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.
132 typedef struct WMAVoiceContext {
134 * @name Global values specified in the stream header / extradata or used all over.
137 GetBitContext gb; ///< packet bitreader. During decoder init,
138 ///< it contains the extradata from the
139 ///< demuxer. During decoding, it contains
141 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
143 int spillover_bitsize; ///< number of bits used to specify
144 ///< #spillover_nbits in the packet header
145 ///< = ceil(log2(ctx->block_align << 3))
146 int history_nsamples; ///< number of samples in history for signal
147 ///< prediction (through ACB)
149 /* postfilter specific values */
150 int do_apf; ///< whether to apply the averaged
151 ///< projection filter (APF)
152 int denoise_strength; ///< strength of denoising in Wiener filter
154 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
155 ///< Wiener filter coefficients (postfilter)
156 int dc_level; ///< Predicted amount of DC noise, based
157 ///< on which a DC removal filter is used
159 int lsps; ///< number of LSPs per frame [10 or 16]
160 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
161 int lsp_def_mode; ///< defines different sets of LSP defaults
163 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
164 ///< per-frame (independent coding)
165 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
166 ///< per superframe (residual coding)
168 int min_pitch_val; ///< base value for pitch parsing code
169 int max_pitch_val; ///< max value + 1 for pitch parsing
170 int pitch_nbits; ///< number of bits used to specify the
171 ///< pitch value in the frame header
172 int block_pitch_nbits; ///< number of bits used to specify the
173 ///< first block's pitch value
174 int block_pitch_range; ///< range of the block pitch
175 int block_delta_pitch_nbits; ///< number of bits used to specify the
176 ///< delta pitch between this and the last
177 ///< block's pitch value, used in all but
179 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
180 ///< from -this to +this-1)
181 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
187 * @name Packet values specified in the packet header or related to a packet.
189 * A packet is considered to be a single unit of data provided to this
190 * decoder by the demuxer.
193 int spillover_nbits; ///< number of bits of the previous packet's
194 ///< last superframe preceding this
195 ///< packet's first full superframe (useful
196 ///< for re-synchronization also)
197 int has_residual_lsps; ///< if set, superframes contain one set of
198 ///< LSPs that cover all frames, encoded as
199 ///< independent and residual LSPs; if not
200 ///< set, each frame contains its own, fully
201 ///< independent, LSPs
202 int skip_bits_next; ///< number of bits to skip at the next call
203 ///< to #wmavoice_decode_packet() (since
204 ///< they're part of the previous superframe)
206 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
207 ///< cache for superframe data split over
208 ///< multiple packets
209 int sframe_cache_size; ///< set to >0 if we have data from an
210 ///< (incomplete) superframe from a previous
211 ///< packet that spilled over in the current
212 ///< packet; specifies the amount of bits in
214 PutBitContext pb; ///< bitstream writer for #sframe_cache
219 * @name Frame and superframe values
220 * Superframe and frame data - these can change from frame to frame,
221 * although some of them do in that case serve as a cache / history for
222 * the next frame or superframe.
225 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
227 int last_pitch_val; ///< pitch value of the previous frame
228 int last_acb_type; ///< frame type [0-2] of the previous frame
229 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
230 ///< << 16) / #MAX_FRAMESIZE
231 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
233 int aw_idx_is_ext; ///< whether the AW index was encoded in
234 ///< 8 bits (instead of 6)
235 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
236 ///< can apply the pulse, relative to the
237 ///< value in aw_first_pulse_off. The exact
238 ///< position of the first AW-pulse is within
239 ///< [pulse_off, pulse_off + this], and
240 ///< depends on bitstream values; [16 or 24]
241 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
242 ///< that this number can be negative (in
243 ///< which case it basically means "zero")
244 int aw_first_pulse_off[2]; ///< index of first sample to which to
245 ///< apply AW-pulses, or -0xff if unset
246 int aw_next_pulse_off_cache; ///< the position (relative to start of the
247 ///< second block) at which pulses should
248 ///< start to be positioned, serves as a
249 ///< cache for pitch-adaptive window pulses
252 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
253 ///< only used for comfort noise in #pRNG()
254 float gain_pred_err[6]; ///< cache for gain prediction
255 float excitation_history[MAX_SIGNAL_HISTORY];
256 ///< cache of the signal of previous
257 ///< superframes, used as a history for
258 ///< signal generation
259 float synth_history[MAX_LSPS]; ///< see #excitation_history
263 * @name Postfilter values
265 * Variables used for postfilter implementation, mostly history for
266 * smoothing and so on, and context variables for FFT/iFFT.
269 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
270 ///< postfilter (for denoise filter)
271 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
272 ///< transform, part of postfilter)
273 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
275 float postfilter_agc; ///< gain control memory, used in
276 ///< #adaptive_gain_control()
277 float dcf_mem[2]; ///< DC filter history
278 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
279 ///< zero filter output (i.e. excitation)
281 float denoise_filter_cache[MAX_FRAMESIZE];
282 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
283 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
284 ///< aligned buffer for LPC tilting
285 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
286 ///< aligned buffer for denoise coefficients
287 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
288 ///< aligned buffer for postfilter speech
296 * Set up the variable bit mode (VBM) tree from container extradata.
297 * @param gb bit I/O context.
298 * The bit context (s->gb) should be loaded with byte 23-46 of the
299 * container extradata (i.e. the ones containing the VBM tree).
300 * @param vbm_tree pointer to array to which the decoded VBM tree will be
302 * @return 0 on success, <0 on error.
304 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
306 int cntr[8] = { 0 }, n, res;
308 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
309 for (n = 0; n < 17; n++) {
310 res = get_bits(gb, 3);
311 if (cntr[res] > 3) // should be >= 3 + (res == 7))
313 vbm_tree[res * 3 + cntr[res]++] = n;
318 static av_cold void wmavoice_init_static_data(AVCodec *codec)
320 static const uint8_t bits[] = {
323 10, 10, 10, 12, 12, 12,
326 static const uint16_t codes[] = {
327 0x0000, 0x0001, 0x0002, // 00/01/10
328 0x000c, 0x000d, 0x000e, // 11+00/01/10
329 0x003c, 0x003d, 0x003e, // 1111+00/01/10
330 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
331 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
332 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
333 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
336 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
337 bits, 1, 1, codes, 2, 2, 132);
341 * Set up decoder with parameters from demuxer (extradata etc.).
