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 } frame_descs[17] = {
108 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0 },
109 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0 },
110 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0 },
111 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 },
112 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 },
113 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0 },
114 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 },
115 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 },
116 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 },
117 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 },
118 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 },
119 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 },
120 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 },
121 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 },
122 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 },
123 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 },
124 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }
128 * WMA Voice decoding context.
130 typedef struct WMAVoiceContext {
132 * @name Global values specified in the stream header / extradata or used all over.
135 GetBitContext gb; ///< packet bitreader. During decoder init,
136 ///< it contains the extradata from the
137 ///< demuxer. During decoding, it contains
139 int8_t vbm_tree[25]; ///< converts VLC codes to frame type
141 int spillover_bitsize; ///< number of bits used to specify
142 ///< #spillover_nbits in the packet header
143 ///< = ceil(log2(ctx->block_align << 3))
144 int history_nsamples; ///< number of samples in history for signal
145 ///< prediction (through ACB)
147 /* postfilter specific values */
148 int do_apf; ///< whether to apply the averaged
149 ///< projection filter (APF)
150 int denoise_strength; ///< strength of denoising in Wiener filter
152 int denoise_tilt_corr; ///< Whether to apply tilt correction to the
153 ///< Wiener filter coefficients (postfilter)
154 int dc_level; ///< Predicted amount of DC noise, based
155 ///< on which a DC removal filter is used
157 int lsps; ///< number of LSPs per frame [10 or 16]
158 int lsp_q_mode; ///< defines quantizer defaults [0, 1]
159 int lsp_def_mode; ///< defines different sets of LSP defaults
162 int min_pitch_val; ///< base value for pitch parsing code
163 int max_pitch_val; ///< max value + 1 for pitch parsing
164 int pitch_nbits; ///< number of bits used to specify the
165 ///< pitch value in the frame header
166 int block_pitch_nbits; ///< number of bits used to specify the
167 ///< first block's pitch value
168 int block_pitch_range; ///< range of the block pitch
169 int block_delta_pitch_nbits; ///< number of bits used to specify the
170 ///< delta pitch between this and the last
171 ///< block's pitch value, used in all but
173 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is
174 ///< from -this to +this-1)
175 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale
181 * @name Packet values specified in the packet header or related to a packet.
183 * A packet is considered to be a single unit of data provided to this
184 * decoder by the demuxer.
187 int spillover_nbits; ///< number of bits of the previous packet's
188 ///< last superframe preceding this
189 ///< packet's first full superframe (useful
190 ///< for re-synchronization also)
191 int has_residual_lsps; ///< if set, superframes contain one set of
192 ///< LSPs that cover all frames, encoded as
193 ///< independent and residual LSPs; if not
194 ///< set, each frame contains its own, fully
195 ///< independent, LSPs
196 int skip_bits_next; ///< number of bits to skip at the next call
197 ///< to #wmavoice_decode_packet() (since
198 ///< they're part of the previous superframe)
200 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + AV_INPUT_BUFFER_PADDING_SIZE];
201 ///< cache for superframe data split over
202 ///< multiple packets
203 int sframe_cache_size; ///< set to >0 if we have data from an
204 ///< (incomplete) superframe from a previous
205 ///< packet that spilled over in the current
206 ///< packet; specifies the amount of bits in
208 PutBitContext pb; ///< bitstream writer for #sframe_cache
213 * @name Frame and superframe values
214 * Superframe and frame data - these can change from frame to frame,
215 * although some of them do in that case serve as a cache / history for
216 * the next frame or superframe.
219 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
221 int last_pitch_val; ///< pitch value of the previous frame
222 int last_acb_type; ///< frame type [0-2] of the previous frame
223 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val)
224 ///< << 16) / #MAX_FRAMESIZE
225 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE
227 int aw_idx_is_ext; ///< whether the AW index was encoded in
228 ///< 8 bits (instead of 6)
229 int aw_pulse_range; ///< the range over which #aw_pulse_set1()
230 ///< can apply the pulse, relative to the
231 ///< value in aw_first_pulse_off. The exact
232 ///< position of the first AW-pulse is within
233 ///< [pulse_off, pulse_off + this], and
234 ///< depends on bitstream values; [16 or 24]
235 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note
236 ///< that this number can be negative (in
237 ///< which case it basically means "zero")
238 int aw_first_pulse_off[2]; ///< index of first sample to which to
239 ///< apply AW-pulses, or -0xff if unset
240 int aw_next_pulse_off_cache; ///< the position (relative to start of the
241 ///< second block) at which pulses should
242 ///< start to be positioned, serves as a
243 ///< cache for pitch-adaptive window pulses
246 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is
247 ///< only used for comfort noise in #pRNG()
248 int nb_superframes; ///< number of superframes in current packet
249 float gain_pred_err[6]; ///< cache for gain prediction
250 float excitation_history[MAX_SIGNAL_HISTORY];
251 ///< cache of the signal of previous
252 ///< superframes, used as a history for
253 ///< signal generation
254 float synth_history[MAX_LSPS]; ///< see #excitation_history
258 * @name Postfilter values
260 * Variables used for postfilter implementation, mostly history for
261 * smoothing and so on, and context variables for FFT/iFFT.
264 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the
265 ///< postfilter (for denoise filter)
266 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert
267 ///< transform, part of postfilter)
268 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi]
270 float postfilter_agc; ///< gain control memory, used in
271 ///< #adaptive_gain_control()
272 float dcf_mem[2]; ///< DC filter history
273 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE];
274 ///< zero filter output (i.e. excitation)
276 float denoise_filter_cache[MAX_FRAMESIZE];
277 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache
278 DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80];
279 ///< aligned buffer for LPC tilting
280 DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80];
281 ///< aligned buffer for denoise coefficients
282 DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16];
283 ///< aligned buffer for postfilter speech
291 * Set up the variable bit mode (VBM) tree from container extradata.
292 * @param gb bit I/O context.
293 * The bit context (s->gb) should be loaded with byte 23-46 of the
294 * container extradata (i.e. the ones containing the VBM tree).
295 * @param vbm_tree pointer to array to which the decoded VBM tree will be
297 * @return 0 on success, <0 on error.
