2 * Windows Media Audio Voice decoder.
3 * Copyright (c) 2009 Ronald S. Bultje
5 * This file is part of FFmpeg.
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.
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.
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
24 * @brief Windows Media Audio Voice compatible decoder
25 * @author Ronald S. Bultje <rsbultje@gmail.com>
30 #include "libavutil/channel_layout.h"
31 #include "libavutil/float_dsp.h"
32 #include "libavutil/mem.h"
37 #include "wmavoice_data.h"
38 #include "celp_filters.h"
39 #include "acelp_vectors.h"
40 #include "acelp_filters.h"
46 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame
47 #define MAX_LSPS 16 ///< maximum filter order
48 #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple
49 ///< of 16 for ASM input buffer alignment
50 #define MAX_FRAMES 3 ///< maximum number of frames per superframe
51 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame
52 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history
53 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
54 ///< maximum number of samples per superframe
55 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
56 ///< was split over two packets
57 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration
60 * Frame type VLC coding.
62 static VLC frame_type_vlc;
65 * Adaptive codebook types.
68 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed)
69 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
70 ///< we interpolate to get a per-sample pitch.
71 ///< Signal is generated using an asymmetric sinc
73 ///< @note see #wmavoice_ipol1_coeffs
74 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using
75 ///< a Hamming sinc window function
76 ///< @note see #wmavoice_ipol2_coeffs
80 * Fixed codebook types.
83 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence
84 ///< generated from a hardcoded (fixed) codebook
85 ///< with per-frame (low) gain values
86 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block
88 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals,
89 ///< used in particular for low-bitrate streams
90 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
91 ///< combinations of either single pulses or
96 * Description of frame types.
98 static const struct frame_type_desc {
99 uint8_t n_blocks; ///< amount of blocks per frame (each block
100 ///< (contains 160/#n_blocks samples)
101 uint8_t log_n_blocks; ///< log2(#n_blocks)
102 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*)
103 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*)
104 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs
105 ///< (rather than just one single pulse)
106 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
107 uint16_t frame_size; ///< the amount of bits that make up the block
108 ///< data (per frame)
109 } frame_descs[17] = {
110 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 },
111 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 },
112 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 },
113 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 },
114 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
115 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
116 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
117 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
118 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 },
119 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 },
120 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 },
121 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 },
122 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 },
123 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 },
124 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 },
125 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 },
126 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 }
130 * WMA Voice decoding context.
134 * @name Global values specified in the stream header / extradata or used all over.
138 GetBitContext gb; ///< packet bitreader. During decoder init,
139 ///< it contains the extradata from the
140 ///< demuxer. During decoding, it contains
142 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
144 int spillover_bitsize; ///< number of bits used to specify
145 ///< #spillover_nbits in the packet header
146 ///< = ceil(log2(ctx->block_align << 3))
147 int history_nsamples; ///< number of samples in history for signal
148 ///< prediction (through ACB)
150 /* postfilter specific values */
151 int do_apf; ///< whether to apply the averaged
152 ///< projection filter (APF)
153 int denoise_strength; ///< strength of denoising in Wiener filter
155 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
156 ///< Wiener filter coefficients (postfilter)
157 int dc_level; ///< Predicted amount of DC noise, based
158 ///< on which a DC removal filter is used
160 int lsps; ///< number of LSPs per frame [10 or 16]
161 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
162 int lsp_def_mode; ///< defines different sets of LSP defaults
164 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
165 ///< per-frame (independent coding)
166 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
167 ///< per superframe (residual coding)
169 int min_pitch_val; ///< base value for pitch parsing code
170 int max_pitch_val; ///< max value + 1 for pitch parsing
171 int pitch_nbits; ///< number of bits used to specify the
172 ///< pitch value in the frame header
173 int block_pitch_nbits; ///< number of bits used to specify the
174 ///< first block's pitch value
175 int block_pitch_range; ///< range of the block pitch
176 int block_delta_pitch_nbits; ///< number of bits used to specify the
177 ///< delta pitch between this and the last
178 ///< block's pitch value, used in all but
180 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
181 ///< from -this to +this-1)
182 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
188 * @name Packet values specified in the packet header or related to a packet.
190 * A packet is considered to be a single unit of data provided to this
191 * decoder by the demuxer.
194 int spillover_nbits; ///< number of bits of the previous packet's
195 ///< last superframe preceding this
196 ///< packet's first full superframe (useful
197 ///< for re-synchronization also)
198 int has_residual_lsps; ///< if set, superframes contain one set of
199 ///< LSPs that cover all frames, encoded as
200 ///< independent and residual LSPs; if not
201 ///< set, each frame contains its own, fully
202 ///< independent, LSPs
203 int skip_bits_next; ///< number of bits to skip at the next call
204 ///< to #wmavoice_decode_packet() (since
205 ///< they're part of the previous superframe)
207 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
208 ///< cache for superframe data split over
209 ///< multiple packets
210 int sframe_cache_size; ///< set to >0 if we have data from an
211 ///< (incomplete) superframe from a previous
212 ///< packet that spilled over in the current
213 ///< packet; specifies the amount of bits in
215 PutBitContext pb; ///< bitstream writer for #sframe_cache
220 * @name Frame and superframe values
221 * Superframe and frame data - these can change from frame to frame,
222 * although some of them do in that case serve as a cache / history for
223 * the next frame or superframe.
226 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
228 int last_pitch_val; ///< pitch value of the previous frame
229 int last_acb_type; ///< frame type [0-2] of the previous frame
230 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
231 ///< << 16) / #MAX_FRAMESIZE
232 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
234 int aw_idx_is_ext; ///< whether the AW index was encoded in
235 ///< 8 bits (instead of 6)
236 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
237 ///< can apply the pulse, relative to the
238 ///< value in aw_first_pulse_off. The exact
239 ///< position of the first AW-pulse is within
240 ///< [pulse_off, pulse_off + this], and
241 ///< depends on bitstream values; [16 or 24]
242 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
243 ///< that this number can be negative (in
244 ///< which case it basically means "zero")
245 int aw_first_pulse_off[2]; ///< index of first sample to which to
246 ///< apply AW-pulses, or -0xff if unset
247 int aw_next_pulse_off_cache; ///< the position (relative to start of the
248 ///< second block) at which pulses should
249 ///< start to be positioned, serves as a
250 ///< cache for pitch-adaptive window pulses
253 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
254 ///< only used for comfort noise in #pRNG()
255 float gain_pred_err[6]; ///< cache for gain prediction
256 float excitation_history[MAX_SIGNAL_HISTORY];
257 ///< cache of the signal of previous
258 ///< superframes, used as a history for
259 ///< signal generation
260 float synth_history[MAX_LSPS]; ///< see #excitation_history
264 * @name Postfilter values
266 * Variables used for postfilter implementation, mostly history for
267 * smoothing and so on, and context variables for FFT/iFFT.
