2 * Windows Media Audio Voice decoder.
3 * Copyright (c) 2009 Ronald S. Bultje
5 * This file is part of Libav.
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
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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 Libav; 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>
28 #define UNCHECKED_BITSTREAM_READER 1
32 #include "libavutil/channel_layout.h"
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/mem.h"
39 #include "wmavoice_data.h"
40 #include "celp_filters.h"
41 #include "acelp_vectors.h"
42 #include "acelp_filters.h"
48 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame
49 #define MAX_LSPS 16 ///< maximum filter order
50 #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple
51 ///< of 16 for ASM input buffer alignment
52 #define MAX_FRAMES 3 ///< maximum number of frames per superframe
53 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame
54 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history
55 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES)
56 ///< maximum number of samples per superframe
57 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that
58 ///< was split over two packets
59 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration
62 * Frame type VLC coding.
64 static VLC frame_type_vlc;
67 * Adaptive codebook types.
70 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed)
71 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which
72 ///< we interpolate to get a per-sample pitch.
73 ///< Signal is generated using an asymmetric sinc
75 ///< @note see #wmavoice_ipol1_coeffs
76 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using
77 ///< a Hamming sinc window function
78 ///< @note see #wmavoice_ipol2_coeffs
82 * Fixed codebook types.
85 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence
86 ///< generated from a hardcoded (fixed) codebook
87 ///< with per-frame (low) gain values
88 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block
90 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals,
91 ///< used in particular for low-bitrate streams
92 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in
93 ///< combinations of either single pulses or
98 * Description of frame types.
100 static const struct frame_type_desc {
101 uint8_t n_blocks; ///< amount of blocks per frame (each block
102 ///< (contains 160/#n_blocks samples)
103 uint8_t log_n_blocks; ///< log2(#n_blocks)
104 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*)
105 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*)
106 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs
107 ///< (rather than just one single pulse)
108 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES
109 uint16_t frame_size; ///< the amount of bits that make up the block
110 ///< data (per frame)
111 } frame_descs[17] = {
112 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 },
113 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 },
114 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 },
115 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 },
116 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 },
117 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 },
118 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 },
119 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 },
120 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 },
121 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 },
122 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 },
123 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 },
124 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 },
125 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 },
126 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 },
127 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 },
128 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 }
132 * WMA Voice decoding context.
136 * @name Global values specified in the stream header / extradata or used all over.
140 GetBitContext gb; ///< packet bitreader. During decoder init,
141 ///< it contains the extradata from the
142 ///< demuxer. During decoding, it contains
144 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
146 int spillover_bitsize; ///< number of bits used to specify
147 ///< #spillover_nbits in the packet header
148 ///< = ceil(log2(ctx->block_align << 3))
149 int history_nsamples; ///< number of samples in history for signal
150 ///< prediction (through ACB)
152 /* postfilter specific values */
153 int do_apf; ///< whether to apply the averaged
154 ///< projection filter (APF)
155 int denoise_strength; ///< strength of denoising in Wiener filter
157 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
158 ///< Wiener filter coefficients (postfilter)
159 int dc_level; ///< Predicted amount of DC noise, based
160 ///< on which a DC removal filter is used
162 int lsps; ///< number of LSPs per frame [10 or 16]
163 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
164 int lsp_def_mode; ///< defines different sets of LSP defaults
166 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
167 ///< per-frame (independent coding)
168 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded
169 ///< per superframe (residual coding)
171 int min_pitch_val; ///< base value for pitch parsing code
172 int max_pitch_val; ///< max value + 1 for pitch parsing
173 int pitch_nbits; ///< number of bits used to specify the
174 ///< pitch value in the frame header
175 int block_pitch_nbits; ///< number of bits used to specify the
176 ///< first block's pitch value
177 int block_pitch_range; ///< range of the block pitch
178 int block_delta_pitch_nbits; ///< number of bits used to specify the
179 ///< delta pitch between this and the last
180 ///< block's pitch value, used in all but
182 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
183 ///< from -this to +this-1)
184 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
190 * @name Packet values specified in the packet header or related to a packet.
192 * A packet is considered to be a single unit of data provided to this
193 * decoder by the demuxer.
196 int spillover_nbits; ///< number of bits of the previous packet's
197 ///< last superframe preceding this
198 ///< packet's first full superframe (useful
199 ///< for re-synchronization also)
200 int has_residual_lsps; ///< if set, superframes contain one set of
201 ///< LSPs that cover all frames, encoded as
202 ///< independent and residual LSPs; if not
203 ///< set, each frame contains its own, fully
204 ///< independent, LSPs
205 int skip_bits_next; ///< number of bits to skip at the next call
206 ///< to #wmavoice_decode_packet() (since
207 ///< they're part of the previous superframe)
209 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE];
210 ///< cache for superframe data split over
211 ///< multiple packets
212 int sframe_cache_size; ///< set to >0 if we have data from an
213 ///< (incomplete) superframe from a previous
214 ///< packet that spilled over in the current
215 ///< packet; specifies the amount of bits in
217 PutBitContext pb; ///< bitstream writer for #sframe_cache
222 * @name Frame and superframe values
223 * Superframe and frame data - these can change from frame to frame,
224 * although some of them do in that case serve as a cache / history for
225 * the next frame or superframe.
228 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
230 int last_pitch_val; ///< pitch value of the previous frame
231 int last_acb_type; ///< frame type [0-2] of the previous frame
232 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
233 ///< << 16) / #MAX_FRAMESIZE
234 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
236 int aw_idx_is_ext; ///< whether the AW index was encoded in
237 ///< 8 bits (instead of 6)
238 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
239 ///< can apply the pulse, relative to the
240 ///< value in aw_first_pulse_off. The exact
241 ///< position of the first AW-pulse is within
242 ///< [pulse_off, pulse_off + this], and
243 ///< depends on bitstream values; [16 or 24]
244 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
245 ///< that this number can be negative (in
246 ///< which case it basically means "zero")
247 int aw_first_pulse_off[2]; ///< index of first sample to which to
248 ///< apply AW-pulses, or -0xff if unset
249 int aw_next_pulse_off_cache; ///< the position (relative to start of the
250 ///< second block) at which pulses should
251 ///< start to be positioned, serves as a
252 ///< cache for pitch-adaptive window pulses
255 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
256 ///< only used for comfort noise in #pRNG()
257 float gain_pred_err[6]; ///< cache for gain prediction
258 float excitation_history[MAX_SIGNAL_HISTORY];
259 ///< cache of the signal of previous
260 ///< superframes, used as a history for
261 ///< signal generation
262 float synth_history[MAX_LSPS]; ///< see #excitation_history
266 * @name Postfilter values
268 * Variables used for postfilter implementation, mostly history for
269 * smoothing and so on, and context variables for FFT/iFFT.