299 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
301 int cntr[8] = { 0 }, n, res;
303 memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25);
304 for (n = 0; n < 17; n++) {
305 res = get_bits(gb, 3);
306 if (cntr[res] > 3) // should be >= 3 + (res == 7))
308 vbm_tree[res * 3 + cntr[res]++] = n;
313 static av_cold void wmavoice_init_static_data(AVCodec *codec)
315 static const uint8_t bits[] = {
318 10, 10, 10, 12, 12, 12,
321 static const uint16_t codes[] = {
322 0x0000, 0x0001, 0x0002, // 00/01/10
323 0x000c, 0x000d, 0x000e, // 11+00/01/10
324 0x003c, 0x003d, 0x003e, // 1111+00/01/10
325 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10
326 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10
327 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10
328 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx
331 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
332 bits, 1, 1, codes, 2, 2, 132);
335 static av_cold void wmavoice_flush(AVCodecContext *ctx)
337 WMAVoiceContext *s = ctx->priv_data;
340 s->postfilter_agc = 0;
341 s->sframe_cache_size = 0;
342 s->skip_bits_next = 0;
343 for (n = 0; n < s->lsps; n++)
344 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
345 memset(s->excitation_history, 0,
346 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY);
347 memset(s->synth_history, 0,
348 sizeof(*s->synth_history) * MAX_LSPS);
349 memset(s->gain_pred_err, 0,
350 sizeof(s->gain_pred_err));
353 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0,
354 sizeof(*s->synth_filter_out_buf) * s->lsps);
355 memset(s->dcf_mem, 0,
356 sizeof(*s->dcf_mem) * 2);
357 memset(s->zero_exc_pf, 0,
358 sizeof(*s->zero_exc_pf) * s->history_nsamples);
359 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache));
364 * Set up decoder with parameters from demuxer (extradata etc.).
366 static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
368 int n, flags, pitch_range, lsp16_flag;
369 WMAVoiceContext *s = ctx->priv_data;
373 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
374 * - byte 19-22: flags field (annoyingly in LE; see below for known
376 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
379 if (ctx->extradata_size != 46) {
380 av_log(ctx, AV_LOG_ERROR,
381 "Invalid extradata size %d (should be 46)\n",
382 ctx->extradata_size);
383 return AVERROR_INVALIDDATA;
385 if (ctx->block_align <= 0) {
386 av_log(ctx, AV_LOG_ERROR, "Invalid block alignment %d.\n", ctx->block_align);
387 return AVERROR_INVALIDDATA;
390 flags = AV_RL32(ctx->extradata + 18);
391 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
392 s->do_apf = flags & 0x1;
394 ff_rdft_init(&s->rdft, 7, DFT_R2C);
395 ff_rdft_init(&s->irdft, 7, IDFT_C2R);
396 ff_dct_init(&s->dct, 6, DCT_I);
397 ff_dct_init(&s->dst, 6, DST_I);
399 ff_sine_window_init(s->cos, 256);
400 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0]));
401 for (n = 0; n < 255; n++) {
402 s->sin[n] = -s->sin[510 - n];
403 s->cos[510 - n] = s->cos[n];
406 s->denoise_strength = (flags >> 2) & 0xF;
407 if (s->denoise_strength >= 12) {
408 av_log(ctx, AV_LOG_ERROR,
409 "Invalid denoise filter strength %d (max=11)\n",
410 s->denoise_strength);
411 return AVERROR_INVALIDDATA;
413 s->denoise_tilt_corr = !!(flags & 0x40);
414 s->dc_level = (flags >> 7) & 0xF;
415 s->lsp_q_mode = !!(flags & 0x2000);
416 s->lsp_def_mode = !!(flags & 0x4000);
417 lsp16_flag = flags & 0x1000;
423 for (n = 0; n < s->lsps; n++)
424 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
426 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3);
427 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) {
428 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n");
429 return AVERROR_INVALIDDATA;
432 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8;
433 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8;
434 pitch_range = s->max_pitch_val - s->min_pitch_val;
435 if (pitch_range <= 0) {
436 av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n");
437 return AVERROR_INVALIDDATA;
439 s->pitch_nbits = av_ceil_log2(pitch_range);
440 s->last_pitch_val = 40;
441 s->last_acb_type = ACB_TYPE_NONE;
442 s->history_nsamples = s->max_pitch_val + 8;
444 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) {
445 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8,
446 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8;
448 av_log(ctx, AV_LOG_ERROR,
449 "Unsupported samplerate %d (min=%d, max=%d)\n",
450 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz
452 return AVERROR(ENOSYS);
455 s->block_conv_table[0] = s->min_pitch_val;
456 s->block_conv_table[1] = (pitch_range * 25) >> 6;
457 s->block_conv_table[2] = (pitch_range * 44) >> 6;
458 s->block_conv_table[3] = s->max_pitch_val - 1;
459 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF;
460 if (s->block_delta_pitch_hrange <= 0) {
461 av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n");
462 return AVERROR_INVALIDDATA;
464 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange);
465 s->block_pitch_range = s->block_conv_table[2] +
466 s->block_conv_table[3] + 1 +
467 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val);
468 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range);
471 ctx->channel_layout = AV_CH_LAYOUT_MONO;
472 ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
478 * @name Postfilter functions
479 * Postfilter functions (gain control, wiener denoise filter, DC filter,
480 * kalman smoothening, plus surrounding code to wrap it)
484 * Adaptive gain control (as used in postfilter).
486 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except
487 * that the energy here is calculated using sum(abs(...)), whereas the
488 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)).
490 * @param out output buffer for filtered samples
491 * @param in input buffer containing the samples as they are after the
492 * postfilter steps so far
493 * @param speech_synth input buffer containing speech synth before postfilter
494 * @param size input buffer size
495 * @param alpha exponential filter factor
496 * @param gain_mem pointer to filter memory (single float)
498 static void adaptive_gain_control(float *out, const float *in,
499 const float *speech_synth,
500 int size, float alpha, float *gain_mem)
503 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
504 float mem = *gain_mem;
506 for (i = 0; i < size; i++) {
507 speech_energy += fabsf(speech_synth[i]);
508 postfilter_energy += fabsf(in[i]);
510 gain_scale_factor = postfilter_energy == 0.0 ? 0.0 :
511 (1.0 - alpha) * speech_energy / postfilter_energy;
513 for (i = 0; i < size; i++) {
514 mem = alpha * mem + gain_scale_factor;
515 out[i] = in[i] * mem;
522 * Kalman smoothing function.
