tmp) break;
1608 lsps[l + 1] = lsps[l];
1609 }
1610 lsps[l + 1] = tmp;
1611 }
1612 break;
1613 }
1614 }
1615}
1616
1617/**
1618 * Test if there's enough bits to read 1 superframe.
1619 *
1620 * @param orig_gb bit I/O context used for reading. This function
1621 * does not modify the state of the bitreader; it
1622 * only uses it to copy the current stream position
1623 * @param s WMA Voice decoding context private data
32e543f8 1624 * @return -1 if unsupported, 1 on not enough bits or 0 if OK.
fa65584f
RB
1625 */
1626static int check_bits_for_superframe(GetBitContext *orig_gb,
1627 WMAVoiceContext *s)
1628{
1629 GetBitContext s_gb, *gb = &s_gb;
1630 int n, need_bits, bd_idx;
1631 const struct frame_type_desc *frame_desc;
1632
1633 /* initialize a copy */
1634 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
1635 skip_bits_long(gb, get_bits_count(orig_gb));
1636 assert(get_bits_left(gb) == get_bits_left(orig_gb));
1637
1638 /* superframe header */
1639 if (get_bits_left(gb) < 14)
1640 return 1;
1641 if (!get_bits1(gb))
1642 return -1; // WMAPro-in-WMAVoice superframe
1643 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe
1644 if (s->has_residual_lsps) { // residual LSPs (for all frames)
1645 if (get_bits_left(gb) < s->sframe_lsp_bitsize)
1646 return 1;
1647 skip_bits_long(gb, s->sframe_lsp_bitsize);
1648 }
1649
1650 /* frames */
1651 for (n = 0; n < MAX_FRAMES; n++) {
1652 int aw_idx_is_ext = 0;
1653
1654 if (!s->has_residual_lsps) { // independent LSPs (per-frame)
1655 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
1656 skip_bits_long(gb, s->frame_lsp_bitsize);
1657 }
1658 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
1659 if (bd_idx < 0)
1660 return -1; // invalid frame type VLC code
1661 frame_desc = &frame_descs[bd_idx];
1662 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1663 if (get_bits_left(gb) < s->pitch_nbits)
1664 return 1;
1665 skip_bits_long(gb, s->pitch_nbits);
1666 }
1667 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1668 skip_bits(gb, 8);
1669 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1670 int tmp = get_bits(gb, 6);
1671 if (tmp >= 0x36) {
1672 skip_bits(gb, 2);
1673 aw_idx_is_ext = 1;
1674 }
1675 }
1676
1677 /* blocks */
1678 if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
1679 need_bits = s->block_pitch_nbits +
1680 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
1681 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1682 need_bits = 2 * !aw_idx_is_ext;
1683 } else
1684 need_bits = 0;
1685 need_bits += frame_desc->frame_size;
1686 if (get_bits_left(gb) < need_bits)
1687 return 1;
1688 skip_bits_long(gb, need_bits);
1689 }
1690
1691 return 0;
1692}
1693
1694/**
1695 * Synthesize output samples for a single superframe. If we have any data
1696 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1697 * in s->gb.
1698 *
1699 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1700 * to give a total of 480 samples per frame. See #synth_frame() for frame
1701 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1702 * (if these are globally specified for all frames (residually); they can
1703 * also be specified individually per-frame. See the s->has_residual_lsps
1704 * option), and can specify the number of samples encoded in this superframe
1705 * (if less than 480), usually used to prevent blanks at track boundaries.
1706 *
1707 * @param ctx WMA Voice decoder context
1708 * @param samples pointer to output buffer for voice samples
1709 * @param data_size pointer containing the size of #samples on input, and the
1710 * amount of #samples filled on output
1711 * @return 0 on success, <0 on error or 1 if there was not enough data to
1712 * fully parse the superframe
1713 */
1714static int synth_superframe(AVCodecContext *ctx,
1715 float *samples, int *data_size)
1716{
1717 WMAVoiceContext *s = ctx->priv_data;
1718 GetBitContext *gb = &s->gb, s_gb;
1719 int n, res, n_samples = 480;
1720 double lsps[MAX_FRAMES][MAX_LSPS];
1721 const double *mean_lsf = s->lsps == 16 ?