272 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
273 ///< postfilter (for denoise filter)
274 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
275 ///< transform, part of postfilter)
276 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
278 float postfilter_agc; ///< gain control memory, used in
279 ///< #adaptive_gain_control()
280 float dcf_mem[2]; ///< DC filter history
281 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
282 ///< zero filter output (i.e. excitation)
284 float denoise_filter_cache[MAX_FRAMESIZE];
285 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
286 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
287 ///< aligned buffer for LPC tilting
288 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
289 ///< aligned buffer for denoise coefficients
290 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
291 ///< aligned buffer for postfilter speech
299 * Set up the variable bit mode (VBM) tree from container extradata.
300 * @param gb bit I/O context.
301 * The bit context (s->gb) should be loaded with byte 23-46 of the
302 * container extradata (i.e. the ones containing the VBM tree).
303 * @param vbm_tree pointer to array to which the decoded VBM tree will be
305 * @return 0 on success, <0 on error.
307 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
309 static const uint8_t bits[] = {
312 10, 10, 10, 12, 12, 12,
315 static const uint16_t codes[] = {
316 0x0000, 0x0001, 0x0002, // 00/01/10
317 0x000c, 0x000d, 0x000e, // 11+00/01/10
318 0x003c, 0x003d, 0x003e, // 1111+00/01/10
319 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
320 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
321 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
322 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
324 int cntr[8] = { 0 }, n, res;
326 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
327 for (n = 0; n < 17; n++) {
328 res = get_bits(gb, 3);
329 if (cntr[res] > 3) // should be >= 3 + (res == 7))
331 vbm_tree[res * 3 + cntr[res]++] = n;
333 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
334 bits, 1, 1, codes, 2, 2, 132);
339 * Set up decoder with parameters from demuxer (extradata etc.).
341 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
343 int n, flags, pitch_range, lsp16_flag;
344 WMAVoiceContext *s = ctx->priv_data;
348 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
349 * - byte 19-22: flags field (annoyingly in LE; see below for known
351 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
354 if (ctx->extradata_size != 46) {
355 av_log(ctx, AV_LOG_ERROR,
356 "Invalid extradata size %d (should be 46)\n",
357 ctx->extradata_size);
360 flags = AV_RL32(ctx->extradata + 18);
361 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
362 s->do_apf = flags & 0x1;
364 ff_rdft_init(&s->rdft, 7, DFT_R2C);
365 ff_rdft_init(&s->irdft, 7, IDFT_C2R);
366 ff_dct_init(&s->dct, 6, DCT_I);
367 ff_dct_init(&s->dst, 6, DST_I);
369 ff_sine_window_init(s->cos, 256);
370 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
371 for (n = 0; n < 255; n++) {
372 s->sin[n] = -s->sin[510 - n];
373 s->cos[510 - n] = s->cos[n];
376 s->denoise_strength = (flags >> 2) & 0xF;
377 if (s->denoise_strength >= 12) {
378 av_log(ctx, AV_LOG_ERROR,
379 "Invalid denoise filter strength %d (max=11)\n",
380 s->denoise_strength);
383 s->denoise_tilt_corr = !!(flags & 0x40);
384 s->dc_level = (flags >> 7) & 0xF;
385 s->lsp_q_mode = !!(flags & 0x2000);
386 s->lsp_def_mode = !!(flags & 0x4000);
387 lsp16_flag = flags & 0x1000;
390 s->frame_lsp_bitsize = 34;
391 s->sframe_lsp_bitsize = 60;
394 s->frame_lsp_bitsize = 24;
395 s->sframe_lsp_bitsize = 48;
397 for (n = 0; n < s->lsps; n++)
398 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
400 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
401 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
402 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
406 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
407 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
408 pitch_range = s->max_pitch_val - s->min_pitch_val;
409 if (pitch_range <= 0) {
410 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
413 s->pitch_nbits = av_ceil_log2(pitch_range);
414 s->last_pitch_val = 40;
415 s->last_acb_type = ACB_TYPE_NONE;
416 s->history_nsamples = s->max_pitch_val + 8;
418 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
419 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
420 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
422 av_log(ctx, AV_LOG_ERROR,
423 "Unsupported samplerate %d (min=%d, max=%d)\n",
424 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
429 s->block_conv_table[0] = s->min_pitch_val;
430 s->block_conv_table[1] = (pitch_range * 25) >> 6;
431 s->block_conv_table[2] = (pitch_range * 44) >> 6;
432 s->block_conv_table[3] = s->max_pitch_val - 1;
433 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
434 if (s->block_delta_pitch_hrange <= 0) {
435 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
438 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
439 s->block_pitch_range = s->block_conv_table[2] +
440 s->block_conv_table[3] + 1 +
441 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
442 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
445 ctx->channel_layout = AV_CH_LAYOUT_MONO;
446 ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
448 avcodec_get_frame_defaults(&s->frame);
449 ctx->coded_frame = &s->frame;
455 * @name Postfilter functions
456 * Postfilter functions (gain control, wiener denoise filter, DC filter,
457 * kalman smoothening, plus surrounding code to wrap it)
461 * Adaptive gain control (as used in postfilter).
463 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
464 * that the energy here is calculated using sum(abs(...)), whereas the
465 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
467 * @param out output buffer for filtered samples
468 * @param in input buffer containing the samples as they are after the
469 * postfilter steps so far
470 * @param speech_synth input buffer containing speech synth before postfilter
471 * @param size input buffer size
472 * @param alpha exponential filter factor
473 * @param gain_mem pointer to filter memory (single float)
475 static void adaptive_gain_control(float *out, const float *in,
476 const float *speech_synth,
477 int size, float alpha, float *gain_mem)
480 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
481 float mem = *gain_mem;
483 for (i = 0; i < size; i++) {
484 speech_energy += fabsf(speech_synth[i]);
485 postfilter_energy += fabsf(in[i]);
487 gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy;
489 for (i = 0; i < size; i++) {
490 mem = alpha * mem + gain_scale_factor;
491 out[i] = in[i] * mem;
498 * Kalman smoothing function.
500 * This function looks back pitch +/- 3 samples back into history to find
501 * the best fitting curve (that one giving the optimal gain of the two
502 * signals, i.e. the highest dot product between the two), and then
503 * uses that signal history to smoothen the output of the speech synthesis
506 * @param s WMA Voice decoding context
507 * @param pitch pitch of the speech signal
508 * @param in input speech signal
509 * @param out output pointer for smoothened signal
510 * @param size input/output buffer size
512 * @returns -1 if no smoothening took place, e.g. because no optimal
513 * fit could be found, or 0 on success.