270 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
271 ///< postfilter (for denoise filter)
272 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
273 ///< transform, part of postfilter)
274 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
276 float postfilter_agc; ///< gain control memory, used in
277 ///< #adaptive_gain_control()
278 float dcf_mem[2]; ///< DC filter history
279 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
280 ///< zero filter output (i.e. excitation)
282 float denoise_filter_cache[MAX_FRAMESIZE];
283 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
284 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
285 ///< aligned buffer for LPC tilting
286 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
287 ///< aligned buffer for denoise coefficients
288 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
289 ///< aligned buffer for postfilter speech
297 * Set up the variable bit mode (VBM) tree from container extradata.
298 * @param gb bit I/O context.
299 * The bit context (s->gb) should be loaded with byte 23-46 of the
300 * container extradata (i.e. the ones containing the VBM tree).
301 * @param vbm_tree pointer to array to which the decoded VBM tree will be
303 * @return 0 on success, <0 on error.
305 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
307 static const uint8_t bits[] = {
310 10, 10, 10, 12, 12, 12,
313 static const uint16_t codes[] = {
314 0x0000, 0x0001, 0x0002, // 00/01/10
315 0x000c, 0x000d, 0x000e, // 11+00/01/10
316 0x003c, 0x003d, 0x003e, // 1111+00/01/10
317 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
318 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
319 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
320 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
322 int cntr[8] = { 0 }, n, res;
324 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
325 for (n = 0; n < 17; n++) {
326 res = get_bits(gb, 3);
327 if (cntr[res] > 3) // should be >= 3 + (res == 7))
329 vbm_tree[res * 3 + cntr[res]++] = n;
331 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
332 bits, 1, 1, codes, 2, 2, 132);
337 * Set up decoder with parameters from demuxer (extradata etc.).
339 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
341 int n, flags, pitch_range, lsp16_flag;
342 WMAVoiceContext *s = ctx->priv_data;
346 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
347 * - byte 19-22: flags field (annoyingly in LE; see below for known
349 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
352 if (ctx->extradata_size != 46) {
353 av_log(ctx, AV_LOG_ERROR,
354 "Invalid extradata size %d (should be 46)\n",
355 ctx->extradata_size);
358 flags = AV_RL32(ctx->extradata + 18);
359 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
360 s->do_apf = flags & 0x1;
362 ff_rdft_init(&s->rdft, 7, DFT_R2C);
363 ff_rdft_init(&s->irdft, 7, IDFT_C2R);
364 ff_dct_init(&s->dct, 6, DCT_I);
365 ff_dct_init(&s->dst, 6, DST_I);
367 ff_sine_window_init(s->cos, 256);
368 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
369 for (n = 0; n < 255; n++) {
370 s->sin[n] = -s->sin[510 - n];
371 s->cos[510 - n] = s->cos[n];
374 s->denoise_strength = (flags >> 2) & 0xF;
375 if (s->denoise_strength >= 12) {
376 av_log(ctx, AV_LOG_ERROR,
377 "Invalid denoise filter strength %d (max=11)\n",
378 s->denoise_strength);
381 s->denoise_tilt_corr = !!(flags & 0x40);
382 s->dc_level = (flags >> 7) & 0xF;
383 s->lsp_q_mode = !!(flags & 0x2000);
384 s->lsp_def_mode = !!(flags & 0x4000);
385 lsp16_flag = flags & 0x1000;
388 s->frame_lsp_bitsize = 34;
389 s->sframe_lsp_bitsize = 60;
392 s->frame_lsp_bitsize = 24;
393 s->sframe_lsp_bitsize = 48;
395 for (n = 0; n < s->lsps; n++)
396 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
398 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
399 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
400 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
404 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
405 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
406 pitch_range = s->max_pitch_val - s->min_pitch_val;
407 if (pitch_range <= 0) {
408 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
411 s->pitch_nbits = av_ceil_log2(pitch_range);
412 s->last_pitch_val = 40;
413 s->last_acb_type = ACB_TYPE_NONE;
414 s->history_nsamples = s->max_pitch_val + 8;
416 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
417 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
418 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
420 av_log(ctx, AV_LOG_ERROR,
421 "Unsupported samplerate %d (min=%d, max=%d)\n",
422 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
427 s->block_conv_table[0] = s->min_pitch_val;
428 s->block_conv_table[1] = (pitch_range * 25) >> 6;
429 s->block_conv_table[2] = (pitch_range * 44) >> 6;
430 s->block_conv_table[3] = s->max_pitch_val - 1;
431 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
432 if (s->block_delta_pitch_hrange <= 0) {
433 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
436 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
437 s->block_pitch_range = s->block_conv_table[2] +
438 s->block_conv_table[3] + 1 +
439 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
440 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
443 ctx->channel_layout = AV_CH_LAYOUT_MONO;
444 ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
446 avcodec_get_frame_defaults(&s->frame);
447 ctx->coded_frame = &s->frame;
453 * @name Postfilter functions
454 * Postfilter functions (gain control, wiener denoise filter, DC filter,
455 * kalman smoothening, plus surrounding code to wrap it)
459 * Adaptive gain control (as used in postfilter).
461 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
462 * that the energy here is calculated using sum(abs(...)), whereas the
463 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
465 * @param out output buffer for filtered samples
466 * @param in input buffer containing the samples as they are after the
467 * postfilter steps so far
468 * @param speech_synth input buffer containing speech synth before postfilter
469 * @param size input buffer size
470 * @param alpha exponential filter factor
471 * @param gain_mem pointer to filter memory (single float)
473 static void adaptive_gain_control(float *out, const float *in,
474 const float *speech_synth,
475 int size, float alpha, float *gain_mem)
478 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
479 float mem = *gain_mem;
481 for (i = 0; i < size; i++) {
482 speech_energy += fabsf(speech_synth[i]);
483 postfilter_energy += fabsf(in[i]);
485 gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
487 for (i = 0; i < size; i++) {
488 mem = alpha * mem + gain_scale_factor;
489 out[i] = in[i] * mem;
496 * Kalman smoothing function.
498 * This function looks back pitch +/- 3 samples back into history to find
499 * the best fitting curve (that one giving the optimal gain of the two
500 * signals, i.e. the highest dot product between the two), and then
501 * uses that signal history to smoothen the output of the speech synthesis
504 * @param s WMA Voice decoding context
505 * @param pitch pitch of the speech signal
506 * @param in input speech signal
507 * @param out output pointer for smoothened signal
508 * @param size input/output buffer size
510 * @returns -1 if no smoothening took place, e.g. because no optimal
511 * fit could be found, or 0 on success.
513 static int kalman_smoothen(WMAVoiceContext *s, int pitch,
514 const float *in, float *out, int size)
517 float optimal_gain = 0, dot;
518 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
519 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
520 *best_hist_ptr = NULL;
522 /* find best fitting point in history */
524 dot = avpriv_scalarproduct_float_c(in, ptr, size);
525 if (dot > optimal_gain) {
529 } while (--ptr >= end);
531 if (optimal_gain <= 0)
533 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
534 if (dot <= 0) // would be 1.0
537 if (optimal_gain <= dot) {
538 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
542 /* actual smoothing */
543 for (n = 0; n < size; n++)
544 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
550 * Get the tilt factor of a formant filter from its transfer function
551 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
552 * but somehow (??) it does a speech synthesis filter in the
553 * middle, which is missing here
555 * @param lpcs LPC coefficients
556 * @param n_lpcs Size of LPC buffer
557 * @returns the tilt factor
559 static float tilt_factor(const float *lpcs, int n_lpcs)
563 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
564 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
570 * Derive denoise filter coefficients (in real domain) from the LPCs.