343 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
345 int n, flags, pitch_range, lsp16_flag;
346 WMAVoiceContext *s = ctx->priv_data;
350 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
351 * - byte 19-22: flags field (annoyingly in LE; see below for known
353 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
356 if (ctx->extradata_size != 46) {
357 av_log(ctx, AV_LOG_ERROR,
358 "Invalid extradata size %d (should be 46)\n",
359 ctx->extradata_size);
360 return AVERROR_INVALIDDATA;
362 flags = AV_RL32(ctx->extradata + 18);
363 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
364 s->do_apf = flags & 0x1;
366 ff_rdft_init(&s->rdft, 7, DFT_R2C);
367 ff_rdft_init(&s->irdft, 7, IDFT_C2R);
368 ff_dct_init(&s->dct, 6, DCT_I);
369 ff_dct_init(&s->dst, 6, DST_I);
371 ff_sine_window_init(s->cos, 256);
372 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
373 for (n = 0; n < 255; n++) {
374 s->sin[n] = -s->sin[510 - n];
375 s->cos[510 - n] = s->cos[n];
378 s->denoise_strength = (flags >> 2) & 0xF;
379 if (s->denoise_strength >= 12) {
380 av_log(ctx, AV_LOG_ERROR,
381 "Invalid denoise filter strength %d (max=11)\n",
382 s->denoise_strength);
383 return AVERROR_INVALIDDATA;
385 s->denoise_tilt_corr = !!(flags & 0x40);
386 s->dc_level = (flags >> 7) & 0xF;
387 s->lsp_q_mode = !!(flags & 0x2000);
388 s->lsp_def_mode = !!(flags & 0x4000);
389 lsp16_flag = flags & 0x1000;
392 s->frame_lsp_bitsize = 34;
393 s->sframe_lsp_bitsize = 60;
396 s->frame_lsp_bitsize = 24;
397 s->sframe_lsp_bitsize = 48;
399 for (n = 0; n < s->lsps; n++)
400 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
402 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
403 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
404 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
405 return AVERROR_INVALIDDATA;
408 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
409 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
410 pitch_range = s->max_pitch_val - s->min_pitch_val;
411 if (pitch_range <= 0) {
412 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
413 return AVERROR_INVALIDDATA;
415 s->pitch_nbits = av_ceil_log2(pitch_range);
416 s->last_pitch_val = 40;
417 s->last_acb_type = ACB_TYPE_NONE;
418 s->history_nsamples = s->max_pitch_val + 8;
420 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
421 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
422 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
424 av_log(ctx, AV_LOG_ERROR,
425 "Unsupported samplerate %d (min=%d, max=%d)\n",
426 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
428 return AVERROR(ENOSYS);
431 s->block_conv_table[0] = s->min_pitch_val;
432 s->block_conv_table[1] = (pitch_range * 25) >> 6;
433 s->block_conv_table[2] = (pitch_range * 44) >> 6;
434 s->block_conv_table[3] = s->max_pitch_val - 1;
435 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
436 if (s->block_delta_pitch_hrange <= 0) {
437 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
438 return AVERROR_INVALIDDATA;
440 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
441 s->block_pitch_range = s->block_conv_table[2] +
442 s->block_conv_table[3] + 1 +
443 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
444 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
447 ctx->channel_layout = AV_CH_LAYOUT_MONO;
448 ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
454 * @name Postfilter functions
455 * Postfilter functions (gain control, wiener denoise filter, DC filter,
456 * kalman smoothening, plus surrounding code to wrap it)
460 * Adaptive gain control (as used in postfilter).
462 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
463 * that the energy here is calculated using sum(abs(...)), whereas the
464 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
466 * @param out output buffer for filtered samples
467 * @param in input buffer containing the samples as they are after the
468 * postfilter steps so far
469 * @param speech_synth input buffer containing speech synth before postfilter
470 * @param size input buffer size
471 * @param alpha exponential filter factor
472 * @param gain_mem pointer to filter memory (single float)
474 static void adaptive_gain_control(float *out, const float *in,
475 const float *speech_synth,
476 int size, float alpha, float *gain_mem)
479 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
480 float mem = *gain_mem;
482 for (i = 0; i < size; i++) {
483 speech_energy += fabsf(speech_synth[i]);
484 postfilter_energy += fabsf(in[i]);
486 gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
488 for (i = 0; i < size; i++) {
489 mem = alpha * mem + gain_scale_factor;
490 out[i] = in[i] * mem;
497 * Kalman smoothing function.
499 * This function looks back pitch +/- 3 samples back into history to find
500 * the best fitting curve (that one giving the optimal gain of the two
501 * signals, i.e. the highest dot product between the two), and then
502 * uses that signal history to smoothen the output of the speech synthesis
505 * @param s WMA Voice decoding context
506 * @param pitch pitch of the speech signal
507 * @param in input speech signal
508 * @param out output pointer for smoothened signal
509 * @param size input/output buffer size
511 * @returns -1 if no smoothening took place, e.g. because no optimal
512 * fit could be found, or 0 on success.
514 static int kalman_smoothen(WMAVoiceContext *s, int pitch,
515 const float *in, float *out, int size)
518 float optimal_gain = 0, dot;
519 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
520 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
521 *best_hist_ptr = NULL;
523 /* find best fitting point in history */
525 dot = avpriv_scalarproduct_float_c(in, ptr, size);
526 if (dot > optimal_gain) {
530 } while (--ptr >= end);
532 if (optimal_gain <= 0)
534 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
535 if (dot <= 0) // would be 1.0
538 if (optimal_gain <= dot) {
539 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
543 /* actual smoothing */
544 for (n = 0; n < size; n++)
545 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
551 * Get the tilt factor of a formant filter from its transfer function
552 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
553 * but somehow (??) it does a speech synthesis filter in the
554 * middle, which is missing here
556 * @param lpcs LPC coefficients
557 * @param n_lpcs Size of LPC buffer
558 * @returns the tilt factor
560 static float tilt_factor(const float *lpcs, int n_lpcs)
564 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
565 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
571 * Derive denoise filter coefficients (in real domain) from the LPCs.
573 static void calc_input_response(WMAVoiceContext *s, float *lpcs,
574 int fcb_type, float *coeffs, int remainder)
576 float last_coeff, min = 15.0, max = -15.0;
577 float irange, angle_mul, gain_mul, range, sq;
580 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
581 s->rdft.rdft_calc(&s->rdft, lpcs);
582 #define log_range(var, assign) do { \
583 float tmp = log10f(assign); var = tmp; \
584 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
586 log_range(last_coeff, lpcs[1] * lpcs[1]);
587 for (n = 1; n < 64; n++)
588 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
589 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
590 log_range(lpcs[0], lpcs[0] * lpcs[0]);
593 lpcs[64] = last_coeff;
595 /* Now, use this spectrum to pick out these frequencies with higher
596 * (relative) power/energy (which we then take to be "not noise"),
597 * and set up a table (still in lpc[]) of (relative) gains per frequency.