515 static int kalman_smoothen(WMAVoiceContext *s, int pitch,
516 const float *in, float *out, int size)
519 float optimal_gain = 0, dot;
520 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
521 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
524 /* find best fitting point in history */
526 dot = avpriv_scalarproduct_float_c(in, ptr, size);
527 if (dot > optimal_gain) {
531 } while (--ptr >= end);
533 if (optimal_gain <= 0)
535 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
536 if (dot <= 0) // would be 1.0
539 if (optimal_gain <= dot) {
540 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
544 /* actual smoothing */
545 for (n = 0; n < size; n++)
546 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
552 * Get the tilt factor of a formant filter from its transfer function
553 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
554 * but somehow (??) it does a speech synthesis filter in the
555 * middle, which is missing here
557 * @param lpcs LPC coefficients
558 * @param n_lpcs Size of LPC buffer
559 * @returns the tilt factor
561 static float tilt_factor(const float *lpcs, int n_lpcs)
565 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
566 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
572 * Derive denoise filter coefficients (in real domain) from the LPCs.
574 static void calc_input_response(WMAVoiceContext *s, float *lpcs,
575 int fcb_type, float *coeffs, int remainder)
577 float last_coeff, min = 15.0, max = -15.0;
578 float irange, angle_mul, gain_mul, range, sq;
581 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
582 s->rdft.rdft_calc(&s->rdft, lpcs);
583 #define log_range(var, assign) do { \
584 float tmp = log10f(assign); var = tmp; \
585 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
587 log_range(last_coeff, lpcs[1] * lpcs[1]);
588 for (n = 1; n < 64; n++)
589 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
590 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
591 log_range(lpcs[0], lpcs[0] * lpcs[0]);
594 lpcs[64] = last_coeff;
596 /* Now, use this spectrum to pick out these frequencies with higher
597 * (relative) power/energy (which we then take to be "not noise"),
598 * and set up a table (still in lpc[]) of (relative) gains per frequency.
599 * These frequencies will be maintained, while others ("noise") will be
600 * decreased in the filter output. */
601 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
602 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
604 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
605 for (n = 0; n <= 64; n++) {
608 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
609 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
610 lpcs[n] = angle_mul * pwr;
612 /* 70.57 =~ 1/log10(1.0331663) */
613 idx = (pwr * gain_mul - 0.0295) * 70.570526123;
614 if (idx > 127) { // fallback if index falls outside table range
615 coeffs[n] = wmavoice_energy_table[127] *
616 powf(1.0331663, idx - 127);
618 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
621 /* calculate the Hilbert transform of the gains, which we do (since this
622 * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
623 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
624 * "moment" of the LPCs in this filter. */
625 s->dct.dct_calc(&s->dct, lpcs);
626 s->dst.dct_calc(&s->dst, lpcs);
628 /* Split out the coefficient indexes into phase/magnitude pairs */
629 idx = 255 + av_clip(lpcs[64], -255, 255);
630 coeffs[0] = coeffs[0] * s->cos[idx];
631 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
632 last_coeff = coeffs[64] * s->cos[idx];
634 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
635 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
636 coeffs[n * 2] = coeffs[n] * s->cos[idx];
640 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
641 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
642 coeffs[n * 2] = coeffs[n] * s->cos[idx];
644 coeffs[1] = last_coeff;
646 /* move into real domain */
647 s->irdft.rdft_calc(&s->irdft, coeffs);
649 /* tilt correction and normalize scale */
650 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
651 if (s->denoise_tilt_corr) {
654 coeffs[remainder - 1] = 0;
655 ff_tilt_compensation(&tilt_mem,
656 -1.8 * tilt_factor(coeffs, remainder - 1),
659 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
661 for (n = 0; n < remainder; n++)
666 * This function applies a Wiener filter on the (noisy) speech signal as
667 * a means to denoise it.
669 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
670 * - using this power spectrum, calculate (for each frequency) the Wiener
671 * filter gain, which depends on the frequency power and desired level
672 * of noise subtraction (when set too high, this leads to artifacts)
673 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
675 * - by doing a phase shift, calculate the Hilbert transform of this array
676 * of per-frequency filter-gains to get the filtering coefficients;
677 * - smoothen/normalize/de-tilt these filter coefficients as desired;
678 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
679 * to get the denoised speech signal;
680 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
681 * the frame boundary) are saved and applied to subsequent frames by an
682 * overlap-add method (otherwise you get clicking-artifacts).
684 * @param s WMA Voice decoding context
685 * @param fcb_type Frame (codebook) type
686 * @param synth_pf input: the noisy speech signal, output: denoised speech
687 * data; should be 16-byte aligned (for ASM purposes)
688 * @param size size of the speech data
689 * @param lpcs LPCs used to synthesize this frame's speech data
691 static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
692 float *synth_pf, int size,
695 int remainder, lim, n;
697 if (fcb_type != FCB_TYPE_SILENCE) {
698 float *tilted_lpcs = s->tilted_lpcs_pf,
699 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
701 tilted_lpcs[0] = 1.0;
702 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
703 memset(&tilted_lpcs[s->lsps + 1], 0,
704 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
705 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
706 tilted_lpcs, s->lsps + 2);
708 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
709 * size is applied to the next frame. All input beyond this is zero,
710 * and thus all output beyond this will go towards zero, hence we can
711 * limit to min(size-1, 127-size) as a performance consideration. */
712 remainder = FFMIN(127 - size, size - 1);
713 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
715 /* apply coefficients (in frequency spectrum domain), i.e. complex
716 * number multiplication */
717 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
718 s->rdft.rdft_calc(&s->rdft, synth_pf);
719 s->rdft.rdft_calc(&s->rdft, coeffs);
720 synth_pf[0] *= coeffs[0];
721 synth_pf[1] *= coeffs[1];
722 for (n = 1; n < 64; n++) {
723 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
724 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
725 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
727 s->irdft.rdft_calc(&s->irdft, synth_pf);
730 /* merge filter output with the history of previous runs */
731 if (s->denoise_filter_cache_size) {
732 lim = FFMIN(s->denoise_filter_cache_size, size);
733 for (n = 0; n < lim; n++)
734 synth_pf[n] += s->denoise_filter_cache[n];
735 s->denoise_filter_cache_size -= lim;
736 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
737 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
740 /* move remainder of filter output into a cache for future runs */
741 if (fcb_type != FCB_TYPE_SILENCE) {
742 lim = FFMIN(remainder, s->denoise_filter_cache_size);
743 for (n = 0; n < lim; n++)
744 s->denoise_filter_cache[n] += synth_pf[size + n];
745 if (lim < remainder) {
746 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
747 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
748 s->denoise_filter_cache_size = remainder;
754 * Averaging projection filter, the postfilter used in WMAVoice.