572 static void calc_input_response(WMAVoiceContext *s, float *lpcs,
573 int fcb_type, float *coeffs, int remainder)
575 float last_coeff, min = 15.0, max = -15.0;
576 float irange, angle_mul, gain_mul, range, sq;
579 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
580 s->rdft.rdft_calc(&s->rdft, lpcs);
581 #define log_range(var, assign) do { \
582 float tmp = log10f(assign); var = tmp; \
583 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
585 log_range(last_coeff, lpcs[1] * lpcs[1]);
586 for (n = 1; n < 64; n++)
587 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
588 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
589 log_range(lpcs[0], lpcs[0] * lpcs[0]);
592 lpcs[64] = last_coeff;
594 /* Now, use this spectrum to pick out these frequencies with higher
595 * (relative) power/energy (which we then take to be "not noise"),
596 * and set up a table (still in lpc[]) of (relative) gains per frequency.
597 * These frequencies will be maintained, while others ("noise") will be
598 * decreased in the filter output. */
599 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
600 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
602 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
603 for (n = 0; n <= 64; n++) {
606 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
607 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
608 lpcs[n] = angle_mul * pwr;
610 /* 70.57 =~ 1/log10(1.0331663) */
611 idx = (pwr * gain_mul - 0.0295) * 70.570526123;
612 if (idx > 127) { // fallback if index falls outside table range
613 coeffs[n] = wmavoice_energy_table[127] *
614 powf(1.0331663, idx - 127);
616 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
619 /* calculate the Hilbert transform of the gains, which we do (since this
620 * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
621 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
622 * "moment" of the LPCs in this filter. */
623 s->dct.dct_calc(&s->dct, lpcs);
624 s->dst.dct_calc(&s->dst, lpcs);
626 /* Split out the coefficient indexes into phase/magnitude pairs */
627 idx = 255 + av_clip(lpcs[64], -255, 255);
628 coeffs[0] = coeffs[0] * s->cos[idx];
629 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
630 last_coeff = coeffs[64] * s->cos[idx];
632 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
633 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
634 coeffs[n * 2] = coeffs[n] * s->cos[idx];
638 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
639 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
640 coeffs[n * 2] = coeffs[n] * s->cos[idx];
642 coeffs[1] = last_coeff;
644 /* move into real domain */
645 s->irdft.rdft_calc(&s->irdft, coeffs);
647 /* tilt correction and normalize scale */
648 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
649 if (s->denoise_tilt_corr) {
652 coeffs[remainder - 1] = 0;
653 ff_tilt_compensation(&tilt_mem,
654 -1.8 * tilt_factor(coeffs, remainder - 1),
657 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
659 for (n = 0; n < remainder; n++)
664 * This function applies a Wiener filter on the (noisy) speech signal as
665 * a means to denoise it.
667 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
668 * - using this power spectrum, calculate (for each frequency) the Wiener
669 * filter gain, which depends on the frequency power and desired level
670 * of noise subtraction (when set too high, this leads to artifacts)
671 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
673 * - by doing a phase shift, calculate the Hilbert transform of this array
674 * of per-frequency filter-gains to get the filtering coefficients;
675 * - smoothen/normalize/de-tilt these filter coefficients as desired;
676 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
677 * to get the denoised speech signal;
678 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
679 * the frame boundary) are saved and applied to subsequent frames by an
680 * overlap-add method (otherwise you get clicking-artifacts).
682 * @param s WMA Voice decoding context
683 * @param fcb_type Frame (codebook) type
684 * @param synth_pf input: the noisy speech signal, output: denoised speech
685 * data; should be 16-byte aligned (for ASM purposes)
686 * @param size size of the speech data
687 * @param lpcs LPCs used to synthesize this frame's speech data
689 static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
690 float *synth_pf, int size,
693 int remainder, lim, n;
695 if (fcb_type != FCB_TYPE_SILENCE) {
696 float *tilted_lpcs = s->tilted_lpcs_pf,
697 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
699 tilted_lpcs[0] = 1.0;
700 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
701 memset(&tilted_lpcs[s->lsps + 1], 0,
702 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
703 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
704 tilted_lpcs, s->lsps + 2);
706 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
707 * size is applied to the next frame. All input beyond this is zero,
708 * and thus all output beyond this will go towards zero, hence we can
709 * limit to min(size-1, 127-size) as a performance consideration. */
710 remainder = FFMIN(127 - size, size - 1);
711 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
713 /* apply coefficients (in frequency spectrum domain), i.e. complex
714 * number multiplication */
715 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
716 s->rdft.rdft_calc(&s->rdft, synth_pf);
717 s->rdft.rdft_calc(&s->rdft, coeffs);
718 synth_pf[0] *= coeffs[0];
719 synth_pf[1] *= coeffs[1];
720 for (n = 1; n < 64; n++) {
721 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
722 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
723 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
725 s->irdft.rdft_calc(&s->irdft, synth_pf);
728 /* merge filter output with the history of previous runs */
729 if (s->denoise_filter_cache_size) {
730 lim = FFMIN(s->denoise_filter_cache_size, size);
731 for (n = 0; n < lim; n++)
732 synth_pf[n] += s->denoise_filter_cache[n];
733 s->denoise_filter_cache_size -= lim;
734 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
735 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
738 /* move remainder of filter output into a cache for future runs */
739 if (fcb_type != FCB_TYPE_SILENCE) {
740 lim = FFMIN(remainder, s->denoise_filter_cache_size);
741 for (n = 0; n < lim; n++)
742 s->denoise_filter_cache[n] += synth_pf[size + n];
743 if (lim < remainder) {
744 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
745 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
746 s->denoise_filter_cache_size = remainder;
752 * Averaging projection filter, the postfilter used in WMAVoice.