524 * This function looks back pitch +/- 3 samples back into history to find
525 * the best fitting curve (that one giving the optimal gain of the two
526 * signals, i.e. the highest dot product between the two), and then
527 * uses that signal history to smoothen the output of the speech synthesis
530 * @param s WMA Voice decoding context
531 * @param pitch pitch of the speech signal
532 * @param in input speech signal
533 * @param out output pointer for smoothened signal
534 * @param size input/output buffer size
536 * @returns -1 if no smoothening took place, e.g. because no optimal
537 * fit could be found, or 0 on success.
539 static int kalman_smoothen(WMAVoiceContext *s, int pitch,
540 const float *in, float *out, int size)
543 float optimal_gain = 0, dot;
544 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)],
545 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)],
546 *best_hist_ptr = NULL;
548 /* find best fitting point in history */
550 dot = avpriv_scalarproduct_float_c(in, ptr, size);
551 if (dot > optimal_gain) {
555 } while (--ptr >= end);
557 if (optimal_gain <= 0)
559 dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
560 if (dot <= 0) // would be 1.0
563 if (optimal_gain <= dot) {
564 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
568 /* actual smoothing */
569 for (n = 0; n < size; n++)
570 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
576 * Get the tilt factor of a formant filter from its transfer function
577 * @see #tilt_factor() in amrnbdec.c, which does essentially the same,
578 * but somehow (??) it does a speech synthesis filter in the
579 * middle, which is missing here
581 * @param lpcs LPC coefficients
582 * @param n_lpcs Size of LPC buffer
583 * @returns the tilt factor
585 static float tilt_factor(const float *lpcs, int n_lpcs)
589 rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
590 rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
596 * Derive denoise filter coefficients (in real domain) from the LPCs.
598 static void calc_input_response(WMAVoiceContext *s, float *lpcs,
599 int fcb_type, float *coeffs, int remainder)
601 float last_coeff, min = 15.0, max = -15.0;
602 float irange, angle_mul, gain_mul, range, sq;
605 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */
606 s->rdft.rdft_calc(&s->rdft, lpcs);
607 #define log_range(var, assign) do { \
608 float tmp = log10f(assign); var = tmp; \
609 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \
611 log_range(last_coeff, lpcs[1] * lpcs[1]);
612 for (n = 1; n < 64; n++)
613 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] +
614 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]);
615 log_range(lpcs[0], lpcs[0] * lpcs[0]);
618 lpcs[64] = last_coeff;
620 /* Now, use this spectrum to pick out these frequencies with higher
621 * (relative) power/energy (which we then take to be "not noise"),
622 * and set up a table (still in lpc[]) of (relative) gains per frequency.
623 * These frequencies will be maintained, while others ("noise") will be
624 * decreased in the filter output. */
625 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63]
626 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) :
628 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
629 for (n = 0; n <= 64; n++) {
632 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1);
633 pwr = wmavoice_denoise_power_table[s->denoise_strength][idx];
634 lpcs[n] = angle_mul * pwr;
636 /* 70.57 =~ 1/log10(1.0331663) */
637 idx = (pwr * gain_mul - 0.0295) * 70.570526123;
638 if (idx > 127) { // fall back if index falls outside table range
639 coeffs[n] = wmavoice_energy_table[127] *
640 powf(1.0331663, idx - 127);
642 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
645 /* calculate the Hilbert transform of the gains, which we do (since this
646 * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
647 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
648 * "moment" of the LPCs in this filter. */
649 s->dct.dct_calc(&s->dct, lpcs);
650 s->dst.dct_calc(&s->dst, lpcs);
652 /* Split out the coefficient indexes into phase/magnitude pairs */
653 idx = 255 + av_clip(lpcs[64], -255, 255);
654 coeffs[0] = coeffs[0] * s->cos[idx];
655 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255);
656 last_coeff = coeffs[64] * s->cos[idx];
658 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255);
659 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
660 coeffs[n * 2] = coeffs[n] * s->cos[idx];
664 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255);
665 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx];
666 coeffs[n * 2] = coeffs[n] * s->cos[idx];
668 coeffs[1] = last_coeff;
670 /* move into real domain */
671 s->irdft.rdft_calc(&s->irdft, coeffs);
673 /* tilt correction and normalize scale */
674 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
675 if (s->denoise_tilt_corr) {
678 coeffs[remainder - 1] = 0;
679 ff_tilt_compensation(&tilt_mem,
680 -1.8 * tilt_factor(coeffs, remainder - 1),
683 sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
685 for (n = 0; n < remainder; n++)
690 * This function applies a Wiener filter on the (noisy) speech signal as
691 * a means to denoise it.
693 * - take RDFT of LPCs to get the power spectrum of the noise + speech;
694 * - using this power spectrum, calculate (for each frequency) the Wiener
695 * filter gain, which depends on the frequency power and desired level
696 * of noise subtraction (when set too high, this leads to artifacts)
697 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse
699 * - by doing a phase shift, calculate the Hilbert transform of this array
700 * of per-frequency filter-gains to get the filtering coefficients;
701 * - smoothen/normalize/de-tilt these filter coefficients as desired;
702 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT
703 * to get the denoised speech signal;
704 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond
705 * the frame boundary) are saved and applied to subsequent frames by an
706 * overlap-add method (otherwise you get clicking-artifacts).