756 * This uses the following steps:
757 * - A zero-synthesis filter (generate excitation from synth signal)
758 * - Kalman smoothing on excitation, based on pitch
759 * - Re-synthesized smoothened output
760 * - Iterative Wiener denoise filter
761 * - Adaptive gain filter
764 * @param s WMAVoice decoding context
765 * @param synth Speech synthesis output (before postfilter)
766 * @param samples Output buffer for filtered samples
767 * @param size Buffer size of synth & samples
768 * @param lpcs Generated LPCs used for speech synthesis
769 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
770 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
771 * @param pitch Pitch of the input signal
773 static void postfilter(WMAVoiceContext *s, const float *synth,
774 float *samples, int size,
775 const float *lpcs, float *zero_exc_pf,
776 int fcb_type, int pitch)
778 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
779 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
780 *synth_filter_in = zero_exc_pf;
782 assert(size <= MAX_FRAMESIZE / 2);
784 /* generate excitation from input signal */
785 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
787 if (fcb_type >= FCB_TYPE_AW_PULSES &&
788 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
789 synth_filter_in = synth_filter_in_buf;
791 /* re-synthesize speech after smoothening, and keep history */
792 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
793 synth_filter_in, size, s->lsps);
794 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
795 sizeof(synth_pf[0]) * s->lsps);
797 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
799 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
802 if (s->dc_level > 8) {
803 /* remove ultra-low frequency DC noise / highpass filter;
804 * coefficients are identical to those used in SIPR decoding,
805 * and very closely resemble those used in AMR-NB decoding. */
806 ff_acelp_apply_order_2_transfer_function(samples, samples,
807 (const float[2]) { -1.99997, 1.0 },
808 (const float[2]) { -1.9330735188, 0.93589198496 },
809 0.93980580475, s->dcf_mem, size);
818 * @param lsps output pointer to the array that will hold the LSPs
819 * @param num number of LSPs to be dequantized
820 * @param values quantized values, contains n_stages values
821 * @param sizes range (i.e. max value) of each quantized value
822 * @param n_stages number of dequantization runs
823 * @param table dequantization table to be used
824 * @param mul_q LSF multiplier
825 * @param base_q base (lowest) LSF values
827 static void dequant_lsps(double *lsps, int num,
828 const uint16_t *values,
829 const uint16_t *sizes,
830 int n_stages, const uint8_t *table,
832 const double *base_q)
836 memset(lsps, 0, num * sizeof(*lsps));
837 for (n = 0; n < n_stages; n++) {
838 const uint8_t *t_off = &table[values[n] * num];
839 double base = base_q[n], mul = mul_q[n];
841 for (m = 0; m < num; m++)
842 lsps[m] += base + mul * t_off[m];
844 table += sizes[n] * num;
849 * @name LSP dequantization routines
850 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
851 * @note we assume enough bits are available, caller should check.
852 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
853 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
857 * Parse 10 independently-coded LSPs.
859 static void dequant_lsp10i(GetBitContext *gb, double *lsps)
861 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
862 static const double mul_lsf[4] = {
863 5.2187144800e-3, 1.4626986422e-3,
864 9.6179549166e-4, 1.1325736225e-3
866 static const double base_lsf[4] = {
867 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
868 M_PI * -3.3486e-2, M_PI * -5.7408e-2
872 v[0] = get_bits(gb, 8);
873 v[1] = get_bits(gb, 6);
874 v[2] = get_bits(gb, 5);
875 v[3] = get_bits(gb, 5);
877 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
882 * Parse 10 independently-coded LSPs, and then derive the tables to
883 * generate LSPs for the other frames from them (residual coding).
885 static void dequant_lsp10r(GetBitContext *gb,
886 double *i_lsps, const double *old,
887 double *a1, double *a2, int q_mode)
889 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
890 static const double mul_lsf[3] = {
891 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
893 static const double base_lsf[3] = {
894 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
896 const float (*ipol_tab)[2][10] = q_mode ?
897 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
898 uint16_t interpol, v[3];
901 dequant_lsp10i(gb, i_lsps);
903 interpol = get_bits(gb, 5);
904 v[0] = get_bits(gb, 7);
905 v[1] = get_bits(gb, 6);
906 v[2] = get_bits(gb, 6);
908 for (n = 0; n < 10; n++) {
909 double delta = old[n] - i_lsps[n];
910 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
911 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
914 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
919 * Parse 16 independently-coded LSPs.
921 static void dequant_lsp16i(GetBitContext *gb, double *lsps)
923 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
924 static const double mul_lsf[5] = {
925 3.3439586280e-3, 6.9908173703e-4,
926 3.3216608306e-3, 1.0334960326e-3,
929 static const double base_lsf[5] = {
930 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
931 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
936 v[0] = get_bits(gb, 8);
937 v[1] = get_bits(gb, 6);
938 v[2] = get_bits(gb, 7);
939 v[3] = get_bits(gb, 6);
940 v[4] = get_bits(gb, 7);
942 dequant_lsps( lsps, 5, v, vec_sizes, 2,
943 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
944 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
945 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
946 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
947 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
951 * Parse 16 independently-coded LSPs, and then derive the tables to
952 * generate LSPs for the other frames from them (residual coding).
954 static void dequant_lsp16r(GetBitContext *gb,
955 double *i_lsps, const double *old,
956 double *a1, double *a2, int q_mode)
958 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
959 static const double mul_lsf[3] = {
960 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
962 static const double base_lsf[3] = {
963 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
965 const float (*ipol_tab)[2][16] = q_mode ?
966 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
967 uint16_t interpol, v[3];
970 dequant_lsp16i(gb, i_lsps);
972 interpol = get_bits(gb, 5);
973 v[0] = get_bits(gb, 7);
974 v[1] = get_bits(gb, 7);
975 v[2] = get_bits(gb, 7);
977 for (n = 0; n < 16; n++) {
978 double delta = old[n] - i_lsps[n];
979 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
980 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
983 dequant_lsps( a2, 10, v, vec_sizes, 1,
984 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
985 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
986 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
987 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
988 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
993 * @name Pitch-adaptive window coding functions
994 * The next few functions are for pitch-adaptive window coding.
998 * Parse the offset of the first pitch-adaptive window pulses, and
999 * the distribution of pulses between the two blocks in this frame.
1000 * @param s WMA Voice decoding context private data
1001 * @param gb bit I/O context
1002 * @param pitch pitch for each block in this frame
1004 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
1007 static const int16_t start_offset[94] = {
1008 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1009 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1010 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1011 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1012 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1013 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1014 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1015 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1019 /* position of pulse */
1020 s->aw_idx_is_ext = 0;
1021 if ((bits = get_bits(gb, 6)) >= 54) {
1022 s->aw_idx_is_ext = 1;
1023 bits += (bits - 54) * 3 + get_bits(gb, 2);
1026 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1027 * the distribution of the pulses in each block contained in this frame. */
1028 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1029 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1030 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1031 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1032 offset += s->aw_n_pulses[0] * pitch[0];
1033 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1034 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1036 /* if continuing from a position before the block, reset position to
1037 * start of block (when corrected for the range over which it can be
1038 * spread in aw_pulse_set1()). */
1039 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1040 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1041 s->aw_first_pulse_off[1] -= pitch[1];
1042 if (start_offset[bits] < 0)
1043 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1044 s->aw_first_pulse_off[0] -= pitch[0];
1049 * Apply second set of pitch-adaptive window pulses.