754 * This uses the following steps:
755 * - A zero-synthesis filter (generate excitation from synth signal)
756 * - Kalman smoothing on excitation, based on pitch
757 * - Re-synthesized smoothened output
758 * - Iterative Wiener denoise filter
759 * - Adaptive gain filter
762 * @param s WMAVoice decoding context
763 * @param synth Speech synthesis output (before postfilter)
764 * @param samples Output buffer for filtered samples
765 * @param size Buffer size of synth & samples
766 * @param lpcs Generated LPCs used for speech synthesis
767 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
768 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
769 * @param pitch Pitch of the input signal
771 static void postfilter(WMAVoiceContext *s, const float *synth,
772 float *samples, int size,
773 const float *lpcs, float *zero_exc_pf,
774 int fcb_type, int pitch)
776 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
777 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
778 *synth_filter_in = zero_exc_pf;
780 av_assert0(size <= MAX_FRAMESIZE / 2);
782 /* generate excitation from input signal */
783 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
785 if (fcb_type >= FCB_TYPE_AW_PULSES &&
786 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
787 synth_filter_in = synth_filter_in_buf;
789 /* re-synthesize speech after smoothening, and keep history */
790 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
791 synth_filter_in, size, s->lsps);
792 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
793 sizeof(synth_pf[0]) * s->lsps);
795 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
797 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
800 if (s->dc_level > 8) {
801 /* remove ultra-low frequency DC noise / highpass filter;
802 * coefficients are identical to those used in SIPR decoding,
803 * and very closely resemble those used in AMR-NB decoding. */
804 ff_acelp_apply_order_2_transfer_function(samples, samples,
805 (const float[2]) { -1.99997, 1.0 },
806 (const float[2]) { -1.9330735188, 0.93589198496 },
807 0.93980580475, s->dcf_mem, size);
816 * @param lsps output pointer to the array that will hold the LSPs
817 * @param num number of LSPs to be dequantized
818 * @param values quantized values, contains n_stages values
819 * @param sizes range (i.e. max value) of each quantized value
820 * @param n_stages number of dequantization runs
821 * @param table dequantization table to be used
822 * @param mul_q LSF multiplier
823 * @param base_q base (lowest) LSF values
825 static void dequant_lsps(double *lsps, int num,
826 const uint16_t *values,
827 const uint16_t *sizes,
828 int n_stages, const uint8_t *table,
830 const double *base_q)
834 memset(lsps, 0, num * sizeof(*lsps));
835 for (n = 0; n < n_stages; n++) {
836 const uint8_t *t_off = &table[values[n] * num];
837 double base = base_q[n], mul = mul_q[n];
839 for (m = 0; m < num; m++)
840 lsps[m] += base + mul * t_off[m];
842 table += sizes[n] * num;
847 * @name LSP dequantization routines
848 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
849 * @note we assume enough bits are available, caller should check.
850 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
851 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
855 * Parse 10 independently-coded LSPs.
857 static void dequant_lsp10i(GetBitContext *gb, double *lsps)
859 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
860 static const double mul_lsf[4] = {
861 5.2187144800e-3, 1.4626986422e-3,
862 9.6179549166e-4, 1.1325736225e-3
864 static const double base_lsf[4] = {
865 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
866 M_PI * -3.3486e-2, M_PI * -5.7408e-2
870 v[0] = get_bits(gb, 8);
871 v[1] = get_bits(gb, 6);
872 v[2] = get_bits(gb, 5);
873 v[3] = get_bits(gb, 5);
875 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
880 * Parse 10 independently-coded LSPs, and then derive the tables to
881 * generate LSPs for the other frames from them (residual coding).
883 static void dequant_lsp10r(GetBitContext *gb,
884 double *i_lsps, const double *old,
885 double *a1, double *a2, int q_mode)
887 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
888 static const double mul_lsf[3] = {
889 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
891 static const double base_lsf[3] = {
892 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
894 const float (*ipol_tab)[2][10] = q_mode ?
895 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
896 uint16_t interpol, v[3];
899 dequant_lsp10i(gb, i_lsps);
901 interpol = get_bits(gb, 5);
902 v[0] = get_bits(gb, 7);
903 v[1] = get_bits(gb, 6);
904 v[2] = get_bits(gb, 6);
906 for (n = 0; n < 10; n++) {
907 double delta = old[n] - i_lsps[n];
908 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
909 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
912 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
917 * Parse 16 independently-coded LSPs.
919 static void dequant_lsp16i(GetBitContext *gb, double *lsps)
921 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
922 static const double mul_lsf[5] = {
923 3.3439586280e-3, 6.9908173703e-4,
924 3.3216608306e-3, 1.0334960326e-3,
927 static const double base_lsf[5] = {
928 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
929 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
934 v[0] = get_bits(gb, 8);
935 v[1] = get_bits(gb, 6);
936 v[2] = get_bits(gb, 7);
937 v[3] = get_bits(gb, 6);
938 v[4] = get_bits(gb, 7);
940 dequant_lsps( lsps, 5, v, vec_sizes, 2,
941 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
942 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
943 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
944 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
945 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
949 * Parse 16 independently-coded LSPs, and then derive the tables to
950 * generate LSPs for the other frames from them (residual coding).
952 static void dequant_lsp16r(GetBitContext *gb,
953 double *i_lsps, const double *old,
954 double *a1, double *a2, int q_mode)
956 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
957 static const double mul_lsf[3] = {
958 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
960 static const double base_lsf[3] = {
961 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
963 const float (*ipol_tab)[2][16] = q_mode ?
964 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
965 uint16_t interpol, v[3];
968 dequant_lsp16i(gb, i_lsps);
970 interpol = get_bits(gb, 5);
971 v[0] = get_bits(gb, 7);
972 v[1] = get_bits(gb, 7);
973 v[2] = get_bits(gb, 7);
975 for (n = 0; n < 16; n++) {
976 double delta = old[n] - i_lsps[n];
977 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
978 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
981 dequant_lsps( a2, 10, v, vec_sizes, 1,
982 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
983 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
984 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
985 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
986 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
991 * @name Pitch-adaptive window coding functions
992 * The next few functions are for pitch-adaptive window coding.
996 * Parse the offset of the first pitch-adaptive window pulses, and
997 * the distribution of pulses between the two blocks in this frame.
998 * @param s WMA Voice decoding context private data
999 * @param gb bit I/O context
1000 * @param pitch pitch for each block in this frame
1002 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
1005 static const int16_t start_offset[94] = {
1006 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1007 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1008 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1009 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1010 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1011 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1012 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1013 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1017 /* position of pulse */
1018 s->aw_idx_is_ext = 0;
1019 if ((bits = get_bits(gb, 6)) >= 54) {
1020 s->aw_idx_is_ext = 1;
1021 bits += (bits - 54) * 3 + get_bits(gb, 2);
1024 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1025 * the distribution of the pulses in each block contained in this frame. */
1026 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1027 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1028 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1029 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1030 offset += s->aw_n_pulses[0] * pitch[0];
1031 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1032 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1034 /* if continuing from a position before the block, reset position to
1035 * start of block (when corrected for the range over which it can be
1036 * spread in aw_pulse_set1()). */
1037 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1038 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1039 s->aw_first_pulse_off[1] -= pitch[1];
1040 if (start_offset[bits] < 0)
1041 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1042 s->aw_first_pulse_off[0] -= pitch[0];
1047 * Apply second set of pitch-adaptive window pulses.