708 * @param s WMA Voice decoding context
709 * @param fcb_type Frame (codebook) type
710 * @param synth_pf input: the noisy speech signal, output: denoised speech
711 * data; should be 16-byte aligned (for ASM purposes)
712 * @param size size of the speech data
713 * @param lpcs LPCs used to synthesize this frame's speech data
715 static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
716 float *synth_pf, int size,
719 int remainder, lim, n;
721 if (fcb_type != FCB_TYPE_SILENCE) {
722 float *tilted_lpcs = s->tilted_lpcs_pf,
723 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
725 tilted_lpcs[0] = 1.0;
726 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps);
727 memset(&tilted_lpcs[s->lsps + 1], 0,
728 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1));
729 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps),
730 tilted_lpcs, s->lsps + 2);
732 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame
733 * size is applied to the next frame. All input beyond this is zero,
734 * and thus all output beyond this will go towards zero, hence we can
735 * limit to min(size-1, 127-size) as a performance consideration. */
736 remainder = FFMIN(127 - size, size - 1);
737 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder);
739 /* apply coefficients (in frequency spectrum domain), i.e. complex
740 * number multiplication */
741 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size));
742 s->rdft.rdft_calc(&s->rdft, synth_pf);
743 s->rdft.rdft_calc(&s->rdft, coeffs);
744 synth_pf[0] *= coeffs[0];
745 synth_pf[1] *= coeffs[1];
746 for (n = 1; n < 64; n++) {
747 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1];
748 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1];
749 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1];
751 s->irdft.rdft_calc(&s->irdft, synth_pf);
754 /* merge filter output with the history of previous runs */
755 if (s->denoise_filter_cache_size) {
756 lim = FFMIN(s->denoise_filter_cache_size, size);
757 for (n = 0; n < lim; n++)
758 synth_pf[n] += s->denoise_filter_cache[n];
759 s->denoise_filter_cache_size -= lim;
760 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size],
761 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size);
764 /* move remainder of filter output into a cache for future runs */
765 if (fcb_type != FCB_TYPE_SILENCE) {
766 lim = FFMIN(remainder, s->denoise_filter_cache_size);
767 for (n = 0; n < lim; n++)
768 s->denoise_filter_cache[n] += synth_pf[size + n];
769 if (lim < remainder) {
770 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim],
771 sizeof(s->denoise_filter_cache[0]) * (remainder - lim));
772 s->denoise_filter_cache_size = remainder;
778 * Averaging projection filter, the postfilter used in WMAVoice.
780 * This uses the following steps:
781 * - A zero-synthesis filter (generate excitation from synth signal)
782 * - Kalman smoothing on excitation, based on pitch
783 * - Re-synthesized smoothened output
784 * - Iterative Wiener denoise filter
785 * - Adaptive gain filter
788 * @param s WMAVoice decoding context
789 * @param synth Speech synthesis output (before postfilter)
790 * @param samples Output buffer for filtered samples
791 * @param size Buffer size of synth & samples
792 * @param lpcs Generated LPCs used for speech synthesis
793 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned)
794 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses)
795 * @param pitch Pitch of the input signal
797 static void postfilter(WMAVoiceContext *s, const float *synth,
798 float *samples, int size,
799 const float *lpcs, float *zero_exc_pf,
800 int fcb_type, int pitch)
802 float synth_filter_in_buf[MAX_FRAMESIZE / 2],
803 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16],
804 *synth_filter_in = zero_exc_pf;
806 av_assert0(size <= MAX_FRAMESIZE / 2);
808 /* generate excitation from input signal */
809 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
811 if (fcb_type >= FCB_TYPE_AW_PULSES &&
812 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size))
813 synth_filter_in = synth_filter_in_buf;
815 /* re-synthesize speech after smoothening, and keep history */
816 ff_celp_lp_synthesis_filterf(synth_pf, lpcs,
817 synth_filter_in, size, s->lsps);
818 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps],
819 sizeof(synth_pf[0]) * s->lsps);
821 wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
823 adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
826 if (s->dc_level > 8) {
827 /* remove ultra-low frequency DC noise / highpass filter;
828 * coefficients are identical to those used in SIPR decoding,
829 * and very closely resemble those used in AMR-NB decoding. */
830 ff_acelp_apply_order_2_transfer_function(samples, samples,
831 (const float[2]) { -1.99997, 1.0 },
832 (const float[2]) { -1.9330735188, 0.93589198496 },
833 0.93980580475, s->dcf_mem, size);
842 * @param lsps output pointer to the array that will hold the LSPs
843 * @param num number of LSPs to be dequantized
844 * @param values quantized values, contains n_stages values
845 * @param sizes range (i.e. max value) of each quantized value
846 * @param n_stages number of dequantization runs
847 * @param table dequantization table to be used
848 * @param mul_q LSF multiplier
849 * @param base_q base (lowest) LSF values
851 static void dequant_lsps(double *lsps, int num,
852 const uint16_t *values,
853 const uint16_t *sizes,
854 int n_stages, const uint8_t *table,
856 const double *base_q)
860 memset(lsps, 0, num * sizeof(*lsps));
861 for (n = 0; n < n_stages; n++) {
862 const uint8_t *t_off = &table[values[n] * num];
863 double base = base_q[n], mul = mul_q[n];
865 for (m = 0; m < num; m++)
866 lsps[m] += base + mul * t_off[m];
868 table += sizes[n] * num;
873 * @name LSP dequantization routines
874 * LSP dequantization routines, for 10/16LSPs and independent/residual coding.
875 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits;
876 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits.
880 * Parse 10 independently-coded LSPs.
882 static void dequant_lsp10i(GetBitContext *gb, double *lsps)
884 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 };
885 static const double mul_lsf[4] = {
886 5.2187144800e-3, 1.4626986422e-3,
887 9.6179549166e-4, 1.1325736225e-3
889 static const double base_lsf[4] = {
890 M_PI * -2.15522e-1, M_PI * -6.1646e-2,
891 M_PI * -3.3486e-2, M_PI * -5.7408e-2
895 v[0] = get_bits(gb, 8);
896 v[1] = get_bits(gb, 6);
897 v[2] = get_bits(gb, 5);
898 v[3] = get_bits(gb, 5);
900 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
905 * Parse 10 independently-coded LSPs, and then derive the tables to
906 * generate LSPs for the other frames from them (residual coding).
908 static void dequant_lsp10r(GetBitContext *gb,
909 double *i_lsps, const double *old,
910 double *a1, double *a2, int q_mode)
912 static const uint16_t vec_sizes[3] = { 128, 64, 64 };
913 static const double mul_lsf[3] = {
914 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3
916 static const double base_lsf[3] = {
917 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
919 const float (*ipol_tab)[2][10] = q_mode ?
920 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
921 uint16_t interpol, v[3];
924 dequant_lsp10i(gb, i_lsps);
926 interpol = get_bits(gb, 5);
927 v[0] = get_bits(gb, 7);
928 v[1] = get_bits(gb, 6);
929 v[2] = get_bits(gb, 6);
931 for (n = 0; n < 10; n++) {
932 double delta = old[n] - i_lsps[n];
933 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
934 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
937 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
942 * Parse 16 independently-coded LSPs.