1050 * @param s WMA Voice decoding context private data
1051 * @param gb bit I/O context
1052 * @param block_idx block index in frame [0, 1]
1053 * @param fcb structure containing fixed codebook vector info
1055 static void 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;
1138 * Apply first set of pitch-adaptive window pulses.
1139 * @param s WMA Voice decoding context private data
1140 * @param gb bit I/O context
1141 * @param block_idx block index in frame [0, 1]
1142 * @param fcb storage location for fixed codebook pulse info
1144 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1145 int block_idx, AMRFixed *fcb)
1147 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1150 if (s->aw_n_pulses[block_idx] > 0) {
1151 int n, v_mask, i_mask, sh, n_pulses;
1153 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1158 } else { // 4 pulses, 1:sign + 2:index each
1165 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1166 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1167 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1168 s->aw_first_pulse_off[block_idx];
1169 while (fcb->x[fcb->n] < 0)
1170 fcb->x[fcb->n] += fcb->pitch_lag;
1171 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1175 int num2 = (val & 0x1FF) >> 1, delta, idx;
1177 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1178 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1179 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1180 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1181 v = (val & 0x200) ? -1.0 : 1.0;
1183 fcb->no_repeat_mask |= 3 << fcb->n;
1184 fcb->x[fcb->n] = idx - delta;
1186 fcb->x[fcb->n + 1] = idx;
1187 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1195 * Generate a random number from frame_cntr and block_idx, which will lief
1196 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1197 * table of size 1000 of which you want to read block_size entries).
1199 * @param frame_cntr current frame number
1200 * @param block_num current block index
1201 * @param block_size amount of entries we want to read from a table
1202 * that has 1000 entries
1203 * @return a (non-)random number in the [0, 1000 - block_size] range.
1205 static int pRNG(int frame_cntr, int block_num, int block_size)
1207 /* array to simplify the calculation of z:
1208 * y = (x % 9) * 5 + 6;
1209 * z = (49995 * x) / y;
1210 * Since y only has 9 values, we can remove the division by using a
1211 * LUT and using FASTDIV-style divisions. For each of the 9 values
1212 * of y, we can rewrite z as:
1213 * z = x * (49995 / y) + x * ((49995 % y) / y)
1214 * In this table, each col represents one possible value of y, the
1215 * first number is 49995 / y, and the second is the FASTDIV variant
1216 * of 49995 % y / y. */
1217 static const unsigned int div_tbl[9][2] = {
1218 { 8332, 3 * 715827883U }, // y = 6
1219 { 4545, 0 * 390451573U }, // y = 11
1220 { 3124, 11 * 268435456U }, // y = 16
1221 { 2380, 15 * 204522253U }, // y = 21
1222 { 1922, 23 * 165191050U }, // y = 26
1223 { 1612, 23 * 138547333U }, // y = 31
1224 { 1388, 27 * 119304648U }, // y = 36
1225 { 1219, 16 * 104755300U }, // y = 41
1226 { 1086, 39 * 93368855U } // y = 46
1228 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1229 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1230 // so this is effectively a modulo (%)
1231 y = x - 9 * MULH(477218589, x); // x % 9
1232 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1233 // z = x * 49995 / (y * 5 + 6)
1234 return z % (1000 - block_size);
1238 * Parse hardcoded signal for a single block.
1239 * @note see #synth_block().
1241 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1242 int block_idx, int size,
1243 const struct frame_type_desc *frame_desc,
1249 assert(size <= MAX_FRAMESIZE);
1251 /* Set the offset from which we start reading wmavoice_std_codebook */
1252 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1253 r_idx = pRNG(s->frame_cntr, block_idx, size);
1254 gain = s->silence_gain;
1255 } else /* FCB_TYPE_HARDCODED */ {
1256 r_idx = get_bits(gb, 8);
1257 gain = wmavoice_gain_universal[get_bits(gb, 6)];
1260 /* Clear gain prediction parameters */
1261 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1263 /* Apply gain to hardcoded codebook and use that as excitation signal */
1264 for (n = 0; n < size; n++)
1265 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1269 * Parse FCB/ACB signal for a single block.
1270 * @note see #synth_block().
1272 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1273 int block_idx, int size,
1274 int block_pitch_sh2,
1275 const struct frame_type_desc *frame_desc,
1278 static const float gain_coeff[6] = {
1279 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1281 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1282 int n, idx, gain_weight;
1285 assert(size <= MAX_FRAMESIZE / 2);
1286 memset(pulses, 0, sizeof(*pulses) * size);
1288 fcb.pitch_lag = block_pitch_sh2 >> 2;
1289 fcb.pitch_fac = 1.0;
1290 fcb.no_repeat_mask = 0;
1293 /* For the other frame types, this is where we apply the innovation
1294 * (fixed) codebook pulses of the speech signal. */
1295 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1296 aw_pulse_set1(s, gb, block_idx, &fcb);
1297 aw_pulse_set2(s, gb, block_idx, &fcb);
1298 } else /* FCB_TYPE_EXC_PULSES */ {
1299 int offset_nbits = 5 - frame_desc->log_n_blocks;
1301 fcb.no_repeat_mask = -1;
1302 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1303 * (instead of double) for a subset of pulses */
1304 for (n = 0; n < 5; n++) {
1308 sign = get_bits1(gb) ? 1.0 : -1.0;
1309 pos1 = get_bits(gb, offset_nbits);
1310 fcb.x[fcb.n] = n + 5 * pos1;
1311 fcb.y[fcb.n++] = sign;
1312 if (n < frame_desc->dbl_pulses) {
1313 pos2 = get_bits(gb, offset_nbits);
1314 fcb.x[fcb.n] = n + 5 * pos2;
1315 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1319 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1321 /* Calculate gain for adaptive & fixed codebook signal.
1322 * see ff_amr_set_fixed_gain(). */
1323 idx = get_bits(gb, 7);
1324 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
1326 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1327 acb_gain = wmavoice_gain_codebook_acb[idx];
1328 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1329 -2.9957322736 /* log(0.05) */,
1330 1.6094379124 /* log(5.0) */);
1332 gain_weight = 8 >> frame_desc->log_n_blocks;
1333 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1334 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1335 for (n = 0; n < gain_weight; n++)
1336 s->gain_pred_err[n] = pred_err;
1338 /* Calculation of adaptive codebook */
1339 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1341 for (n = 0; n < size; n += len) {
1343 int abs_idx = block_idx * size + n;
1344 int pitch_sh16 = (s->last_pitch_val << 16) +
1345 s->pitch_diff_sh16 * abs_idx;
1346 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1347 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1348 idx = idx_sh16 >> 16;
1349 if (s->pitch_diff_sh16) {
1350 if (s->pitch_diff_sh16 > 0) {
1351 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1353 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1354 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1359 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1360 wmavoice_ipol1_coeffs, 17,
1363 } else /* ACB_TYPE_HAMMING */ {
1364 int block_pitch = block_pitch_sh2 >> 2;
1365 idx = block_pitch_sh2 & 3;
1367 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1368 wmavoice_ipol2_coeffs, 4,
1371 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1372 sizeof(float) * size);
1375 /* Interpolate ACB/FCB and use as excitation signal */
1376 ff_weighted_vector_sumf(excitation, excitation, pulses,
1377 acb_gain, fcb_gain, size);
1381 * Parse data in a single block.