1048 * @param s WMA Voice decoding context private data
1049 * @param gb bit I/O context
1050 * @param block_idx block index in frame [0, 1]
1051 * @param fcb structure containing fixed codebook vector info
1053 static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1054 int block_idx, AMRFixed *fcb)
1056 uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
1057 uint16_t *use_mask = use_mask_mem + 2;
1058 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1059 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1060 * of idx are the position of the bit within a particular item in the
1061 * array (0 being the most significant bit, and 15 being the least
1062 * significant bit), and the remainder (>> 4) is the index in the
1063 * use_mask[]-array. This is faster and uses less memory than using a
1064 * 80-byte/80-int array. */
1065 int pulse_off = s->aw_first_pulse_off[block_idx],
1066 pulse_start, n, idx, range, aidx, start_off = 0;
1068 /* set offset of first pulse to within this block */
1069 if (s->aw_n_pulses[block_idx] > 0)
1070 while (pulse_off + s->aw_pulse_range < 1)
1071 pulse_off += fcb->pitch_lag;
1073 /* find range per pulse */
1074 if (s->aw_n_pulses[0] > 0) {
1075 if (block_idx == 0) {
1077 } else /* block_idx = 1 */ {
1079 if (s->aw_n_pulses[block_idx] > 0)
1080 pulse_off = s->aw_next_pulse_off_cache;
1084 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1086 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1087 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1088 * we exclude that range from being pulsed again in this function. */
1089 memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
1090 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
1091 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1092 if (s->aw_n_pulses[block_idx] > 0)
1093 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1094 int excl_range = s->aw_pulse_range; // always 16 or 24
1095 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1096 int first_sh = 16 - (idx & 15);
1097 *use_mask_ptr++ &= 0xFFFFu << first_sh;
1098 excl_range -= first_sh;
1099 if (excl_range >= 16) {
1100 *use_mask_ptr++ = 0;
1101 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1103 *use_mask_ptr &= 0xFFFF >> excl_range;
1106 /* find the 'aidx'th offset that is not excluded */
1107 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1108 for (n = 0; n <= aidx; pulse_start++) {
1109 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1110 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1111 if (use_mask[0]) idx = 0x0F;
1112 else if (use_mask[1]) idx = 0x1F;
1113 else if (use_mask[2]) idx = 0x2F;
1114 else if (use_mask[3]) idx = 0x3F;
1115 else if (use_mask[4]) idx = 0x4F;
1117 idx -= av_log2_16bit(use_mask[idx >> 4]);
1119 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1120 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1126 fcb->x[fcb->n] = start_off;
1127 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1130 /* set offset for next block, relative to start of that block */
1131 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1132 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1136 * Apply first set of pitch-adaptive window pulses.
1137 * @param s WMA Voice decoding context private data
1138 * @param gb bit I/O context
1139 * @param block_idx block index in frame [0, 1]
1140 * @param fcb storage location for fixed codebook pulse info
1142 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1143 int block_idx, AMRFixed *fcb)
1145 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1148 if (s->aw_n_pulses[block_idx] > 0) {
1149 int n, v_mask, i_mask, sh, n_pulses;
1151 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1156 } else { // 4 pulses, 1:sign + 2:index each
1163 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1164 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1165 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1166 s->aw_first_pulse_off[block_idx];
1167 while (fcb->x[fcb->n] < 0)
1168 fcb->x[fcb->n] += fcb->pitch_lag;
1169 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1173 int num2 = (val & 0x1FF) >> 1, delta, idx;
1175 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1176 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1177 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1178 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1179 v = (val & 0x200) ? -1.0 : 1.0;
1181 fcb->no_repeat_mask |= 3 << fcb->n;
1182 fcb->x[fcb->n] = idx - delta;
1184 fcb->x[fcb->n + 1] = idx;
1185 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1193 * Generate a random number from frame_cntr and block_idx, which will lief
1194 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1195 * table of size 1000 of which you want to read block_size entries).
1197 * @param frame_cntr current frame number
1198 * @param block_num current block index
1199 * @param block_size amount of entries we want to read from a table
1200 * that has 1000 entries
1201 * @return a (non-)random number in the [0, 1000 - block_size] range.
1203 static int pRNG(int frame_cntr, int block_num, int block_size)
1205 /* array to simplify the calculation of z:
1206 * y = (x % 9) * 5 + 6;
1207 * z = (49995 * x) / y;
1208 * Since y only has 9 values, we can remove the division by using a
1209 * LUT and using FASTDIV-style divisions. For each of the 9 values
1210 * of y, we can rewrite z as:
1211 * z = x * (49995 / y) + x * ((49995 % y) / y)
1212 * In this table, each col represents one possible value of y, the
1213 * first number is 49995 / y, and the second is the FASTDIV variant
1214 * of 49995 % y / y. */
1215 static const unsigned int div_tbl[9][2] = {
1216 { 8332, 3 * 715827883U }, // y = 6
1217 { 4545, 0 * 390451573U }, // y = 11
1218 { 3124, 11 * 268435456U }, // y = 16
1219 { 2380, 15 * 204522253U }, // y = 21
1220 { 1922, 23 * 165191050U }, // y = 26
1221 { 1612, 23 * 138547333U }, // y = 31
1222 { 1388, 27 * 119304648U }, // y = 36
1223 { 1219, 16 * 104755300U }, // y = 41
1224 { 1086, 39 * 93368855U } // y = 46
1226 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1227 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1228 // so this is effectively a modulo (%)
1229 y = x - 9 * MULH(477218589, x); // x % 9
1230 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1231 // z = x * 49995 / (y * 5 + 6)
1232 return z % (1000 - block_size);
1236 * Parse hardcoded signal for a single block.
1237 * @note see #synth_block().
1239 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1240 int block_idx, int size,
1241 const struct frame_type_desc *frame_desc,
1247 av_assert0(size <= MAX_FRAMESIZE);
1249 /* Set the offset from which we start reading wmavoice_std_codebook */
1250 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1251 r_idx = pRNG(s->frame_cntr, block_idx, size);
1252 gain = s->silence_gain;
1253 } else /* FCB_TYPE_HARDCODED */ {
1254 r_idx = get_bits(gb, 8);
1255 gain = wmavoice_gain_universal[get_bits(gb, 6)];
1258 /* Clear gain prediction parameters */
1259 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1261 /* Apply gain to hardcoded codebook and use that as excitation signal */
1262 for (n = 0; n < size; n++)
1263 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1267 * Parse FCB/ACB signal for a single block.
1268 * @note see #synth_block().
1270 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1271 int block_idx, int size,
1272 int block_pitch_sh2,
1273 const struct frame_type_desc *frame_desc,
1276 static const float gain_coeff[6] = {
1277 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1279 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1280 int n, idx, gain_weight;
1283 av_assert0(size <= MAX_FRAMESIZE / 2);
1284 memset(pulses, 0, sizeof(*pulses) * size);
1286 fcb.pitch_lag = block_pitch_sh2 >> 2;
1287 fcb.pitch_fac = 1.0;
1288 fcb.no_repeat_mask = 0;
1291 /* For the other frame types, this is where we apply the innovation
1292 * (fixed) codebook pulses of the speech signal. */
1293 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1294 aw_pulse_set1(s, gb, block_idx, &fcb);
1295 aw_pulse_set2(s, gb, block_idx, &fcb);
1296 } else /* FCB_TYPE_EXC_PULSES */ {
1297 int offset_nbits = 5 - frame_desc->log_n_blocks;
1299 fcb.no_repeat_mask = -1;
1300 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1301 * (instead of double) for a subset of pulses */
1302 for (n = 0; n < 5; n++) {
1306 sign = get_bits1(gb) ? 1.0 : -1.0;
1307 pos1 = get_bits(gb, offset_nbits);
1308 fcb.x[fcb.n] = n + 5 * pos1;
1309 fcb.y[fcb.n++] = sign;
1310 if (n < frame_desc->dbl_pulses) {
1311 pos2 = get_bits(gb, offset_nbits);
1312 fcb.x[fcb.n] = n + 5 * pos2;
1313 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1317 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1319 /* Calculate gain for adaptive & fixed codebook signal.