944 static void dequant_lsp16i(GetBitContext *gb, double *lsps)
946 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 };
947 static const double mul_lsf[5] = {
948 3.3439586280e-3, 6.9908173703e-4,
949 3.3216608306e-3, 1.0334960326e-3,
952 static const double base_lsf[5] = {
953 M_PI * -1.27576e-1, M_PI * -2.4292e-2,
954 M_PI * -1.28094e-1, M_PI * -3.2128e-2,
959 v[0] = get_bits(gb, 8);
960 v[1] = get_bits(gb, 6);
961 v[2] = get_bits(gb, 7);
962 v[3] = get_bits(gb, 6);
963 v[4] = get_bits(gb, 7);
965 dequant_lsps( lsps, 5, v, vec_sizes, 2,
966 wmavoice_dq_lsp16i1, mul_lsf, base_lsf);
967 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2,
968 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]);
969 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1,
970 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]);
974 * Parse 16 independently-coded LSPs, and then derive the tables to
975 * generate LSPs for the other frames from them (residual coding).
977 static void dequant_lsp16r(GetBitContext *gb,
978 double *i_lsps, const double *old,
979 double *a1, double *a2, int q_mode)
981 static const uint16_t vec_sizes[3] = { 128, 128, 128 };
982 static const double mul_lsf[3] = {
983 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3
985 static const double base_lsf[3] = {
986 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
988 const float (*ipol_tab)[2][16] = q_mode ?
989 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
990 uint16_t interpol, v[3];
993 dequant_lsp16i(gb, i_lsps);
995 interpol = get_bits(gb, 5);
996 v[0] = get_bits(gb, 7);
997 v[1] = get_bits(gb, 7);
998 v[2] = get_bits(gb, 7);
1000 for (n = 0; n < 16; n++) {
1001 double delta = old[n] - i_lsps[n];
1002 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n];
1003 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n];
1006 dequant_lsps( a2, 10, v, vec_sizes, 1,
1007 wmavoice_dq_lsp16r1, mul_lsf, base_lsf);
1008 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1,
1009 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]);
1010 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1,
1011 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]);
1016 * @name Pitch-adaptive window coding functions
1017 * The next few functions are for pitch-adaptive window coding.
1021 * Parse the offset of the first pitch-adaptive window pulses, and
1022 * the distribution of pulses between the two blocks in this frame.
1023 * @param s WMA Voice decoding context private data
1024 * @param gb bit I/O context
1025 * @param pitch pitch for each block in this frame
1027 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
1030 static const int16_t start_offset[94] = {
1031 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11,
1032 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26,
1033 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43,
1034 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67,
1035 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91,
1036 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115,
1037 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139,
1038 141, 143, 145, 147, 149, 151, 153, 155, 157, 159
1042 /* position of pulse */
1043 s->aw_idx_is_ext = 0;
1044 if ((bits = get_bits(gb, 6)) >= 54) {
1045 s->aw_idx_is_ext = 1;
1046 bits += (bits - 54) * 3 + get_bits(gb, 2);
1049 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count
1050 * the distribution of the pulses in each block contained in this frame. */
1051 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16;
1052 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ;
1053 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0];
1054 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2;
1055 offset += s->aw_n_pulses[0] * pitch[0];
1056 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1];
1057 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2;
1059 /* if continuing from a position before the block, reset position to
1060 * start of block (when corrected for the range over which it can be
1061 * spread in aw_pulse_set1()). */
1062 if (start_offset[bits] < MAX_FRAMESIZE / 2) {
1063 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0)
1064 s->aw_first_pulse_off[1] -= pitch[1];
1065 if (start_offset[bits] < 0)
1066 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0)
1067 s->aw_first_pulse_off[0] -= pitch[0];
1072 * Apply second set of pitch-adaptive window pulses.
1073 * @param s WMA Voice decoding context private data
1074 * @param gb bit I/O context
1075 * @param block_idx block index in frame [0, 1]
1076 * @param fcb structure containing fixed codebook vector info
1077 * @return -1 on error, 0 otherwise
1079 static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1080 int block_idx, AMRFixed *fcb)
1082 uint16_t use_mask_mem[9]; // only 5 are used, rest is padding
1083 uint16_t *use_mask = use_mask_mem + 2;
1084 /* in this function, idx is the index in the 80-bit (+ padding) use_mask
1085 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits
1086 * of idx are the position of the bit within a particular item in the
1087 * array (0 being the most significant bit, and 15 being the least
1088 * significant bit), and the remainder (>> 4) is the index in the
1089 * use_mask[]-array. This is faster and uses less memory than using a
1090 * 80-byte/80-int array. */
1091 int pulse_off = s->aw_first_pulse_off[block_idx],
1092 pulse_start, n, idx, range, aidx, start_off = 0;
1094 /* set offset of first pulse to within this block */
1095 if (s->aw_n_pulses[block_idx] > 0)
1096 while (pulse_off + s->aw_pulse_range < 1)
1097 pulse_off += fcb->pitch_lag;
1099 /* find range per pulse */
1100 if (s->aw_n_pulses[0] > 0) {
1101 if (block_idx == 0) {
1103 } else /* block_idx = 1 */ {
1105 if (s->aw_n_pulses[block_idx] > 0)
1106 pulse_off = s->aw_next_pulse_off_cache;
1110 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1112 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly,
1113 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus
1114 * we exclude that range from being pulsed again in this function. */
1115 memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0]));
1116 memset( use_mask, -1, 5 * sizeof(use_mask[0]));
1117 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0]));
1118 if (s->aw_n_pulses[block_idx] > 0)
1119 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) {
1120 int excl_range = s->aw_pulse_range; // always 16 or 24
1121 uint16_t *use_mask_ptr = &use_mask[idx >> 4];
1122 int first_sh = 16 - (idx & 15);
1123 *use_mask_ptr++ &= 0xFFFFu << first_sh;
1124 excl_range -= first_sh;
1125 if (excl_range >= 16) {
1126 *use_mask_ptr++ = 0;
1127 *use_mask_ptr &= 0xFFFF >> (excl_range - 16);
1129 *use_mask_ptr &= 0xFFFF >> excl_range;
1132 /* find the 'aidx'th offset that is not excluded */
1133 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4);
1134 for (n = 0; n <= aidx; pulse_start++) {
1135 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ;
1136 if (idx >= MAX_FRAMESIZE / 2) { // find from zero
1137 if (use_mask[0]) idx = 0x0F;
1138 else if (use_mask[1]) idx = 0x1F;
1139 else if (use_mask[2]) idx = 0x2F;
1140 else if (use_mask[3]) idx = 0x3F;
1141 else if (use_mask[4]) idx = 0x4F;
1143 idx -= av_log2_16bit(use_mask[idx >> 4]);
1145 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1146 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1152 fcb->x[fcb->n] = start_off;
1153 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1156 /* set offset for next block, relative to start of that block */
1157 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag;
1158 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0;
1163 * Apply first set of pitch-adaptive window pulses.