1722 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1723 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1724 float synth[MAX_LSPS + MAX_SFRAMESIZE];
1725
1726 memcpy(synth, s->synth_history,
1727 s->lsps * sizeof(*synth));
1728 memcpy(excitation, s->excitation_history,
1729 s->history_nsamples * sizeof(*excitation));
1730
1731 if (s->sframe_cache_size > 0) {
1732 gb = &s_gb;
1733 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1734 s->sframe_cache_size = 0;
1735 }
1736
1737 if ((res = check_bits_for_superframe(gb, s)) == 1) return 1;
1738
1739 /* First bit is speech/music bit, it differentiates between WMAVoice
1740 * speech samples (the actual codec) and WMAVoice music samples, which
1741 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1742 * the wild yet. */
1743 if (!get_bits1(gb)) {
1744 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1);
1745 return -1;
1746 }
1747
1748 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1749 if (get_bits1(gb)) {
1750 if ((n_samples = get_bits(gb, 12)) > 480) {
1751 av_log(ctx, AV_LOG_ERROR,
1752 "Superframe encodes >480 samples (%d), not allowed\n",
1753 n_samples);
1754 return -1;
1755 }
1756 }
1757 /* Parse LSPs, if global for the superframe (can also be per-frame). */
1758 if (s->has_residual_lsps) {
1759 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1760
1761 for (n = 0; n < s->lsps; n++)
1762 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1763
1764 if (s->lsps == 10) {
1765 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1766 } else /* s->lsps == 16 */
1767 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1768
1769 for (n = 0; n < s->lsps; n++) {
1770 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
1771 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1772 lsps[2][n] += mean_lsf[n];
1773 }
1774 for (n = 0; n < 3; n++)
1775 stabilize_lsps(lsps[n], s->lsps);
1776 }
1777
1778 /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */
1779 for (n = 0; n < 3; n++) {
1780 if (!s->has_residual_lsps) {
1781 int m;
1782
1783 if (s->lsps == 10) {
1784 dequant_lsp10i(gb, lsps[n]);
1785 } else /* s->lsps == 16 */
1786 dequant_lsp16i(gb, lsps[n]);
1787
1788 for (m = 0; m < s->lsps; m++)
1789 lsps[n][m] += mean_lsf[m];
1790 stabilize_lsps(lsps[n], s->lsps);
1791 }
1792
9a32573b 1793 if ((res = synth_frame(ctx, gb, n,
fa65584f
RB
1794 &samples[n * MAX_FRAMESIZE],
1795 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1796 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1797 &synth[s->lsps + n * MAX_FRAMESIZE])))
1798 return res;
1799 }
1800
1801 /* Statistics? FIXME - we don't check for length, a slight overrun
1802 * will be caught by internal buffer padding, and anything else
1803 * will be skipped, not read. */
1804 if (get_bits1(gb)) {
1805 res = get_bits(gb, 4);
1806 skip_bits(gb, 10 * (res + 1));
1807 }
1808
1809 /* Specify nr. of output samples */
1810 *data_size = n_samples * sizeof(float);
1811
1812 /* Update history */
1813 memcpy(s->prev_lsps, lsps[2],
1814 s->lsps * sizeof(*s->prev_lsps));
1815 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1816 s->lsps * sizeof(*synth));
1817 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1818 s->history_nsamples * sizeof(*excitation));
9a32573b
RB
1819 if (s->do_apf)
1820 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1821 s->history_nsamples * sizeof(*s->zero_exc_pf));
fa65584f
RB
1822
1823 return 0;
1824}
1825
1826/**
1827 * Parse the packet header at the start of each packet (input data to this
1828 * decoder).