1320 * see ff_amr_set_fixed_gain(). */
1321 idx = get_bits(gb, 7);
1322 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
1324 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1325 acb_gain = wmavoice_gain_codebook_acb[idx];
1326 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1327 -2.9957322736 /* log(0.05) */,
1328 1.6094379124 /* log(5.0) */);
1330 gain_weight = 8 >> frame_desc->log_n_blocks;
1331 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1332 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1333 for (n = 0; n < gain_weight; n++)
1334 s->gain_pred_err[n] = pred_err;
1336 /* Calculation of adaptive codebook */
1337 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1339 for (n = 0; n < size; n += len) {
1341 int abs_idx = block_idx * size + n;
1342 int pitch_sh16 = (s->last_pitch_val << 16) +
1343 s->pitch_diff_sh16 * abs_idx;
1344 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1345 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1346 idx = idx_sh16 >> 16;
1347 if (s->pitch_diff_sh16) {
1348 if (s->pitch_diff_sh16 > 0) {
1349 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1351 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1352 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1357 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1358 wmavoice_ipol1_coeffs, 17,
1361 } else /* ACB_TYPE_HAMMING */ {
1362 int block_pitch = block_pitch_sh2 >> 2;
1363 idx = block_pitch_sh2 & 3;
1365 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1366 wmavoice_ipol2_coeffs, 4,
1369 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1370 sizeof(float) * size);
1373 /* Interpolate ACB/FCB and use as excitation signal */
1374 ff_weighted_vector_sumf(excitation, excitation, pulses,
1375 acb_gain, fcb_gain, size);
1379 * Parse data in a single block.
1380 * @note we assume enough bits are available, caller should check.
1382 * @param s WMA Voice decoding context private data
1383 * @param gb bit I/O context
1384 * @param block_idx index of the to-be-read block
1385 * @param size amount of samples to be read in this block
1386 * @param block_pitch_sh2 pitch for this block << 2
1387 * @param lsps LSPs for (the end of) this frame
1388 * @param prev_lsps LSPs for the last frame
1389 * @param frame_desc frame type descriptor
1390 * @param excitation target memory for the ACB+FCB interpolated signal
1391 * @param synth target memory for the speech synthesis filter output
1392 * @return 0 on success, <0 on error.
1394 static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1395 int block_idx, int size,
1396 int block_pitch_sh2,
1397 const double *lsps, const double *prev_lsps,
1398 const struct frame_type_desc *frame_desc,
1399 float *excitation, float *synth)
1401 double i_lsps[MAX_LSPS];
1402 float lpcs[MAX_LSPS];
1406 if (frame_desc->acb_type == ACB_TYPE_NONE)
1407 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1409 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1410 frame_desc, excitation);
1412 /* convert interpolated LSPs to LPCs */
1413 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1414 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1415 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1416 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1418 /* Speech synthesis */
1419 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1423 * Synthesize output samples for a single frame.
1424 * @note we assume enough bits are available, caller should check.
1426 * @param ctx WMA Voice decoder context
1427 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1428 * @param frame_idx Frame number within superframe [0-2]
1429 * @param samples pointer to output sample buffer, has space for at least 160
1431 * @param lsps LSP array
1432 * @param prev_lsps array of previous frame's LSPs
1433 * @param excitation target buffer for excitation signal
1434 * @param synth target buffer for synthesized speech data
1435 * @return 0 on success, <0 on error.
1437 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1439 const double *lsps, const double *prev_lsps,
1440 float *excitation, float *synth)
1442 WMAVoiceContext *s = ctx->priv_data;
1443 int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val);
1444 int pitch[MAX_BLOCKS], av_uninit(last_block_pitch);
1446 /* Parse frame type ("frame header"), see frame_descs */
1447 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
1450 av_log(ctx, AV_LOG_ERROR,
1451 "Invalid frame type VLC code, skipping\n");
1455 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1457 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1458 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1459 /* Pitch is provided per frame, which is interpreted as the pitch of
1460 * the last sample of the last block of this frame. We can interpolate
1461 * the pitch of other blocks (and even pitch-per-sample) by gradually
1462 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1463 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1464 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1465 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1466 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1467 if (s->last_acb_type == ACB_TYPE_NONE ||
1468 20 * abs(cur_pitch_val - s->last_pitch_val) >
1469 (cur_pitch_val + s->last_pitch_val))
1470 s->last_pitch_val = cur_pitch_val;
1472 /* pitch per block */
1473 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1474 int fac = n * 2 + 1;
1476 pitch[n] = (MUL16(fac, cur_pitch_val) +
1477 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1478 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1481 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1482 s->pitch_diff_sh16 =
1483 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1486 /* Global gain (if silence) and pitch-adaptive window coordinates */
1487 switch (frame_descs[bd_idx].fcb_type) {
1488 case FCB_TYPE_SILENCE:
1489 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1491 case FCB_TYPE_AW_PULSES:
1492 aw_parse_coords(s, gb, pitch);
1496 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1499 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1500 switch (frame_descs[bd_idx].acb_type) {
1501 case ACB_TYPE_HAMMING: {
1502 /* Pitch is given per block. Per-block pitches are encoded as an
1503 * absolute value for the first block, and then delta values
1504 * relative to this value) for all subsequent blocks. The scale of
1505 * this pitch value is semi-logaritmic compared to its use in the
1506 * decoder, so we convert it to normal scale also. */
1508 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1509 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1510 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1513 block_pitch = get_bits(gb, s->block_pitch_nbits);
1515 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1516 get_bits(gb, s->block_delta_pitch_nbits);
1517 /* Convert last_ so that any next delta is within _range */
1518 last_block_pitch = av_clip(block_pitch,
1519 s->block_delta_pitch_hrange,
1520 s->block_pitch_range -
1521 s->block_delta_pitch_hrange);
1523 /* Convert semi-log-style scale back to normal scale */
1524 if (block_pitch < t1) {
1525 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1528 if (block_pitch < t2) {
1530 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1533 if (block_pitch < t3) {
1535 (s->block_conv_table[2] + block_pitch) << 2;
1537 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1540 pitch[n] = bl_pitch_sh2 >> 2;
1544 case ACB_TYPE_ASYMMETRIC: {
1545 bl_pitch_sh2 = pitch[n] << 2;
1549 default: // ACB_TYPE_NONE has no pitch
1554 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1555 lsps, prev_lsps, &frame_descs[bd_idx],
1556 &excitation[n * block_nsamples],
1557 &synth[n * block_nsamples]);
1560 /* Averaging projection filter, if applicable. Else, just copy samples
1561 * from synthesis buffer */
1563 double i_lsps[MAX_LSPS];
1564 float lpcs[MAX_LSPS];
1566 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1567 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1568 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1569 postfilter(s, synth, samples, 80, lpcs,
1570 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1571 frame_descs[bd_idx].fcb_type, pitch[0]);
1573 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1574 i_lsps[n] = cos(lsps[n]);
1575 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1576 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1577 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1578 frame_descs[bd_idx].fcb_type, pitch[0]);
1580 memcpy(samples, synth, 160 * sizeof(synth[0]));
1582 /* Cache values for next frame */
1584 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1585 s->last_acb_type = frame_descs[bd_idx].acb_type;
1586 switch (frame_descs[bd_idx].acb_type) {
1588 s->last_pitch_val = 0;
1590 case ACB_TYPE_ASYMMETRIC:
1591 s->last_pitch_val = cur_pitch_val;
1593 case ACB_TYPE_HAMMING:
1594 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1602 * Ensure minimum value for first item, maximum value for last value,
1603 * proper spacing between each value and proper ordering.