1164 * @param s WMA Voice decoding context private data
1165 * @param gb bit I/O context
1166 * @param block_idx block index in frame [0, 1]
1167 * @param fcb storage location for fixed codebook pulse info
1169 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1170 int block_idx, AMRFixed *fcb)
1172 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1175 if (s->aw_n_pulses[block_idx] > 0) {
1176 int n, v_mask, i_mask, sh, n_pulses;
1178 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1183 } else { // 4 pulses, 1:sign + 2:index each
1190 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) {
1191 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0;
1192 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n +
1193 s->aw_first_pulse_off[block_idx];
1194 while (fcb->x[fcb->n] < 0)
1195 fcb->x[fcb->n] += fcb->pitch_lag;
1196 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2)
1200 int num2 = (val & 0x1FF) >> 1, delta, idx;
1202 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; }
1203 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; }
1204 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; }
1205 else { delta = 7; idx = num2 + 1 - 3 * 75; }
1206 v = (val & 0x200) ? -1.0 : 1.0;
1208 fcb->no_repeat_mask |= 3 << fcb->n;
1209 fcb->x[fcb->n] = idx - delta;
1211 fcb->x[fcb->n + 1] = idx;
1212 fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1220 * Generate a random number from frame_cntr and block_idx, which will live
1221 * in the range [0, 1000 - block_size] (so it can be used as an index in a
1222 * table of size 1000 of which you want to read block_size entries).
1224 * @param frame_cntr current frame number
1225 * @param block_num current block index
1226 * @param block_size amount of entries we want to read from a table
1227 * that has 1000 entries
1228 * @return a (non-)random number in the [0, 1000 - block_size] range.
1230 static int pRNG(int frame_cntr, int block_num, int block_size)
1232 /* array to simplify the calculation of z:
1233 * y = (x % 9) * 5 + 6;
1234 * z = (49995 * x) / y;
1235 * Since y only has 9 values, we can remove the division by using a
1236 * LUT and using FASTDIV-style divisions. For each of the 9 values
1237 * of y, we can rewrite z as:
1238 * z = x * (49995 / y) + x * ((49995 % y) / y)
1239 * In this table, each col represents one possible value of y, the
1240 * first number is 49995 / y, and the second is the FASTDIV variant
1241 * of 49995 % y / y. */
1242 static const unsigned int div_tbl[9][2] = {
1243 { 8332, 3 * 715827883U }, // y = 6
1244 { 4545, 0 * 390451573U }, // y = 11
1245 { 3124, 11 * 268435456U }, // y = 16
1246 { 2380, 15 * 204522253U }, // y = 21
1247 { 1922, 23 * 165191050U }, // y = 26
1248 { 1612, 23 * 138547333U }, // y = 31
1249 { 1388, 27 * 119304648U }, // y = 36
1250 { 1219, 16 * 104755300U }, // y = 41
1251 { 1086, 39 * 93368855U } // y = 46
1253 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr;
1254 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6,
1255 // so this is effectively a modulo (%)
1256 y = x - 9 * MULH(477218589, x); // x % 9
1257 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1]));
1258 // z = x * 49995 / (y * 5 + 6)
1259 return z % (1000 - block_size);
1263 * Parse hardcoded signal for a single block.
1264 * @note see #synth_block().
1266 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1267 int block_idx, int size,
1268 const struct frame_type_desc *frame_desc,
1274 av_assert0(size <= MAX_FRAMESIZE);
1276 /* Set the offset from which we start reading wmavoice_std_codebook */
1277 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) {
1278 r_idx = pRNG(s->frame_cntr, block_idx, size);
1279 gain = s->silence_gain;
1280 } else /* FCB_TYPE_HARDCODED */ {
1281 r_idx = get_bits(gb, 8);
1282 gain = wmavoice_gain_universal[get_bits(gb, 6)];
1285 /* Clear gain prediction parameters */
1286 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1288 /* Apply gain to hardcoded codebook and use that as excitation signal */
1289 for (n = 0; n < size; n++)
1290 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain;
1294 * Parse FCB/ACB signal for a single block.
1295 * @note see #synth_block().
1297 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
1298 int block_idx, int size,
1299 int block_pitch_sh2,
1300 const struct frame_type_desc *frame_desc,
1303 static const float gain_coeff[6] = {
1304 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1306 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1307 int n, idx, gain_weight;
1310 av_assert0(size <= MAX_FRAMESIZE / 2);
1311 memset(pulses, 0, sizeof(*pulses) * size);
1313 fcb.pitch_lag = block_pitch_sh2 >> 2;
1314 fcb.pitch_fac = 1.0;
1315 fcb.no_repeat_mask = 0;
1318 /* For the other frame types, this is where we apply the innovation
1319 * (fixed) codebook pulses of the speech signal. */
1320 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) {
1321 aw_pulse_set1(s, gb, block_idx, &fcb);
1322 if (aw_pulse_set2(s, gb, block_idx, &fcb)) {
1323 /* Conceal the block with silence and return.
1324 * Skip the correct amount of bits to read the next
1325 * block from the correct offset. */
1326 int r_idx = pRNG(s->frame_cntr, block_idx, size);
1328 for (n = 0; n < size; n++)
1330 wmavoice_std_codebook[r_idx + n] * s->silence_gain;
1331 skip_bits(gb, 7 + 1);
1334 } else /* FCB_TYPE_EXC_PULSES */ {
1335 int offset_nbits = 5 - frame_desc->log_n_blocks;
1337 fcb.no_repeat_mask = -1;
1338 /* similar to ff_decode_10_pulses_35bits(), but with single pulses
1339 * (instead of double) for a subset of pulses */
1340 for (n = 0; n < 5; n++) {
1344 sign = get_bits1(gb) ? 1.0 : -1.0;
1345 pos1 = get_bits(gb, offset_nbits);
1346 fcb.x[fcb.n] = n + 5 * pos1;
1347 fcb.y[fcb.n++] = sign;
1348 if (n < frame_desc->dbl_pulses) {
1349 pos2 = get_bits(gb, offset_nbits);
1350 fcb.x[fcb.n] = n + 5 * pos2;
1351 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign;
1355 ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1357 /* Calculate gain for adaptive & fixed codebook signal.