1829 *
1830 * @param s WMA Voice decoding context private data
32e543f8 1831 * @return 1 if not enough bits were available, or 0 on success.
fa65584f
RB
1832 */
1833static int parse_packet_header(WMAVoiceContext *s)
1834{
1835 GetBitContext *gb = &s->gb;
1836 unsigned int res;
1837
1838 if (get_bits_left(gb) < 11)
1839 return 1;
1840 skip_bits(gb, 4); // packet sequence number
1841 s->has_residual_lsps = get_bits1(gb);
1842 do {
1843 res = get_bits(gb, 6); // number of superframes per packet
1844 // (minus first one if there is spillover)
1845 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
1846 return 1;
1847 } while (res == 0x3F);
1848 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
1849
1850 return 0;
1851}
1852
1853/**
1854 * Copy (unaligned) bits from gb/data/size to pb.
1855 *
1856 * @param pb target buffer to copy bits into
1857 * @param data source buffer to copy bits from
1858 * @param size size of the source data, in bytes
1859 * @param gb bit I/O context specifying the current position in the source.
1860 * data. This function might use this to align the bit position to
1861 * a whole-byte boundary before calling #ff_copy_bits() on aligned
1862 * source data
1863 * @param nbits the amount of bits to copy from source to target
1864 *
1865 * @note after calling this function, the current position in the input bit
1866 * I/O context is undefined.
1867 */
1868static void copy_bits(PutBitContext *pb,
1869 const uint8_t *data, int size,
1870 GetBitContext *gb, int nbits)
1871{
1872 int rmn_bytes, rmn_bits;
1873
1874 rmn_bits = rmn_bytes = get_bits_left(gb);
1875 if (rmn_bits < nbits)
1876 return;
1877 rmn_bits &= 7; rmn_bytes >>= 3;
1878 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1879 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1880 ff_copy_bits(pb, data + size - rmn_bytes,
1881 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1882}
1883
1884/**
1885 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1886 * and we expect that the demuxer / application provides it to us as such
1887 * (else you'll probably get garbage as output). Every packet has a size of
1888 * ctx->block_align bytes, starts with a packet header (see
1889 * #parse_packet_header()), and then a series of superframes. Superframe
1890 * boundaries may exceed packets, i.e. superframes can split data over
1891 * multiple (two) packets.
1892 *
1893 * For more information about frames, see #synth_superframe().
1894 */
1895static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1896 int *data_size, AVPacket *avpkt)
1897{
1898 WMAVoiceContext *s = ctx->priv_data;
1899 GetBitContext *gb = &s->gb;
1900 int size, res, pos;
1901
1902 if (*data_size < 480 * sizeof(float)) {
1903 av_log(ctx, AV_LOG_ERROR,
1904 "Output buffer too small (%d given - %lu needed)\n",
1905 *data_size, 480 * sizeof(float));
1906 return -1;
1907 }
1908 *data_size = 0;
1909
1910 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1911 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1912 * feeds us ASF packets, which may concatenate multiple "codec" packets
1913 * in a single "muxer" packet, so we artificially emulate that by
1914 * capping the packet size at ctx->block_align. */
1915 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1916 if (!size)
1917 return 0;
1918 init_get_bits(&s->gb, avpkt->data, size << 3);
1919
1920 /* size == ctx->block_align is used to indicate whether we are dealing with
1921 * a new packet or a packet of which we already read the packet header
1922 * previously. */