1382 * @note we assume enough bits are available, caller should check.
1384 * @param s WMA Voice decoding context private data
1385 * @param gb bit I/O context
1386 * @param block_idx index of the to-be-read block
1387 * @param size amount of samples to be read in this block
1388 * @param block_pitch_sh2 pitch for this block << 2
1389 * @param lsps LSPs for (the end of) this frame
1390 * @param prev_lsps LSPs for the last frame
1391 * @param frame_desc frame type descriptor
1392 * @param excitation target memory for the ACB+FCB interpolated signal
1393 * @param synth target memory for the speech synthesis filter output
1394 * @return 0 on success, <0 on error.
1396 static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1397 int block_idx, int size,
1398 int block_pitch_sh2,
1399 const double *lsps, const double *prev_lsps,
1400 const struct frame_type_desc *frame_desc,
1401 float *excitation, float *synth)
1403 double i_lsps[MAX_LSPS];
1404 float lpcs[MAX_LSPS];
1408 if (frame_desc->acb_type == ACB_TYPE_NONE)
1409 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1411 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1412 frame_desc, excitation);
1414 /* convert interpolated LSPs to LPCs */
1415 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1416 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1417 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1418 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1420 /* Speech synthesis */
1421 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1425 * Synthesize output samples for a single frame.
1426 * @note we assume enough bits are available, caller should check.
1428 * @param ctx WMA Voice decoder context
1429 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1430 * @param frame_idx Frame number within superframe [0-2]
1431 * @param samples pointer to output sample buffer, has space for at least 160
1433 * @param lsps LSP array
1434 * @param prev_lsps array of previous frame's LSPs
1435 * @param excitation target buffer for excitation signal
1436 * @param synth target buffer for synthesized speech data
1437 * @return 0 on success, <0 on error.
1439 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1441 const double *lsps, const double *prev_lsps,
1442 float *excitation, float *synth)
1444 WMAVoiceContext *s = ctx->priv_data;
1445 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val;
1446 int pitch[MAX_BLOCKS], last_block_pitch;
1448 /* Parse frame type ("frame header"), see frame_descs */
1449 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
1452 av_log(ctx, AV_LOG_ERROR,
1453 "Invalid frame type VLC code, skipping\n");
1457 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1459 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1460 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1461 /* Pitch is provided per frame, which is interpreted as the pitch of
1462 * the last sample of the last block of this frame. We can interpolate
1463 * the pitch of other blocks (and even pitch-per-sample) by gradually
1464 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1465 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1466 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1467 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1468 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1469 if (s->last_acb_type == ACB_TYPE_NONE ||
1470 20 * abs(cur_pitch_val - s->last_pitch_val) >
1471 (cur_pitch_val + s->last_pitch_val))
1472 s->last_pitch_val = cur_pitch_val;
1474 /* pitch per block */
1475 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1476 int fac = n * 2 + 1;
1478 pitch[n] = (MUL16(fac, cur_pitch_val) +
1479 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1480 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1483 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1484 s->pitch_diff_sh16 =
1485 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1488 /* Global gain (if silence) and pitch-adaptive window coordinates */
1489 switch (frame_descs[bd_idx].fcb_type) {
1490 case FCB_TYPE_SILENCE:
1491 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1493 case FCB_TYPE_AW_PULSES:
1494 aw_parse_coords(s, gb, pitch);
1498 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1501 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1502 switch (frame_descs[bd_idx].acb_type) {
1503 case ACB_TYPE_HAMMING: {
1504 /* Pitch is given per block. Per-block pitches are encoded as an
1505 * absolute value for the first block, and then delta values
1506 * relative to this value) for all subsequent blocks. The scale of
1507 * this pitch value is semi-logaritmic compared to its use in the
1508 * decoder, so we convert it to normal scale also. */
1510 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1511 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1512 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1515 block_pitch = get_bits(gb, s->block_pitch_nbits);
1517 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1518 get_bits(gb, s->block_delta_pitch_nbits);
1519 /* Convert last_ so that any next delta is within _range */
1520 last_block_pitch = av_clip(block_pitch,
1521 s->block_delta_pitch_hrange,
1522 s->block_pitch_range -
1523 s->block_delta_pitch_hrange);
1525 /* Convert semi-log-style scale back to normal scale */
1526 if (block_pitch < t1) {
1527 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1530 if (block_pitch < t2) {
1532 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1535 if (block_pitch < t3) {
1537 (s->block_conv_table[2] + block_pitch) << 2;
1539 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1542 pitch[n] = bl_pitch_sh2 >> 2;
1546 case ACB_TYPE_ASYMMETRIC: {
1547 bl_pitch_sh2 = pitch[n] << 2;
1551 default: // ACB_TYPE_NONE has no pitch
1556 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1557 lsps, prev_lsps, &frame_descs[bd_idx],
1558 &excitation[n * block_nsamples],
1559 &synth[n * block_nsamples]);
1562 /* Averaging projection filter, if applicable. Else, just copy samples
1563 * from synthesis buffer */
1565 double i_lsps[MAX_LSPS];
1566 float lpcs[MAX_LSPS];
1568 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1569 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1570 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1571 postfilter(s, synth, samples, 80, lpcs,
1572 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1573 frame_descs[bd_idx].fcb_type, pitch[0]);
1575 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1576 i_lsps[n] = cos(lsps[n]);
1577 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1578 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1579 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1580 frame_descs[bd_idx].fcb_type, pitch[0]);
1582 memcpy(samples, synth, 160 * sizeof(synth[0]));
1584 /* Cache values for next frame */
1586 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1587 s->last_acb_type = frame_descs[bd_idx].acb_type;
1588 switch (frame_descs[bd_idx].acb_type) {
1590 s->last_pitch_val = 0;
1592 case ACB_TYPE_ASYMMETRIC:
1593 s->last_pitch_val = cur_pitch_val;
1595 case ACB_TYPE_HAMMING:
1596 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1604 * Ensure minimum value for first item, maximum value for last value,
1605 * proper spacing between each value and proper ordering.
1607 * @param lsps array of LSPs
1608 * @param num size of LSP array
1610 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1611 * useful to put in a generic location later on. Parts are also
1612 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1613 * which is in float.