1605 * @param lsps array of LSPs
1606 * @param num size of LSP array
1608 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1609 * useful to put in a generic location later on. Parts are also
1610 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1611 * which is in float.
1613 static void stabilize_lsps(double *lsps, int num)
1617 /* set minimum value for first, maximum value for last and minimum
1618 * spacing between LSF values.
1619 * Very similar to ff_set_min_dist_lsf(), but in double. */
1620 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1621 for (n = 1; n < num; n++)
1622 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1623 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1625 /* reorder (looks like one-time / non-recursed bubblesort).
1626 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1627 for (n = 1; n < num; n++) {
1628 if (lsps[n] < lsps[n - 1]) {
1629 for (m = 1; m < num; m++) {
1630 double tmp = lsps[m];
1631 for (l = m - 1; l >= 0; l--) {
1632 if (lsps[l] <= tmp) break;
1633 lsps[l + 1] = lsps[l];
1643 * Test if there's enough bits to read 1 superframe.
1645 * @param orig_gb bit I/O context used for reading. This function
1646 * does not modify the state of the bitreader; it
1647 * only uses it to copy the current stream position
1648 * @param s WMA Voice decoding context private data
1649 * @return -1 if unsupported, 1 on not enough bits or 0 if OK.
1651 static int check_bits_for_superframe(GetBitContext *orig_gb,
1654 GetBitContext s_gb, *gb = &s_gb;
1655 int n, need_bits, bd_idx;
1656 const struct frame_type_desc *frame_desc;
1658 /* initialize a copy */
1659 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
1660 skip_bits_long(gb, get_bits_count(orig_gb));
1661 av_assert1(get_bits_left(gb) == get_bits_left(orig_gb));
1663 /* superframe header */
1664 if (get_bits_left(gb) < 14)
1667 return -1; // WMAPro-in-WMAVoice superframe
1668 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe
1669 if (s->has_residual_lsps) { // residual LSPs (for all frames)
1670 if (get_bits_left(gb) < s->sframe_lsp_bitsize)
1672 skip_bits_long(gb, s->sframe_lsp_bitsize);
1676 for (n = 0; n < MAX_FRAMES; n++) {
1677 int aw_idx_is_ext = 0;
1679 if (!s->has_residual_lsps) { // independent LSPs (per-frame)
1680 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
1681 skip_bits_long(gb, s->frame_lsp_bitsize);
1683 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
1685 return -1; // invalid frame type VLC code
1686 frame_desc = &frame_descs[bd_idx];
1687 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1688 if (get_bits_left(gb) < s->pitch_nbits)
1690 skip_bits_long(gb, s->pitch_nbits);
1692 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1694 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1695 int tmp = get_bits(gb, 6);
1703 if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
1704 need_bits = s->block_pitch_nbits +
1705 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
1706 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1707 need_bits = 2 * !aw_idx_is_ext;
1710 need_bits += frame_desc->frame_size;
1711 if (get_bits_left(gb) < need_bits)
1713 skip_bits_long(gb, need_bits);
1720 * Synthesize output samples for a single superframe. If we have any data
1721 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1724 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1725 * to give a total of 480 samples per frame. See #synth_frame() for frame
1726 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1727 * (if these are globally specified for all frames (residually); they can
1728 * also be specified individually per-frame. See the s->has_residual_lsps
1729 * option), and can specify the number of samples encoded in this superframe
1730 * (if less than 480), usually used to prevent blanks at track boundaries.
1732 * @param ctx WMA Voice decoder context
1733 * @return 0 on success, <0 on error or 1 if there was not enough data to
1734 * fully parse the superframe
1736 static int synth_superframe(AVCodecContext *ctx, int *got_frame_ptr)
1738 WMAVoiceContext *s = ctx->priv_data;
1739 GetBitContext *gb = &s->gb, s_gb;
1740 int n, res, n_samples = 480;
1741 double lsps[MAX_FRAMES][MAX_LSPS];
1742 const double *mean_lsf = s->lsps == 16 ?
1743 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1744 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1745 float synth[MAX_LSPS + MAX_SFRAMESIZE];
1748 memcpy(synth, s->synth_history,
1749 s->lsps * sizeof(*synth));
1750 memcpy(excitation, s->excitation_history,
1751 s->history_nsamples * sizeof(*excitation));
1753 if (s->sframe_cache_size > 0) {
1755 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1756 s->sframe_cache_size = 0;
1759 if ((res = check_bits_for_superframe(gb, s)) == 1) {
1764 /* First bit is speech/music bit, it differentiates between WMAVoice
1765 * speech samples (the actual codec) and WMAVoice music samples, which
1766 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1768 if (!get_bits1(gb)) {
1769 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice", 1);
1770 return AVERROR_PATCHWELCOME;
1773 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1774 if (get_bits1(gb)) {
1775 if ((n_samples = get_bits(gb, 12)) > 480) {
1776 av_log(ctx, AV_LOG_ERROR,
1777 "Superframe encodes >480 samples (%d), not allowed\n",
1782 /* Parse LSPs, if global for the superframe (can also be per-frame). */
1783 if (s->has_residual_lsps) {
1784 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1786 for (n = 0; n < s->lsps; n++)
1787 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1789 if (s->lsps == 10) {
1790 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1791 } else /* s->lsps == 16 */
1792 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1794 for (n = 0; n < s->lsps; n++) {
1795 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
1796 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1797 lsps[2][n] += mean_lsf[n];
1799 for (n = 0; n < 3; n++)
1800 stabilize_lsps(lsps[n], s->lsps);
1803 /* get output buffer */
1804 s->frame.nb_samples = 480;
1805 if ((res = ff_get_buffer(ctx, &s->frame)) < 0) {
1806 av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n");
1809 s->frame.nb_samples = n_samples;
1810 samples = (float *)s->frame.data[0];
1812 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1813 for (n = 0; n < 3; n++) {
1814 if (!s->has_residual_lsps) {
1817 if (s->lsps == 10) {
1818 dequant_lsp10i(gb, lsps[n]);
1819 } else /* s->lsps == 16 */
1820 dequant_lsp16i(gb, lsps[n]);
1822 for (m = 0; m < s->lsps; m++)
1823 lsps[n][m] += mean_lsf[m];
1824 stabilize_lsps(lsps[n], s->lsps);
1827 if ((res = synth_frame(ctx, gb, n,
1828 &samples[n * MAX_FRAMESIZE],
1829 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1830 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1831 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
1837 /* Statistics? FIXME - we don't check for length, a slight overrun
1838 * will be caught by internal buffer padding, and anything else
1839 * will be skipped, not read. */
1840 if (get_bits1(gb)) {
1841 res = get_bits(gb, 4);
1842 skip_bits(gb, 10 * (res + 1));
1847 /* Update history */
1848 memcpy(s->prev_lsps, lsps[2],
1849 s->lsps * sizeof(*s->prev_lsps));
1850 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1851 s->lsps * sizeof(*synth));
1852 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1853 s->history_nsamples * sizeof(*excitation));
1855 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1856 s->history_nsamples * sizeof(*s->zero_exc_pf));
1862 * Parse the packet header at the start of each packet (input data to this
1865 * @param s WMA Voice decoding context private data
1866 * @return 1 if not enough bits were available, or 0 on success.