1358 * see ff_amr_set_fixed_gain(). */
1359 idx = get_bits(gb, 7);
1360 fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
1362 5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
1363 acb_gain = wmavoice_gain_codebook_acb[idx];
1364 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
1365 -2.9957322736 /* log(0.05) */,
1366 1.6094379124 /* log(5.0) */);
1368 gain_weight = 8 >> frame_desc->log_n_blocks;
1369 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err,
1370 sizeof(*s->gain_pred_err) * (6 - gain_weight));
1371 for (n = 0; n < gain_weight; n++)
1372 s->gain_pred_err[n] = pred_err;
1374 /* Calculation of adaptive codebook */
1375 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1377 for (n = 0; n < size; n += len) {
1379 int abs_idx = block_idx * size + n;
1380 int pitch_sh16 = (s->last_pitch_val << 16) +
1381 s->pitch_diff_sh16 * abs_idx;
1382 int pitch = (pitch_sh16 + 0x6FFF) >> 16;
1383 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000;
1384 idx = idx_sh16 >> 16;
1385 if (s->pitch_diff_sh16) {
1386 if (s->pitch_diff_sh16 > 0) {
1387 next_idx_sh16 = (idx_sh16) &~ 0xFFFF;
1389 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1390 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1395 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1396 wmavoice_ipol1_coeffs, 17,
1399 } else /* ACB_TYPE_HAMMING */ {
1400 int block_pitch = block_pitch_sh2 >> 2;
1401 idx = block_pitch_sh2 & 3;
1403 ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1404 wmavoice_ipol2_coeffs, 4,
1407 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1408 sizeof(float) * size);
1411 /* Interpolate ACB/FCB and use as excitation signal */
1412 ff_weighted_vector_sumf(excitation, excitation, pulses,
1413 acb_gain, fcb_gain, size);
1417 * Parse data in a single block.
1419 * @param s WMA Voice decoding context private data
1420 * @param gb bit I/O context
1421 * @param block_idx index of the to-be-read block
1422 * @param size amount of samples to be read in this block
1423 * @param block_pitch_sh2 pitch for this block << 2
1424 * @param lsps LSPs for (the end of) this frame
1425 * @param prev_lsps LSPs for the last frame
1426 * @param frame_desc frame type descriptor
1427 * @param excitation target memory for the ACB+FCB interpolated signal
1428 * @param synth target memory for the speech synthesis filter output
1429 * @return 0 on success, <0 on error.
1431 static void synth_block(WMAVoiceContext *s, GetBitContext *gb,
1432 int block_idx, int size,
1433 int block_pitch_sh2,
1434 const double *lsps, const double *prev_lsps,
1435 const struct frame_type_desc *frame_desc,
1436 float *excitation, float *synth)
1438 double i_lsps[MAX_LSPS];
1439 float lpcs[MAX_LSPS];
1443 if (frame_desc->acb_type == ACB_TYPE_NONE)
1444 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1446 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1447 frame_desc, excitation);
1449 /* convert interpolated LSPs to LPCs */
1450 fac = (block_idx + 0.5) / frame_desc->n_blocks;
1451 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1452 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n]));
1453 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1455 /* Speech synthesis */
1456 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1460 * Synthesize output samples for a single frame.
1462 * @param ctx WMA Voice decoder context
1463 * @param gb bit I/O context (s->gb or one for cross-packet superframes)
1464 * @param frame_idx Frame number within superframe [0-2]
1465 * @param samples pointer to output sample buffer, has space for at least 160
1467 * @param lsps LSP array
1468 * @param prev_lsps array of previous frame's LSPs
1469 * @param excitation target buffer for excitation signal
1470 * @param synth target buffer for synthesized speech data
1471 * @return 0 on success, <0 on error.
1473 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1475 const double *lsps, const double *prev_lsps,
1476 float *excitation, float *synth)
1478 WMAVoiceContext *s = ctx->priv_data;
1479 int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val);
1480 int pitch[MAX_BLOCKS], av_uninit(last_block_pitch);
1482 /* Parse frame type ("frame header"), see frame_descs */
1483 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples;
1486 av_log(ctx, AV_LOG_ERROR,
1487 "Invalid frame type VLC code, skipping\n");
1488 return AVERROR_INVALIDDATA;
1491 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1493 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */
1494 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) {
1495 /* Pitch is provided per frame, which is interpreted as the pitch of
1496 * the last sample of the last block of this frame. We can interpolate
1497 * the pitch of other blocks (and even pitch-per-sample) by gradually
1498 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */
1499 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1;
1500 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1;
1501 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits);
1502 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1);
1503 if (s->last_acb_type == ACB_TYPE_NONE ||
1504 20 * abs(cur_pitch_val - s->last_pitch_val) >
1505 (cur_pitch_val + s->last_pitch_val))
1506 s->last_pitch_val = cur_pitch_val;
1508 /* pitch per block */
1509 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1510 int fac = n * 2 + 1;
1512 pitch[n] = (MUL16(fac, cur_pitch_val) +
1513 MUL16((n_blocks_x2 - fac), s->last_pitch_val) +
1514 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2;
1517 /* "pitch-diff-per-sample" for calculation of pitch per sample */
1518 s->pitch_diff_sh16 =
1519 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE;
1522 /* Global gain (if silence) and pitch-adaptive window coordinates */
1523 switch (frame_descs[bd_idx].fcb_type) {
1524 case FCB_TYPE_SILENCE:
1525 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)];
1527 case FCB_TYPE_AW_PULSES:
1528 aw_parse_coords(s, gb, pitch);
1532 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1535 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */
1536 switch (frame_descs[bd_idx].acb_type) {
1537 case ACB_TYPE_HAMMING: {
1538 /* Pitch is given per block. Per-block pitches are encoded as an
1539 * absolute value for the first block, and then delta values