598 * These frequencies will be maintained, while others ("noise") will be
599 * decreased in the filter output. */
600 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
601 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
603 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
604 for (n = 0; n <= 64; n++) {
607 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
608 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
609 lpcs[n] = angle_mul * pwr;
611 /* 70.57 =~ 1/log10(1.0331663) */
612 idx = (pwr * gain_mul - 0.0295) * 70.570526123;
613 if (idx > 127) { // fall back if index falls outside table range
614 coeffs[n] = wmavoice_energy_table[127] *
615 powf(1.0331663, idx - 127);
617 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
620 /* calculate the Hilbert transform of the gains, which we do (since this
621 * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
622 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
623 * "moment" of the LPCs in this filter. */
624 s->dct.dct_calc(&s->dct, lpcs);
625 s->dst.dct_calc(&s->dst, lpcs);
627 /* Split out the coefficient indexes into phase/magnitude pairs */
628 idx = 255 + av_clip(lpcs[64], -255, 255);
629 coeffs[0] = coeffs[0] * s->cos[idx];
630 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
631 last_coeff = coeffs[64] * s->cos[idx];
633 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
634 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
635 coeffs[n * 2] = coeffs[n] * s->cos[idx];
639 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
640 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
641 coeffs[n * 2] = coeffs[n] * s->cos[idx];
643 coeffs[1] = last_coeff;
645 /* move into real domain */
646 s->irdft.rdft_calc(&s->irdft, coeffs);
648 /* tilt correction and normalize scale */
649 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
650 if (s->denoise_tilt_corr) {
653 coeffs[remainder - 1] = 0;
654 ff_tilt_compensation(&tilt_mem,
655 -1.8 * tilt_factor(coeffs, remainder - 1),
658 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
660 for (n = 0; n < remainder; n++)
665 * This function applies a Wiener filter on the (noisy) speech signal as
666 * a means to denoise it.
668 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
669 * - using this power spectrum, calculate (for each frequency) the Wiener
670 * filter gain, which depends on the frequency power and desired level
671 * of noise subtraction (when set too high, this leads to artifacts)
672 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
674 * - by doing a phase shift, calculate the Hilbert transform of this array
675 * of per-frequency filter-gains to get the filtering coefficients;
676 * - smoothen/normalize/de-tilt these filter coefficients as desired;
677 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
678 * to get the denoised speech signal;
679 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
680 * the frame boundary) are saved and applied to subsequent frames by an
681 * overlap-add method (otherwise you get clicking-artifacts).
683 * @param s WMA Voice decoding context
684 * @param fcb_type Frame (codebook) type
685 * @param synth_pf input: the noisy speech signal, output: denoised speech
686 * data; should be 16-byte aligned (for ASM purposes)
687 * @param size size of the speech data
688 * @param lpcs LPCs used to synthesize this frame's speech data
690 static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
691 float *synth_pf, int size,
694 int remainder, lim, n;
696 if (fcb_type != FCB_TYPE_SILENCE) {
697 float *tilted_lpcs = s->tilted_lpcs_pf,
698 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
700 tilted_lpcs[0] = 1.0;
701 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
702 memset(&tilted_lpcs[s->lsps + 1], 0,
703 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
704 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
705 tilted_lpcs, s->lsps + 2);
707 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
708 * size is applied to the next frame. All input beyond this is zero,
709 * and thus all output beyond this will go towards zero, hence we can
710 * limit to min(size-1, 127-size) as a performance consideration. */
711 remainder = FFMIN(127 - size, size - 1);
712 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
714 /* apply coefficients (in frequency spectrum domain), i.e. complex
715 * number multiplication */
716 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
717 s->rdft.rdft_calc(&s->rdft, synth_pf);
718 s->rdft.rdft_calc(&s->rdft, coeffs);
719 synth_pf[0] *= coeffs[0];
720 synth_pf[1] *= coeffs[1];
721 for (n = 1; n < 64; n++) {
722 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
723 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
724 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
726 s->irdft.rdft_calc(&s->irdft, synth_pf);
729 /* merge filter output with the history of previous runs */
730 if (s->denoise_filter_cache_size) {
731 lim = FFMIN(s->denoise_filter_cache_size, size);
732 for (n = 0; n < lim; n++)
733 synth_pf[n] += s->denoise_filter_cache[n];
734 s->denoise_filter_cache_size -= lim;
735 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
736 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
739 /* move remainder of filter output into a cache for future runs */
740 if (fcb_type != FCB_TYPE_SILENCE) {
741 lim = FFMIN(remainder, s->denoise_filter_cache_size);
742 for (n = 0; n < lim; n++)
743 s->denoise_filter_cache[n] += synth_pf[size + n];
744 if (lim < remainder) {
745 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
746 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
747 s->denoise_filter_cache_size = remainder;
753 * Averaging projection filter, the postfilter used in WMAVoice.
755 * This uses the following steps:
756 * - A zero-synthesis filter (generate excitation from synth signal)
757 * - Kalman smoothing on excitation, based on pitch
758 * - Re-synthesized smoothened output
759 * - Iterative Wiener denoise filter
760 * - Adaptive gain filter
763 * @param s WMAVoice decoding context
764 * @param synth Speech synthesis output (before postfilter)
765 * @param samples Output buffer for filtered samples
766 * @param size Buffer size of synth & samples
767 * @param lpcs Generated LPCs used for speech synthesis
768 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
769 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
770 * @param pitch Pitch of the input signal
772 static void postfilter(WMAVoiceContext *s, const float *synth,
773 float *samples, int size,
774 const float *lpcs, float *zero_exc_pf,
775 int fcb_type, int pitch)
777 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
778 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
779 *synth_filter_in = zero_exc_pf;
781 av_assert0(size <= MAX_FRAMESIZE / 2);
783 /* generate excitation from input signal */
784 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
786 if (fcb_type >= FCB_TYPE_AW_PULSES &&
787 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
788 synth_filter_in = synth_filter_in_buf;
790 /* re-synthesize speech after smoothening, and keep history */
791 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
792 synth_filter_in, size, s->lsps);
793 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
794 sizeof(synth_pf[0]) * s->lsps);
796 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
798 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
801 if (s->dc_level > 8) {
802 /* remove ultra-low frequency DC noise / highpass filter;
803 * coefficients are identical to those used in SIPR decoding,
804 * and very closely resemble those used in AMR-NB decoding. */
805 ff_acelp_apply_order_2_transfer_function(samples, samples,
806 (const float[2]) { -1.99997, 1.0 },
807 (const float[2]) { -1.9330735188, 0.93589198496 },
808 0.93980580475, s->dcf_mem, size);
817 * @param lsps output pointer to the array that will hold the LSPs
818 * @param num number of LSPs to be dequantized
819 * @param values quantized values, contains n_stages values
820 * @param sizes range (i.e. max value) of each quantized value
821 * @param n_stages number of dequantization runs
822 * @param table dequantization table to be used
823 * @param mul_q LSF multiplier
824 * @param base_q base (lowest) LSF values
826 static void dequant_lsps(double *lsps, int num,
827 const uint16_t *values,
828 const uint16_t *sizes,
829 int n_stages, const uint8_t *table,
831 const double *base_q)
835 memset(lsps, 0, num * sizeof(*lsps));
836 for (n = 0; n < n_stages; n++) {
837 const uint8_t *t_off = &table[values[n] * num];
838 double base = base_q[n], mul = mul_q[n];
840 for (m = 0; m < num; m++)
841 lsps[m] += base + mul * t_off[m];
843 table += sizes[n] * num;
848 * @name LSP dequantization routines
849 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
850 * @note we assume enough bits are available, caller should check.