1615 static void stabilize_lsps(double *lsps, int num)
1619 /* set minimum value for first, maximum value for last and minimum
1620 * spacing between LSF values.
1621 * Very similar to ff_set_min_dist_lsf(), but in double. */
1622 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1623 for (n = 1; n < num; n++)
1624 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1625 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1627 /* reorder (looks like one-time / non-recursed bubblesort).
1628 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1629 for (n = 1; n < num; n++) {
1630 if (lsps[n] < lsps[n - 1]) {
1631 for (m = 1; m < num; m++) {
1632 double tmp = lsps[m];
1633 for (l = m - 1; l >= 0; l--) {
1634 if (lsps[l] <= tmp) break;
1635 lsps[l + 1] = lsps[l];
1645 * Test if there's enough bits to read 1 superframe.
1647 * @param orig_gb bit I/O context used for reading. This function
1648 * does not modify the state of the bitreader; it
1649 * only uses it to copy the current stream position
1650 * @param s WMA Voice decoding context private data
1651 * @return -1 if unsupported, 1 on not enough bits or 0 if OK.
1653 static int check_bits_for_superframe(GetBitContext *orig_gb,
1656 GetBitContext s_gb, *gb = &s_gb;
1657 int n, need_bits, bd_idx;
1658 const struct frame_type_desc *frame_desc;
1660 /* initialize a copy */
1661 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits);
1662 skip_bits_long(gb, get_bits_count(orig_gb));
1663 assert(get_bits_left(gb) == get_bits_left(orig_gb));
1665 /* superframe header */
1666 if (get_bits_left(gb) < 14)
1669 return -1; // WMAPro-in-WMAVoice superframe
1670 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe
1671 if (s->has_residual_lsps) { // residual LSPs (for all frames)
1672 if (get_bits_left(gb) < s->sframe_lsp_bitsize)
1674 skip_bits_long(gb, s->sframe_lsp_bitsize);
1678 for (n = 0; n < MAX_FRAMES; n++) {
1679 int aw_idx_is_ext = 0;
1681 if (!s->has_residual_lsps) { // independent LSPs (per-frame)
1682 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1;
1683 skip_bits_long(gb, s->frame_lsp_bitsize);
1685 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)];
1687 return -1; // invalid frame type VLC code
1688 frame_desc = &frame_descs[bd_idx];
1689 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1690 if (get_bits_left(gb) < s->pitch_nbits)
1692 skip_bits_long(gb, s->pitch_nbits);
1694 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1696 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1697 int tmp = get_bits(gb, 6);
1705 if (frame_desc->acb_type == ACB_TYPE_HAMMING) {
1706 need_bits = s->block_pitch_nbits +
1707 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits;
1708 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1709 need_bits = 2 * !aw_idx_is_ext;
1712 need_bits += frame_desc->frame_size;
1713 if (get_bits_left(gb) < need_bits)
1715 skip_bits_long(gb, need_bits);
1722 * Synthesize output samples for a single superframe. If we have any data
1723 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1726 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1727 * to give a total of 480 samples per frame. See #synth_frame() for frame
1728 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1729 * (if these are globally specified for all frames (residually); they can
1730 * also be specified individually per-frame. See the s->has_residual_lsps
1731 * option), and can specify the number of samples encoded in this superframe
1732 * (if less than 480), usually used to prevent blanks at track boundaries.
1734 * @param ctx WMA Voice decoder context
1735 * @return 0 on success, <0 on error or 1 if there was not enough data to
1736 * fully parse the superframe
1738 static int synth_superframe(AVCodecContext *ctx, int *got_frame_ptr)
1740 WMAVoiceContext *s = ctx->priv_data;
1741 GetBitContext *gb = &s->gb, s_gb;
1742 int n, res, n_samples = 480;
1743 double lsps[MAX_FRAMES][MAX_LSPS];
1744 const double *mean_lsf = s->lsps == 16 ?
1745 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1746 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1747 float synth[MAX_LSPS + MAX_SFRAMESIZE];
1750 memcpy(synth, s->synth_history,
1751 s->lsps * sizeof(*synth));
1752 memcpy(excitation, s->excitation_history,
1753 s->history_nsamples * sizeof(*excitation));
1755 if (s->sframe_cache_size > 0) {
1757 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1758 s->sframe_cache_size = 0;
1761 if ((res = check_bits_for_superframe(gb, s)) == 1) {
1766 /* First bit is speech/music bit, it differentiates between WMAVoice
1767 * speech samples (the actual codec) and WMAVoice music samples, which
1768 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1770 if (!get_bits1(gb)) {
1771 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice", 1);
1772 return AVERROR_PATCHWELCOME;
1775 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1776 if (get_bits1(gb)) {
1777 if ((n_samples = get_bits(gb, 12)) > 480) {
1778 av_log(ctx, AV_LOG_ERROR,
1779 "Superframe encodes >480 samples (%d), not allowed\n",
1784 /* Parse LSPs, if global for the superframe (can also be per-frame). */
1785 if (s->has_residual_lsps) {
1786 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1788 for (n = 0; n < s->lsps; n++)
1789 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1791 if (s->lsps == 10) {
1792 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1793 } else /* s->lsps == 16 */
1794 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1796 for (n = 0; n < s->lsps; n++) {
1797 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
1798 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1799 lsps[2][n] += mean_lsf[n];
1801 for (n = 0; n < 3; n++)
1802 stabilize_lsps(lsps[n], s->lsps);
1805 /* get output buffer */
1806 s->frame.nb_samples = 480;
1807 if ((res = ff_get_buffer(ctx, &s->frame)) < 0) {
1808 av_log(ctx, AV_LOG_ERROR, "get_buffer() failed\n");
1811 s->frame.nb_samples = n_samples;
1812 samples = (float *)s->frame.data[0];
1814 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1815 for (n = 0; n < 3; n++) {
1816 if (!s->has_residual_lsps) {
1819 if (s->lsps == 10) {
1820 dequant_lsp10i(gb, lsps[n]);
1821 } else /* s->lsps == 16 */
1822 dequant_lsp16i(gb, lsps[n]);
1824 for (m = 0; m < s->lsps; m++)
1825 lsps[n][m] += mean_lsf[m];
1826 stabilize_lsps(lsps[n], s->lsps);
1829 if ((res = synth_frame(ctx, gb, n,
1830 &samples[n * MAX_FRAMESIZE],
1831 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1832 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1833 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
1839 /* Statistics? FIXME - we don't check for length, a slight overrun
1840 * will be caught by internal buffer padding, and anything else
1841 * will be skipped, not read. */
1842 if (get_bits1(gb)) {
1843 res = get_bits(gb, 4);
1844 skip_bits(gb, 10 * (res + 1));
1849 /* Update history */
1850 memcpy(s->prev_lsps, lsps[2],
1851 s->lsps * sizeof(*s->prev_lsps));
1852 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1853 s->lsps * sizeof(*synth));
1854 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1855 s->history_nsamples * sizeof(*excitation));
1857 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1858 s->history_nsamples * sizeof(*s->zero_exc_pf));
1864 * Parse the packet header at the start of each packet (input data to this
1867 * @param s WMA Voice decoding context private data
1868 * @return 1 if not enough bits were available, or 0 on success.