1868 static int parse_packet_header(WMAVoiceContext *s)
1870 GetBitContext *gb = &s->gb;
1873 if (get_bits_left(gb) < 11)
1875 skip_bits(gb, 4); // packet sequence number
1876 s->has_residual_lsps = get_bits1(gb);
1878 res = get_bits(gb, 6); // number of superframes per packet
1879 // (minus first one if there is spillover)
1880 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
1882 } while (res == 0x3F);
1883 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
1889 * Copy (unaligned) bits from gb/data/size to pb.
1891 * @param pb target buffer to copy bits into
1892 * @param data source buffer to copy bits from
1893 * @param size size of the source data, in bytes
1894 * @param gb bit I/O context specifying the current position in the source.
1895 * data. This function might use this to align the bit position to
1896 * a whole-byte boundary before calling #avpriv_copy_bits() on aligned
1898 * @param nbits the amount of bits to copy from source to target
1900 * @note after calling this function, the current position in the input bit
1901 * I/O context is undefined.
1903 static void copy_bits(PutBitContext *pb,
1904 const uint8_t *data, int size,
1905 GetBitContext *gb, int nbits)
1907 int rmn_bytes, rmn_bits;
1909 rmn_bits = rmn_bytes = get_bits_left(gb);
1910 if (rmn_bits < nbits)
1912 if (nbits > pb->size_in_bits - put_bits_count(pb))
1914 rmn_bits &= 7; rmn_bytes >>= 3;
1915 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1916 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1917 avpriv_copy_bits(pb, data + size - rmn_bytes,
1918 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1922 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1923 * and we expect that the demuxer / application provides it to us as such
1924 * (else you'll probably get garbage as output). Every packet has a size of
1925 * ctx->block_align bytes, starts with a packet header (see
1926 * #parse_packet_header()), and then a series of superframes. Superframe
1927 * boundaries may exceed packets, i.e. superframes can split data over
1928 * multiple (two) packets.
1930 * For more information about frames, see #synth_superframe().
1932 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1933 int *got_frame_ptr, AVPacket *avpkt)
1935 WMAVoiceContext *s = ctx->priv_data;
1936 GetBitContext *gb = &s->gb;
1939 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1940 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1941 * feeds us ASF packets, which may concatenate multiple "codec" packets
1942 * in a single "muxer" packet, so we artificially emulate that by
1943 * capping the packet size at ctx->block_align. */
1944 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1949 init_get_bits(&s->gb, avpkt->data, size << 3);
1951 /* size == ctx->block_align is used to indicate whether we are dealing with
1952 * a new packet or a packet of which we already read the packet header
1954 if (size == ctx->block_align) { // new packet header
1955 if ((res = parse_packet_header(s)) < 0)
1958 /* If the packet header specifies a s->spillover_nbits, then we want
1959 * to push out all data of the previous packet (+ spillover) before
1960 * continuing to parse new superframes in the current packet. */
1961 if (s->spillover_nbits > 0) {
1962 if (s->sframe_cache_size > 0) {
1963 int cnt = get_bits_count(gb);
1964 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1965 flush_put_bits(&s->pb);
1966 s->sframe_cache_size += s->spillover_nbits;
1967 if ((res = synth_superframe(ctx, got_frame_ptr)) == 0 &&
1969 cnt += s->spillover_nbits;
1970 s->skip_bits_next = cnt & 7;
1971 *(AVFrame *)data = s->frame;
1974 skip_bits_long (gb, s->spillover_nbits - cnt +
1975 get_bits_count(gb)); // resync
1977 skip_bits_long(gb, s->spillover_nbits); // resync
1979 } else if (s->skip_bits_next)
1980 skip_bits(gb, s->skip_bits_next);
1982 /* Try parsing superframes in current packet */
1983 s->sframe_cache_size = 0;
1984 s->skip_bits_next = 0;
1985 pos = get_bits_left(gb);
1986 if ((res = synth_superframe(ctx, got_frame_ptr)) < 0) {
1988 } else if (*got_frame_ptr) {
1989 int cnt = get_bits_count(gb);
1990 s->skip_bits_next = cnt & 7;
1991 *(AVFrame *)data = s->frame;
1993 } else if ((s->sframe_cache_size = pos) > 0) {
1994 /* rewind bit reader to start of last (incomplete) superframe... */
1995 init_get_bits(gb, avpkt->data, size << 3);
1996 skip_bits_long(gb, (size << 3) - pos);
1997 av_assert1(get_bits_left(gb) == pos);
1999 /* ...and cache it for spillover in next packet */
2000 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
2001 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
2002 // FIXME bad - just copy bytes as whole and add use the
2003 // skip_bits_next field
2009 static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
2011 WMAVoiceContext *s = ctx->priv_data;
2014 ff_rdft_end(&s->rdft);
2015 ff_rdft_end(&s->irdft);
2016 ff_dct_end(&s->dct);
2017 ff_dct_end(&s->dst);
2023 static av_cold void wmavoice_flush(AVCodecContext *ctx)
2025 WMAVoiceContext *s = ctx->priv_data;
2028 s->postfilter_agc = 0;
2029 s->sframe_cache_size = 0;
2030 s->skip_bits_next = 0;
2031 for (n = 0; n < s->lsps; n++)
2032 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2033 memset(s->excitation_history, 0,
2034 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2035 memset(s->synth_history, 0,
2036 sizeof(*s->synth_history) * MAX_LSPS);
2037 memset(s->gain_pred_err, 0,
2038 sizeof(s->gain_pred_err));
2041 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2042 sizeof(*s->synth_filter_out_buf) * s->lsps);
2043 memset(s->dcf_mem, 0,
2044 sizeof(*s->dcf_mem) * 2);
2045 memset(s->zero_exc_pf, 0,
2046 sizeof(*s->zero_exc_pf) * s->history_nsamples);
2047 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2051 AVCodec ff_wmavoice_decoder = {
2053 .type = AVMEDIA_TYPE_AUDIO,
2054 .id = AV_CODEC_ID_WMAVOICE,
2055 .priv_data_size = sizeof(WMAVoiceContext),
2056 .init = wmavoice_decode_init,
2057 .close = wmavoice_decode_end,
2058 .decode = wmavoice_decode_packet,
2059 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
2060 .flush = wmavoice_flush,
2061 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),