851 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
852 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
856 * Parse 10 independently-coded LSPs.
858 static void dequant_lsp10i(GetBitContext *gb, double *lsps)
860 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
861 static const double mul_lsf[4] = {
862 5.2187144800e-3, 1.4626986422e-3,
863 9.6179549166e-4, 1.1325736225e-3
865 static const double base_lsf[4] = {
866 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
867 M_PI * -3.3486e-2, M_PI * -5.7408e-2
871 v[0] = get_bits(gb, 8);
872 v[1] = get_bits(gb, 6);
873 v[2] = get_bits(gb, 5);
874 v[3] = get_bits(gb, 5);
876 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
881 * Parse 10 independently-coded LSPs, and then derive the tables to
882 * generate LSPs for the other frames from them (residual coding).
884 static void dequant_lsp10r(GetBitContext *gb,
885 double *i_lsps, const double *old,
886 double *a1, double *a2, int q_mode)
888 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
889 static const double mul_lsf[3] = {
890 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
892 static const double base_lsf[3] = {
893 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
895 const float (*ipol_tab)[2][10] = q_mode ?
896 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
897 uint16_t interpol, v[3];
900 dequant_lsp10i(gb, i_lsps);
902 interpol = get_bits(gb, 5);
903 v[0] = get_bits(gb, 7);
904 v[1] = get_bits(gb, 6);
905 v[2] = get_bits(gb, 6);
907 for (n = 0; n < 10; n++) {
908 double delta = old[n] - i_lsps[n];
909 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
910 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
913 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
918 * Parse 16 independently-coded LSPs.
920 static void dequant_lsp16i(GetBitContext *gb, double *lsps)
922 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
923 static const double mul_lsf[5] = {
924 3.3439586280e-3, 6.9908173703e-4,
925 3.3216608306e-3, 1.0334960326e-3,
928 static const double base_lsf[5] = {
929 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
930 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
935 v[0] = get_bits(gb, 8);
936 v[1] = get_bits(gb, 6);
937 v[2] = get_bits(gb, 7);
938 v[3] = get_bits(gb, 6);
939 v[4] = get_bits(gb, 7);
941 dequant_lsps( lsps, 5, v, vec_sizes, 2,
942 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
943 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
944 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
945 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
946 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
950 * Parse 16 independently-coded LSPs, and then derive the tables to
951 * generate LSPs for the other frames from them (residual coding).
953 static void dequant_lsp16r(GetBitContext *gb,
954 double *i_lsps, const double *old,
955 double *a1, double *a2, int q_mode)
957 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
958 static const double mul_lsf[3] = {
959 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
961 static const double base_lsf[3] = {
962 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
964 const float (*ipol_tab)[2][16] = q_mode ?
965 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
966 uint16_t interpol, v[3];
969 dequant_lsp16i(gb, i_lsps);
971 interpol = get_bits(gb, 5);
972 v[0] = get_bits(gb, 7);
973 v[1] = get_bits(gb, 7);
974 v[2] = get_bits(gb, 7);
976 for (n = 0; n < 16; n++) {
977 double delta = old[n] - i_lsps[n];
978 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
979 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
982 dequant_lsps( a2, 10, v, vec_sizes, 1,
983 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
984 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
985 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
986 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
987 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
992 * @name Pitch-adaptive window coding functions
993 * The next few functions are for pitch-adaptive window coding.
997 * Parse the offset of the first pitch-adaptive window pulses, and
998 * the distribution of pulses between the two blocks in this frame.
999 * @param s WMA Voice decoding context private data
1000 * @param gb bit I/O context
1001 * @param pitch pitch for each block in this frame
1003 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
1006 static const int16_t start_offset[94] = {
1007 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1008 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1009 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1010 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1011 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1012 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1013 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1014 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1018 /* position of pulse */
1019 s->aw_idx_is_ext = 0;
1020 if ((bits = get_bits(gb, 6)) >= 54) {
1021 s->aw_idx_is_ext = 1;
1022 bits += (bits - 54) * 3 + get_bits(gb, 2);
1025 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1026 * the distribution of the pulses in each block contained in this frame. */
1027 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1028 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1029 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1030 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1031 offset += s->aw_n_pulses[0] * pitch[0];
1032 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1033 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1035 /* if continuing from a position before the block, reset position to
1036 * start of block (when corrected for the range over which it can be
1037 * spread in aw_pulse_set1()). */
1038 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1039 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1040 s->aw_first_pulse_off[1] -= pitch[1];
1041 if (start_offset[bits] < 0)
1042 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1043 s->aw_first_pulse_off[0] -= pitch[0];
1048 * Apply second set of pitch-adaptive window pulses.
1049 * @param s WMA Voice decoding context private data
1050 * @param gb bit I/O context
1051 * @param block_idx block index in frame [0, 1]
1052 * @param fcb structure containing fixed codebook vector info
1053 * @return -1 on error, 0 otherwise
1055 static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1056 int block_idx, AMRFixed *fcb)
1058 uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
1059 uint16_t *use_mask = use_mask_mem + 2;
1060 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1061 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1062 * of idx are the position of the bit within a particular item in the
1063 * array (0 being the most significant bit, and 15 being the least
1064 * significant bit), and the remainder (>> 4) is the index in the
1065 * use_mask[]-array. This is faster and uses less memory than using a
1066 * 80-byte/80-int array. */
1067 int pulse_off = s->aw_first_pulse_off[block_idx],
1068 pulse_start, n, idx, range, aidx, start_off = 0;
1070 /* set offset of first pulse to within this block */
1071 if (s->aw_n_pulses[block_idx] > 0)
1072 while (pulse_off + s->aw_pulse_range < 1)
1073 pulse_off += fcb->pitch_lag;
1075 /* find range per pulse */
1076 if (s->aw_n_pulses[0] > 0) {
1077 if (block_idx == 0) {
1079 } else /* block_idx = 1 */ {
1081 if (s->aw_n_pulses[block_idx] > 0)
1082 pulse_off = s->aw_next_pulse_off_cache;
1086 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1088 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1089 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1090 * we exclude that range from being pulsed again in this function. */
1091 memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
1092 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
1093 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1094 if (s->aw_n_pulses[block_idx] > 0)
1095 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1096 int excl_range = s->aw_pulse_range; // always 16 or 24
1097 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1098 int first_sh = 16 - (idx & 15);
1099 *use_mask_ptr++ &= 0xFFFFu << first_sh;
1100 excl_range -= first_sh;
1101 if (excl_range >= 16) {
1102 *use_mask_ptr++ = 0;
1103 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1105 *use_mask_ptr &= 0xFFFF >> excl_range;
1108 /* find the 'aidx'th offset that is not excluded */
1109 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1110 for (n = 0; n <= aidx; pulse_start++) {
1111 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1112 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1113 if (use_mask[0]) idx = 0x0F;
1114 else if (use_mask[1]) idx = 0x1F;
1115 else if (use_mask[2]) idx = 0x2F;
1116 else if (use_mask[3]) idx = 0x3F;
1117 else if (use_mask[4]) idx = 0x4F;
1119 idx -= av_log2_16bit(use_mask[idx >> 4]);
1121 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1122 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1128 fcb->x[fcb->n] = start_off;
1129 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1132 /* set offset for next block, relative to start of that block */
1133 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1134 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1139 * Apply first set of pitch-adaptive window pulses.