1870 static int parse_packet_header(WMAVoiceContext *s)
1872 GetBitContext *gb = &s->gb;
1875 if (get_bits_left(gb) < 11)
1877 skip_bits(gb, 4); // packet sequence number
1878 s->has_residual_lsps = get_bits1(gb);
1880 res = get_bits(gb, 6); // number of superframes per packet
1881 // (minus first one if there is spillover)
1882 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize)
1884 } while (res == 0x3F);
1885 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
1891 * Copy (unaligned) bits from gb/data/size to pb.
1893 * @param pb target buffer to copy bits into
1894 * @param data source buffer to copy bits from
1895 * @param size size of the source data, in bytes
1896 * @param gb bit I/O context specifying the current position in the source.
1897 * data. This function might use this to align the bit position to
1898 * a whole-byte boundary before calling #avpriv_copy_bits() on aligned
1900 * @param nbits the amount of bits to copy from source to target
1902 * @note after calling this function, the current position in the input bit
1903 * I/O context is undefined.
1905 static void copy_bits(PutBitContext *pb,
1906 const uint8_t *data, int size,
1907 GetBitContext *gb, int nbits)
1909 int rmn_bytes, rmn_bits;
1911 rmn_bits = rmn_bytes = get_bits_left(gb);
1912 if (rmn_bits < nbits)
1914 if (nbits > pb->size_in_bits - put_bits_count(pb))
1916 rmn_bits &= 7; rmn_bytes >>= 3;
1917 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1918 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1919 avpriv_copy_bits(pb, data + size - rmn_bytes,
1920 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1924 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1925 * and we expect that the demuxer / application provides it to us as such
1926 * (else you'll probably get garbage as output). Every packet has a size of
1927 * ctx->block_align bytes, starts with a packet header (see
1928 * #parse_packet_header()), and then a series of superframes. Superframe
1929 * boundaries may exceed packets, i.e. superframes can split data over
1930 * multiple (two) packets.
1932 * For more information about frames, see #synth_superframe().
1934 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1935 int *got_frame_ptr, AVPacket *avpkt)
1937 WMAVoiceContext *s = ctx->priv_data;
1938 GetBitContext *gb = &s->gb;
1941 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1942 * header at each ctx->block_align bytes. However, Libav's ASF demuxer
1943 * feeds us ASF packets, which may concatenate multiple "codec" packets
1944 * in a single "muxer" packet, so we artificially emulate that by
1945 * capping the packet size at ctx->block_align. */
1946 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1951 init_get_bits(&s->gb, avpkt->data, size << 3);
1953 /* size == ctx->block_align is used to indicate whether we are dealing with
1954 * a new packet or a packet of which we already read the packet header
1956 if (size == ctx->block_align) { // new packet header
1957 if ((res = parse_packet_header(s)) < 0)
1960 /* If the packet header specifies a s->spillover_nbits, then we want
1961 * to push out all data of the previous packet (+ spillover) before
1962 * continuing to parse new superframes in the current packet. */
1963 if (s->spillover_nbits > 0) {
1964 if (s->sframe_cache_size > 0) {
1965 int cnt = get_bits_count(gb);
1966 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1967 flush_put_bits(&s->pb);
1968 s->sframe_cache_size += s->spillover_nbits;
1969 if ((res = synth_superframe(ctx, got_frame_ptr)) == 0 &&
1971 cnt += s->spillover_nbits;
1972 s->skip_bits_next = cnt & 7;
1973 *(AVFrame *)data = s->frame;
1976 skip_bits_long (gb, s->spillover_nbits - cnt +
1977 get_bits_count(gb)); // resync
1979 skip_bits_long(gb, s->spillover_nbits); // resync
1981 } else if (s->skip_bits_next)
1982 skip_bits(gb, s->skip_bits_next);
1984 /* Try parsing superframes in current packet */
1985 s->sframe_cache_size = 0;
1986 s->skip_bits_next = 0;
1987 pos = get_bits_left(gb);
1988 if ((res = synth_superframe(ctx, got_frame_ptr)) < 0) {
1990 } else if (*got_frame_ptr) {
1991 int cnt = get_bits_count(gb);
1992 s->skip_bits_next = cnt & 7;
1993 *(AVFrame *)data = s->frame;
1995 } else if ((s->sframe_cache_size = pos) > 0) {
1996 /* rewind bit reader to start of last (incomplete) superframe... */
1997 init_get_bits(gb, avpkt->data, size << 3);
1998 skip_bits_long(gb, (size << 3) - pos);
1999 assert(get_bits_left(gb) == pos);
2001 /* ...and cache it for spillover in next packet */
2002 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
2003 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
2004 // FIXME bad - just copy bytes as whole and add use the
2005 // skip_bits_next field
2011 static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
2013 WMAVoiceContext *s = ctx->priv_data;
2016 ff_rdft_end(&s->rdft);
2017 ff_rdft_end(&s->irdft);
2018 ff_dct_end(&s->dct);
2019 ff_dct_end(&s->dst);
2025 static av_cold void wmavoice_flush(AVCodecContext *ctx)
2027 WMAVoiceContext *s = ctx->priv_data;
2030 s->postfilter_agc = 0;
2031 s->sframe_cache_size = 0;
2032 s->skip_bits_next = 0;
2033 for (n = 0; n < s->lsps; n++)
2034 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
2035 memset(s->excitation_history, 0,
2036 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
2037 memset(s->synth_history, 0,
2038 sizeof(*s->synth_history) * MAX_LSPS);
2039 memset(s->gain_pred_err, 0,
2040 sizeof(s->gain_pred_err));
2043 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
2044 sizeof(*s->synth_filter_out_buf) * s->lsps);
2045 memset(s->dcf_mem, 0,
2046 sizeof(*s->dcf_mem) * 2);
2047 memset(s->zero_exc_pf, 0,
2048 sizeof(*s->zero_exc_pf) * s->history_nsamples);
2049 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
2053 AVCodec ff_wmavoice_decoder = {
2055 .type = AVMEDIA_TYPE_AUDIO,
2056 .id = AV_CODEC_ID_WMAVOICE,
2057 .priv_data_size = sizeof(WMAVoiceContext),
2058 .init = wmavoice_decode_init,
2059 .close = wmavoice_decode_end,
2060 .decode = wmavoice_decode_packet,
2061 .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
2062 .flush = wmavoice_flush,
2063 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),