1923 if (size == ctx->block_align) { // new packet header
1924 if ((res = parse_packet_header(s)) < 0)
1925 return res;
1926
1927 /* If the packet header specifies a s->spillover_nbits, then we want
1928 * to push out all data of the previous packet (+ spillover) before
1929 * continuing to parse new superframes in the current packet. */
1930 if (s->spillover_nbits > 0) {
1931 if (s->sframe_cache_size > 0) {
1932 int cnt = get_bits_count(gb);
1933 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1934 flush_put_bits(&s->pb);
1935 s->sframe_cache_size += s->spillover_nbits;
1936 if ((res = synth_superframe(ctx, data, data_size)) == 0 &&
1937 *data_size > 0) {
1938 cnt += s->spillover_nbits;
1939 s->skip_bits_next = cnt & 7;
1940 return cnt >> 3;
1941 } else
1942 skip_bits_long (gb, s->spillover_nbits - cnt +
1943 get_bits_count(gb)); // resync
1944 } else
1945 skip_bits_long(gb, s->spillover_nbits); // resync
1946 }
1947 } else if (s->skip_bits_next)
1948 skip_bits(gb, s->skip_bits_next);
1949
1950 /* Try parsing superframes in current packet */
1951 s->sframe_cache_size = 0;
1952 s->skip_bits_next = 0;
1953 pos = get_bits_left(gb);
1954 if ((res = synth_superframe(ctx, data, data_size)) < 0) {
1955 return res;
1956 } else if (*data_size > 0) {
1957 int cnt = get_bits_count(gb);
1958 s->skip_bits_next = cnt & 7;
1959 return cnt >> 3;
1960 } else if ((s->sframe_cache_size = pos) > 0) {
1961 /* rewind bit reader to start of last (incomplete) superframe... */
1962 init_get_bits(gb, avpkt->data, size << 3);
1963 skip_bits_long(gb, (size << 3) - pos);
1964 assert(get_bits_left(gb) == pos);
1965
1966 /* ...and cache it for spillover in next packet */
1967 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
1968 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
1969 // FIXME bad - just copy bytes as whole and add use the
1970 // skip_bits_next field
1971 }
1972
1973 return size;
1974}
1975
9a32573b
RB
1976static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
1977{
1978 WMAVoiceContext *s = ctx->priv_data;
1979
1980 if (s->do_apf) {
1981 ff_rdft_end(&s->rdft);
1982 ff_rdft_end(&s->irdft);
1983 ff_dct_end(&s->dct);
1984 ff_dct_end(&s->dst);
1985 }
1986
1987 return 0;
1988}
1989
fa65584f
RB
1990static av_cold void wmavoice_flush(AVCodecContext *ctx)
1991{
1992 WMAVoiceContext *s = ctx->priv_data;
1993 int n;
1994
9a32573b 1995 s->postfilter_agc = 0;
fa65584f
RB
1996 s->sframe_cache_size = 0;
1997 s->skip_bits_next = 0;
1998 for (n = 0; n < s->lsps; n++)
1999 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2000 memset(s->excitation_history, 0,
2001 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2002 memset(s->synth_history, 0,
2003 sizeof(*s->synth_history) * MAX_LSPS);
2004 memset(s->gain_pred_err, 0,
2005 sizeof(s->gain_pred_err));
9a32573b
RB
2006
2007 if (s->do_apf) {
2008 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2009 sizeof(*s->synth_filter_out_buf) * s->lsps);
2010 memset(s->dcf_mem, 0,
2011 sizeof(*s->dcf_mem) * 2);
2012 memset(s->zero_exc_pf, 0,
2013 sizeof(*s->zero_exc_pf) * s->history_nsamples);
2014 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2015 }
fa65584f
RB
2016}
2017
2018AVCodec wmavoice_decoder = {
2019 "wmavoice",
72415b2a 2020 AVMEDIA_TYPE_AUDIO,
fa65584f
RB
2021 CODEC_ID_WMAVOICE,
2022 sizeof(WMAVoiceContext),
2023 wmavoice_decode_init,
2024 NULL,
9a32573b 2025 wmavoice_decode_end,
fa65584f
RB
2026 wmavoice_decode_packet,
2027 CODEC_CAP_SUBFRAMES,
2028 .flush = wmavoice_flush,
2029 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
2030};