1140 * @param s WMA Voice decoding context private data
1141 * @param gb bit I/O context
1142 * @param block_idx block index in frame [0, 1]
1143 * @param fcb storage location for fixed codebook pulse info
1145 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1146 int block_idx, AMRFixed *fcb)
1148 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1151 if (s->aw_n_pulses[block_idx] > 0) {
1152 int n, v_mask, i_mask, sh, n_pulses;
1154 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1159 } else { // 4 pulses, 1:sign + 2:index each
1166 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1167 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1168 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1169 s->aw_first_pulse_off[block_idx];
1170 while (fcb->x[fcb->n] < 0)
1171 fcb->x[fcb->n] += fcb->pitch_lag;
1172 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1176 int num2 = (val & 0x1FF) >> 1, delta, idx;
1178 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1179 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1180 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1181 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1182 v = (val & 0x200) ? -1.0 : 1.0;
1184 fcb->no_repeat_mask |= 3 << fcb->n;
1185 fcb->x[fcb->n] = idx - delta;
1187 fcb->x[fcb->n + 1] = idx;
1188 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1196 * Generate a random number from frame_cntr and block_idx, which will lief
1197 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1198 * table of size 1000 of which you want to read block_size entries).
1200 * @param frame_cntr current frame number
1201 * @param block_num current block index
1202 * @param block_size amount of entries we want to read from a table
1203 * that has 1000 entries
1204 * @return a (non-)random number in the [0, 1000 - block_size] range.
1206 static int pRNG(int frame_cntr, int block_num, int block_size)
1208 /* array to simplify the calculation of z:
1209 * y = (x % 9) * 5 + 6;
1210 * z = (49995 * x) / y;
1211 * Since y only has 9 values, we can remove the division by using a
1212 * LUT and using FASTDIV-style divisions. For each of the 9 values
1213 * of y, we can rewrite z as:
1214 * z = x * (49995 / y) + x * ((49995 % y) / y)
1215 * In this table, each col represents one possible value of y, the
1216 * first number is 49995 / y, and the second is the FASTDIV variant
1217 * of 49995 % y / y. */
1218 static const unsigned int div_tbl[9][2] = {
1219 { 8332, 3 * 715827883U }, // y = 6
1220 { 4545, 0 * 390451573U }, // y = 11
1221 { 3124, 11 * 268435456U }, // y = 16
1222 { 2380, 15 * 204522253U }, // y = 21
1223 { 1922, 23 * 165191050U }, // y = 26
1224 { 1612, 23 * 138547333U }, // y = 31
1225 { 1388, 27 * 119304648U }, // y = 36
1226 { 1219, 16 * 104755300U }, // y = 41
1227 { 1086, 39 * 93368855U } // y = 46
1229 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1230 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1231 // so this is effectively a modulo (%)
1232 y = x - 9 * MULH(477218589, x); // x % 9
1233 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1234 // z = x * 49995 / (y * 5 + 6)
1235 return z % (1000 - block_size);
1239 * Parse hardcoded signal for a single block.
1240 * @note see #synth_block().
1242 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1243 int block_idx, int size,
1244 const struct frame_type_desc *frame_desc,
1250 av_assert0(size <= MAX_FRAMESIZE);
1252 /* Set the offset from which we start reading wmavoice_std_codebook */
1253 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1254 r_idx = pRNG(s->frame_cntr, block_idx, size);
1255 gain = s->silence_gain;
1256 } else /* FCB_TYPE_HARDCODED */ {
1257 r_idx = get_bits(gb, 8);
1258 gain = wmavoice_gain_universal[get_bits(gb, 6)];
1261 /* Clear gain prediction parameters */
1262 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1264 /* Apply gain to hardcoded codebook and use that as excitation signal */
1265 for (n = 0; n < size; n++)
1266 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1270 * Parse FCB/ACB signal for a single block.
1271 * @note see #synth_block().
1273 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1274 int block_idx, int size,
1275 int block_pitch_sh2,
1276 const struct frame_type_desc *frame_desc,
1279 static const float gain_coeff[6] = {
1280 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1282 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1283 int n, idx, gain_weight;
1286 av_assert0(size <= MAX_FRAMESIZE / 2);
1287 memset(pulses, 0, sizeof(*pulses) * size);
1289 fcb.pitch_lag = block_pitch_sh2 >> 2;
1290 fcb.pitch_fac = 1.0;
1291 fcb.no_repeat_mask = 0;
1294 /* For the other frame types, this is where we apply the innovation
1295 * (fixed) codebook pulses of the speech signal. */
1296 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1297 aw_pulse_set1(s, gb, block_idx, &fcb);
1298 if (aw_pulse_set2(s, gb, block_idx, &fcb)) {
1299 /* Conceal the block with silence and return.
1300 * Skip the correct amount of bits to read the next
1301 * block from the correct offset. */
1302 int r_idx = pRNG(s->frame_cntr, block_idx, size);
1304 for (n = 0; n < size; n++)
1306 wmavoice_std_codebook[r_idx + n] * s->silence_gain;
1307 skip_bits(gb, 7 + 1);
1310 } else /* FCB_TYPE_EXC_PULSES */ {
1311 int offset_nbits = 5 - frame_desc->log_n_blocks;
1313 fcb.no_repeat_mask = -1;
1314 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1315 * (instead of double) for a subset of pulses */
1316 for (n = 0; n < 5; n++) {
1320 sign = get_bits1(gb) ? 1.0 : -1.0;
1321 pos1 = get_bits(gb, offset_nbits);
1322 fcb.x[fcb.n] = n + 5 * pos1;
1323 fcb.y[fcb.n++] = sign;
1324 if (n < frame_desc->dbl_pulses) {
1325 pos2 = get_bits(gb, offset_nbits);
1326 fcb.x[fcb.n] = n + 5 * pos2;
1327 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1331 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1333 /* Calculate gain for adaptive & fixed codebook signal.
1334 * see ff_amr_set_fixed_gain(). */
1335 idx = get_bits(gb, 7);
1336 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
1338 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1339 acb_gain = wmavoice_gain_codebook_acb[idx];
1340 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1341 -2.9957322736 /* log(0.05) */,
1342 1.6094379124 /* log(5.0) */);
1344 gain_weight = 8 >> frame_desc->log_n_blocks;
1345 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1346 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1347 for (n = 0; n < gain_weight; n++)
1348 s->gain_pred_err[n] = pred_err;
1350 /* Calculation of adaptive codebook */
1351 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1353 for (n = 0; n < size; n += len) {
1355 int abs_idx = block_idx * size + n;
1356 int pitch_sh16 = (s->last_pitch_val << 16) +
1357 s->pitch_diff_sh16 * abs_idx;
1358 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1359 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1360 idx = idx_sh16 >> 16;
1361 if (s->pitch_diff_sh16) {
1362 if (s->pitch_diff_sh16 > 0) {
1363 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1365 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1366 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1371 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1372 wmavoice_ipol1_coeffs, 17,
1375 } else /* ACB_TYPE_HAMMING */ {
1376 int block_pitch = block_pitch_sh2 >> 2;
1377 idx = block_pitch_sh2 & 3;
1379 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1380 wmavoice_ipol2_coeffs, 4,
1383 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1384 sizeof(float) * size);
1387 /* Interpolate ACB/FCB and use as excitation signal */
1388 ff_weighted_vector_sumf(excitation, excitation, pulses,
1389 acb_gain, fcb_gain, size);
1393 * Parse data in a single block.
1394 * @note we assume enough bits are available, caller should check.
1396 * @param s WMA Voice decoding context private data
1397 * @param gb bit I/O context
1398 * @param block_idx index of the to-be-read block
1399 * @param size amount of samples to be read in this block
1400 * @param block_pitch_sh2 pitch for this block << 2
1401 * @param lsps LSPs for (the end of) this frame
1402 * @param prev_lsps LSPs for the last frame
1403 * @param frame_desc frame type descriptor
1404 * @param excitation target memory for the ACB+FCB interpolated signal
1405 * @param synth target memory for the speech synthesis filter output
1406 * @return 0 on success, <0 on error.
1408 static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1409 int block_idx, int size,
1410 int block_pitch_sh2,
1411 const double *lsps, const double *prev_lsps,
1412 const struct frame_type_desc *frame_desc,
1413 float *excitation, float *synth)
1415 double i_lsps[MAX_LSPS];
1416 float lpcs[MAX_LSPS];
1420 if (frame_desc->acb_type == ACB_TYPE_NONE)
1421 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1423 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1424 frame_desc, excitation);
1426 /* convert interpolated LSPs to LPCs */
1427 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1428 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1429 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1430 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1432 /* Speech synthesis */
1433 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1437 * Synthesize output samples for a single frame.
1438 * @note we assume enough bits are available, caller should check.
1440 * @param ctx WMA Voice decoder context
1441 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1442 * @param frame_idx Frame number within superframe [0-2]
1443 * @param samples pointer to output sample buffer, has space for at least 160
1445 * @param lsps LSP array
1446 * @param prev_lsps array of previous frame's LSPs
1447 * @param excitation target buffer for excitation signal
1448 * @param synth target buffer for synthesized speech data
1449 * @return 0 on success, <0 on error.
1451 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1453 const double *lsps, const double *prev_lsps,
1454 float *excitation, float *synth)
1456 WMAVoiceContext *s = ctx->priv_data;
1457 int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val);
1458 int pitch[MAX_BLOCKS], av_uninit(last_block_pitch);
1460 /* Parse frame type ("frame header"), see frame_descs */
1461 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
1464 av_log(ctx, AV_LOG_ERROR,
1465 "Invalid frame type VLC code, skipping\n");
1466 return AVERROR_INVALIDDATA;
1469 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1471 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1472 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1473 /* Pitch is provided per frame, which is interpreted as the pitch of
1474 * the last sample of the last block of this frame. We can interpolate
1475 * the pitch of other blocks (and even pitch-per-sample) by gradually
1476 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1477 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1478 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1479 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1480 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1481 if (s->last_acb_type == ACB_TYPE_NONE ||
1482 20 * abs(cur_pitch_val - s->last_pitch_val) >
1483 (cur_pitch_val + s->last_pitch_val))
1484 s->last_pitch_val = cur_pitch_val;
1486 /* pitch per block */
1487 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1488 int fac = n * 2 + 1;
1490 pitch[n] = (MUL16(fac, cur_pitch_val) +
1491 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1492 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1495 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1496 s->pitch_diff_sh16 =
1497 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1500 /* Global gain (if silence) and pitch-adaptive window coordinates */
1501 switch (frame_descs[bd_idx].fcb_type) {
1502 case FCB_TYPE_SILENCE:
1503 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1505 case FCB_TYPE_AW_PULSES:
1506 aw_parse_coords(s, gb, pitch);
1510 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1513 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1514 switch (frame_descs[bd_idx].acb_type) {
1515 case ACB_TYPE_HAMMING: {
1516 /* Pitch is given per block. Per-block pitches are encoded as an
1517 * absolute value for the first block, and then delta values
1518 * relative to this value) for all subsequent blocks. The scale of
1519 * this pitch value is semi-logaritmic compared to its use in the
1520 * decoder, so we convert it to normal scale also. */
1522 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1523 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1524 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1527 block_pitch = get_bits(gb, s->block_pitch_nbits);
1529 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1530 get_bits(gb, s->block_delta_pitch_nbits);
1531 /* Convert last_ so that any next delta is within _range */
1532 last_block_pitch = av_clip(block_pitch,
1533 s->block_delta_pitch_hrange,
1534 s->block_pitch_range -
1535 s->block_delta_pitch_hrange);
1537 /* Convert semi-log-style scale back to normal scale */
1538 if (block_pitch < t1) {
1539 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1542 if (block_pitch < t2) {
1544 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1547 if (block_pitch < t3) {
1549 (s->block_conv_table[2] + block_pitch) << 2;
1551 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1554 pitch[n] = bl_pitch_sh2 >> 2;
1558 case ACB_TYPE_ASYMMETRIC: {
1559 bl_pitch_sh2 = pitch[n] << 2;
1563 default: // ACB_TYPE_NONE has no pitch
1568 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1569 lsps, prev_lsps, &frame_descs[bd_idx],
1570 &excitation[n * block_nsamples],
1571 &synth[n * block_nsamples]);
1574 /* Averaging projection filter, if applicable. Else, just copy samples
1575 * from synthesis buffer */
1577 double i_lsps[MAX_LSPS];
1578 float lpcs[MAX_LSPS];
1580 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1581 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1582 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1583 postfilter(s, synth, samples, 80, lpcs,
1584 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1585 frame_descs[bd_idx].fcb_type, pitch[0]);
1587 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1588 i_lsps[n] = cos(lsps[n]);
1589 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1590 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1591 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1592 frame_descs[bd_idx].fcb_type, pitch[0]);
1594 memcpy(samples, synth, 160 * sizeof(synth[0]));
1596 /* Cache values for next frame */
1598 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1599 s->last_acb_type = frame_descs[bd_idx].acb_type;
1600 switch (frame_descs[bd_idx].acb_type) {
1602 s->last_pitch_val = 0;
1604 case ACB_TYPE_ASYMMETRIC:
1605 s->last_pitch_val = cur_pitch_val;
1607 case ACB_TYPE_HAMMING:
1608 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1616 * Ensure minimum value for first item, maximum value for last value,
1617 * proper spacing between each value and proper ordering.
1619 * @param lsps array of LSPs
1620 * @param num size of LSP array
1622 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1623 * useful to put in a generic location later on. Parts are also
1624 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1625 * which is in float.
1627 static void stabilize_lsps(double *lsps, int num)
1631 /* set minimum value for first, maximum value for last and minimum
1632 * spacing between LSF values.
1633 * Very similar to ff_set_min_dist_lsf(), but in double. */
1634 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1635 for (n = 1; n < num; n++)
1636 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1637 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1639 /* reorder (looks like one-time / non-recursed bubblesort).
1640 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1641 for (n = 1; n < num; n++) {
1642 if (lsps[n] < lsps[n - 1]) {
1643 for (m = 1; m < num; m++) {
1644 double tmp = lsps[m];
1645 for (l = m - 1; l >= 0; l--) {
1646 if (lsps[l] <= tmp) break;
1647 lsps[l + 1] = lsps[l];
1657 * Test if there's enough bits to read 1 superframe.
1659 * @param orig_gb bit I/O context used for reading. This function
1660 * does not modify the state of the bitreader; it
1661 * only uses it to copy the current stream position
1662 * @param s WMA Voice decoding context private data
1663 * @return < 0 on error, 1 on not enough bits or 0 if OK.
1665 static int check_bits_for_superframe(GetBitContext *orig_gb,
1668 GetBitContext s_gb, *gb = &s_gb;
1669 int n, need_bits, bd_idx;
1670 const struct frame_type_desc *frame_desc;
1672 /* initialize a copy */
1673 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
1674 skip_bits_long(gb, get_bits_count(orig_gb));
1675 av_assert1(get_bits_left(gb) == get_bits_left(orig_gb));
1677 /* superframe header */
1678 if (get_bits_left(gb) < 14)
1681 return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe
1682 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe
1683 if (s->has_residual_lsps) { // residual LSPs (for all frames)
1684 if (get_bits_left(gb) < s->sframe_lsp_bitsize)
1686 skip_bits_long(gb, s->sframe_lsp_bitsize);
1690 for (n = 0; n < MAX_FRAMES; n++) {
1691 int aw_idx_is_ext = 0;
1693 if (!s->has_residual_lsps) { // independent LSPs (per-frame)
1694 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
1695 skip_bits_long(gb, s->frame_lsp_bitsize);
1697 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
1699 return AVERROR_INVALIDDATA; // invalid frame type VLC code
1700 frame_desc = &frame_descs[bd_idx];
1701 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1702 if (get_bits_left(gb) < s->pitch_nbits)
1704 skip_bits_long(gb, s->pitch_nbits);
1706 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1708 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1709 int tmp = get_bits(gb, 6);
1717 if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
1718 need_bits = s->block_pitch_nbits +
1719 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
1720 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1721 need_bits = 2 * !aw_idx_is_ext;
1724 need_bits += frame_desc->frame_size;
1725 if (get_bits_left(gb) < need_bits)
1727 skip_bits_long(gb, need_bits);
1734 * Synthesize output samples for a single superframe. If we have any data
1735 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1738 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1739 * to give a total of 480 samples per frame. See #synth_frame() for frame
1740 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1741 * (if these are globally specified for all frames (residually); they can
1742 * also be specified individually per-frame. See the s->has_residual_lsps
1743 * option), and can specify the number of samples encoded in this superframe
1744 * (if less than 480), usually used to prevent blanks at track boundaries.
1746 * @param ctx WMA Voice decoder context
1747 * @return 0 on success, <0 on error or 1 if there was not enough data to
1748 * fully parse the superframe
1750 static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
1753 WMAVoiceContext *s = ctx->priv_data;
1754 GetBitContext *gb = &s->gb, s_gb;
1755 int n, res, n_samples = 480;
1756 double lsps[MAX_FRAMES][MAX_LSPS];
1757 const double *mean_lsf = s->lsps == 16 ?
1758 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1759 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1760 float synth[MAX_LSPS + MAX_SFRAMESIZE];
1763 memcpy(synth, s->synth_history,
1764 s->lsps * sizeof(*synth));
1765 memcpy(excitation, s->excitation_history,
1766 s->history_nsamples * sizeof(*excitation));
1768 if (s->sframe_cache_size > 0) {
1770 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1771 s->sframe_cache_size = 0;
1774 if ((res = check_bits_for_superframe(gb, s)) == 1) {
1780 /* First bit is speech/music bit, it differentiates between WMAVoice
1781 * speech samples (the actual codec) and WMAVoice music samples, which
1782 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1784 if (!get_bits1(gb)) {
1785 avpriv_request_sample(ctx, "WMAPro-in-WMAVoice");
1786 return AVERROR_PATCHWELCOME;
1789 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1790 if (get_bits1(gb)) {
1791 if ((n_samples = get_bits(gb, 12)) > 480) {
1792 av_log(ctx, AV_LOG_ERROR,
1793 "Superframe encodes >480 samples (%d), not allowed\n",
1795 return AVERROR_INVALIDDATA;
1798 /* Parse LSPs, if global for the superframe (can also be per-frame). */
1799 if (s->has_residual_lsps) {
1800 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1802 for (n = 0; n < s->lsps; n++)
1803 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1805 if (s->lsps == 10) {
1806 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1807 } else /* s->lsps == 16 */
1808 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1810 for (n = 0; n < s->lsps; n++) {
1811 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
1812 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1813 lsps[2][n] += mean_lsf[n];
1815 for (n = 0; n < 3; n++)
1816 stabilize_lsps(lsps[n], s->lsps);
1819 /* get output buffer */
1820 frame->nb_samples = 480;
1821 if ((res = ff_get_buffer(ctx, frame, 0)) < 0)
1823 frame->nb_samples = n_samples;
1824 samples = (float *)frame->data[0];
1826 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1827 for (n = 0; n < 3; n++) {
1828 if (!s->has_residual_lsps) {
1831 if (s->lsps == 10) {
1832 dequant_lsp10i(gb, lsps[n]);
1833 } else /* s->lsps == 16 */
1834 dequant_lsp16i(gb, lsps[n]);
1836 for (m = 0; m < s->lsps; m++)
1837 lsps[n][m] += mean_lsf[m];
1838 stabilize_lsps(lsps[n], s->lsps);
1841 if ((res = synth_frame(ctx, gb, n,
1842 &samples[n * MAX_FRAMESIZE],
1843 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1844 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1845 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
1851 /* Statistics? FIXME - we don't check for length, a slight overrun
1852 * will be caught by internal buffer padding, and anything else
1853 * will be skipped, not read. */
1854 if (get_bits1(gb)) {
1855 res = get_bits(gb, 4);
1856 skip_bits(gb, 10 * (res + 1));
1861 /* Update history */
1862 memcpy(s->prev_lsps, lsps[2],
1863 s->lsps * sizeof(*s->prev_lsps));
1864 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1865 s->lsps * sizeof(*synth));
1866 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1867 s->history_nsamples * sizeof(*excitation));
1869 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1870 s->history_nsamples * sizeof(*s->zero_exc_pf));
1876 * Parse the packet header at the start of each packet (input data to this
1879 * @param s WMA Voice decoding context private data
1880 * @return 1 if not enough bits were available, or 0 on success.
1882 static int parse_packet_header(WMAVoiceContext *s)
1884 GetBitContext *gb = &s->gb;
1887 if (get_bits_left(gb) < 11)
1889 skip_bits(gb, 4); // packet sequence number
1890 s->has_residual_lsps = get_bits1(gb);
1892 res = get_bits(gb, 6); // number of superframes per packet
1893 // (minus first one if there is spillover)
1894 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
1896 } while (res == 0x3F);
1897 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
1903 * Copy (unaligned) bits from gb/data/size to pb.
1905 * @param pb target buffer to copy bits into
1906 * @param data source buffer to copy bits from
1907 * @param size size of the source data, in bytes
1908 * @param gb bit I/O context specifying the current position in the source.
1909 * data. This function might use this to align the bit position to
1910 * a whole-byte boundary before calling #avpriv_copy_bits() on aligned
1912 * @param nbits the amount of bits to copy from source to target
1914 * @note after calling this function, the current position in the input bit
1915 * I/O context is undefined.
1917 static void copy_bits(PutBitContext *pb,
1918 const uint8_t *data, int size,
1919 GetBitContext *gb, int nbits)
1921 int rmn_bytes, rmn_bits;
1923 rmn_bits = rmn_bytes = get_bits_left(gb);
1924 if (rmn_bits < nbits)
1926 if (nbits > pb->size_in_bits - put_bits_count(pb))
1928 rmn_bits &= 7; rmn_bytes >>= 3;
1929 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1930 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1931 avpriv_copy_bits(pb, data + size - rmn_bytes,
1932 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1936 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1937 * and we expect that the demuxer / application provides it to us as such
1938 * (else you'll probably get garbage as output). Every packet has a size of
1939 * ctx->block_align bytes, starts with a packet header (see
1940 * #parse_packet_header()), and then a series of superframes. Superframe
1941 * boundaries may exceed packets, i.e. superframes can split data over
1942 * multiple (two) packets.
1944 * For more information about frames, see #synth_superframe().
1946 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1947 int *got_frame_ptr, AVPacket *avpkt)
1949 WMAVoiceContext *s = ctx->priv_data;
1950 GetBitContext *gb = &s->gb;
1953 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1954 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1955 * feeds us ASF packets, which may concatenate multiple "codec" packets
1956 * in a single "muxer" packet, so we artificially emulate that by
1957 * capping the packet size at ctx->block_align. */
1958 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1963 init_get_bits(&s->gb, avpkt->data, size << 3);
1965 /* size == ctx->block_align is used to indicate whether we are dealing with
1966 * a new packet or a packet of which we already read the packet header
1968 if (size == ctx->block_align) { // new packet header
1969 if ((res = parse_packet_header(s)) < 0)
1972 /* If the packet header specifies a s->spillover_nbits, then we want
1973 * to push out all data of the previous packet (+ spillover) before
1974 * continuing to parse new superframes in the current packet. */
1975 if (s->spillover_nbits > 0) {
1976 if (s->sframe_cache_size > 0) {
1977 int cnt = get_bits_count(gb);
1978 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1979 flush_put_bits(&s->pb);
1980 s->sframe_cache_size += s->spillover_nbits;
1981 if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 &&
1983 cnt += s->spillover_nbits;
1984 s->skip_bits_next = cnt & 7;
1987 skip_bits_long (gb, s->spillover_nbits - cnt +
1988 get_bits_count(gb)); // resync
1990 skip_bits_long(gb, s->spillover_nbits); // resync
1992 } else if (s->skip_bits_next)
1993 skip_bits(gb, s->skip_bits_next);
1995 /* Try parsing superframes in current packet */
1996 s->sframe_cache_size = 0;
1997 s->skip_bits_next = 0;
1998 pos = get_bits_left(gb);
1999 if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
2001 } else if (*got_frame_ptr) {
2002 int cnt = get_bits_count(gb);
2003 s->skip_bits_next = cnt & 7;
2005 } else if ((s->sframe_cache_size = pos) > 0) {
2006 /* rewind bit reader to start of last (incomplete) superframe... */
2007 init_get_bits(gb, avpkt->data, size << 3);
2008 skip_bits_long(gb, (size << 3) - pos);
2009 av_assert1(get_bits_left(gb) == pos);
2011 /* ...and cache it for spillover in next packet */
2012 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
2013 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
2014 // FIXME bad - just copy bytes as whole and add use the
2015 // skip_bits_next field
2021 static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
2023 WMAVoiceContext *s = ctx->priv_data;
2026 ff_rdft_end(&s->rdft);
2027 ff_rdft_end(&s->irdft);
2028 ff_dct_end(&s->dct);
2029 ff_dct_end(&s->dst);
2035 static av_cold void wmavoice_flush(AVCodecContext *ctx)
2037 WMAVoiceContext *s = ctx->priv_data;
2040 s->postfilter_agc = 0;
2041 s->sframe_cache_size = 0;
2042 s->skip_bits_next = 0;
2043 for (n = 0; n < s->lsps; n++)
2044 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2045 memset(s->excitation_history, 0,
2046 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2047 memset(s->synth_history, 0,
2048 sizeof(*s->synth_history) * MAX_LSPS);
2049 memset(s->gain_pred_err, 0,
2050 sizeof(s->gain_pred_err));
2053 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2054 sizeof(*s->synth_filter_out_buf) * s->lsps);
2055 memset(s->dcf_mem, 0,
2056 sizeof(*s->dcf_mem) * 2);
2057 memset(s->zero_exc_pf, 0,
2058 sizeof(*s->zero_exc_pf) * s->history_nsamples);
2059 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2063 AVCodec ff_wmavoice_decoder = {
2065 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
2066 .type = AVMEDIA_TYPE_AUDIO,
2067 .id = AV_CODEC_ID_WMAVOICE,
2068 .priv_data_size = sizeof(WMAVoiceContext),
2069 .init = wmavoice_decode_init,
2070 .init_static_data = wmavoice_init_static_data,
2071 .close = wmavoice_decode_end,
2072 .decode = wmavoice_decode_packet,
2073 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
2074 .flush = wmavoice_flush,