1540 * relative to this value) for all subsequent blocks. The scale of
1541 * this pitch value is semi-logarithmic compared to its use in the
1542 * decoder, so we convert it to normal scale also. */
1544 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2,
1545 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1,
1546 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1;
1549 block_pitch = get_bits(gb, s->block_pitch_nbits);
1551 block_pitch = last_block_pitch - s->block_delta_pitch_hrange +
1552 get_bits(gb, s->block_delta_pitch_nbits);
1553 /* Convert last_ so that any next delta is within _range */
1554 last_block_pitch = av_clip(block_pitch,
1555 s->block_delta_pitch_hrange,
1556 s->block_pitch_range -
1557 s->block_delta_pitch_hrange);
1559 /* Convert semi-log-style scale back to normal scale */
1560 if (block_pitch < t1) {
1561 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch;
1564 if (block_pitch < t2) {
1566 (s->block_conv_table[1] << 2) + (block_pitch << 1);
1569 if (block_pitch < t3) {
1571 (s->block_conv_table[2] + block_pitch) << 2;
1573 bl_pitch_sh2 = s->block_conv_table[3] << 2;
1576 pitch[n] = bl_pitch_sh2 >> 2;
1580 case ACB_TYPE_ASYMMETRIC: {
1581 bl_pitch_sh2 = pitch[n] << 2;
1585 default: // ACB_TYPE_NONE has no pitch
1590 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2,
1591 lsps, prev_lsps, &frame_descs[bd_idx],
1592 &excitation[n * block_nsamples],
1593 &synth[n * block_nsamples]);
1596 /* Averaging projection filter, if applicable. Else, just copy samples
1597 * from synthesis buffer */
1599 double i_lsps[MAX_LSPS];
1600 float lpcs[MAX_LSPS];
1602 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1603 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n]));
1604 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1605 postfilter(s, synth, samples, 80, lpcs,
1606 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx],
1607 frame_descs[bd_idx].fcb_type, pitch[0]);
1609 for (n = 0; n < s->lsps; n++) // LSF -> LSP
1610 i_lsps[n] = cos(lsps[n]);
1611 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1);
1612 postfilter(s, &synth[80], &samples[80], 80, lpcs,
1613 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80],
1614 frame_descs[bd_idx].fcb_type, pitch[0]);
1616 memcpy(samples, synth, 160 * sizeof(synth[0]));
1618 /* Cache values for next frame */
1620 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%)
1621 s->last_acb_type = frame_descs[bd_idx].acb_type;
1622 switch (frame_descs[bd_idx].acb_type) {
1624 s->last_pitch_val = 0;
1626 case ACB_TYPE_ASYMMETRIC:
1627 s->last_pitch_val = cur_pitch_val;
1629 case ACB_TYPE_HAMMING:
1630 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1638 * Ensure minimum value for first item, maximum value for last value,
1639 * proper spacing between each value and proper ordering.
1641 * @param lsps array of LSPs
1642 * @param num size of LSP array
1644 * @note basically a double version of #ff_acelp_reorder_lsf(), might be
1645 * useful to put in a generic location later on. Parts are also
1646 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(),
1647 * which is in float.
1649 static void stabilize_lsps(double *lsps, int num)
1653 /* set minimum value for first, maximum value for last and minimum
1654 * spacing between LSF values.
1655 * Very similar to ff_set_min_dist_lsf(), but in double. */
1656 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI);
1657 for (n = 1; n < num; n++)
1658 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI);
1659 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI);
1661 /* reorder (looks like one-time / non-recursed bubblesort).
1662 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */
1663 for (n = 1; n < num; n++) {
1664 if (lsps[n] < lsps[n - 1]) {
1665 for (m = 1; m < num; m++) {
1666 double tmp = lsps[m];
1667 for (l = m - 1; l >= 0; l--) {
1668 if (lsps[l] <= tmp) break;
1669 lsps[l + 1] = lsps[l];
1679 * Synthesize output samples for a single superframe. If we have any data
1680 * cached in s->sframe_cache, that will be used instead of whatever is loaded
1683 * WMA Voice superframes contain 3 frames, each containing 160 audio samples,
1684 * to give a total of 480 samples per frame. See #synth_frame() for frame
1685 * parsing. In addition to 3 frames, superframes can also contain the LSPs
1686 * (if these are globally specified for all frames (residually); they can
1687 * also be specified individually per-frame. See the s->has_residual_lsps
1688 * option), and can specify the number of samples encoded in this superframe
1689 * (if less than 480), usually used to prevent blanks at track boundaries.
1691 * @param ctx WMA Voice decoder context
1692 * @return 0 on success, <0 on error or 1 if there was not enough data to
1693 * fully parse the superframe
1695 static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
1698 WMAVoiceContext *s = ctx->priv_data;
1699 GetBitContext *gb = &s->gb, s_gb;
1700 int n, res, n_samples = MAX_SFRAMESIZE;
1701 double lsps[MAX_FRAMES][MAX_LSPS];
1702 const double *mean_lsf = s->lsps == 16 ?
1703 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode];
1704 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12];
1705 float synth[MAX_LSPS + MAX_SFRAMESIZE];
1708 memcpy(synth, s->synth_history,
1709 s->lsps * sizeof(*synth));
1710 memcpy(excitation, s->excitation_history,
1711 s->history_nsamples * sizeof(*excitation));
1713 if (s->sframe_cache_size > 0) {
1715 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1716 s->sframe_cache_size = 0;
1719 /* First bit is speech/music bit, it differentiates between WMAVoice
1720 * speech samples (the actual codec) and WMAVoice music samples, which
1721 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in
1723 if (!get_bits1(gb)) {
1724 avpriv_request_sample(ctx, "WMAPro-in-WMAVoice");
1725 return AVERROR_PATCHWELCOME;
1728 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */
1729 if (get_bits1(gb)) {
1730 if ((n_samples = get_bits(gb, 12)) > MAX_SFRAMESIZE) {
1731 av_log(ctx, AV_LOG_ERROR,
1732 "Superframe encodes > %d samples (%d), not allowed\n",
1733 MAX_SFRAMESIZE, n_samples);
1734 return AVERROR_INVALIDDATA;
1738 /* Parse LSPs, if global for the superframe (can also be per-frame). */
1739 if (s->has_residual_lsps) {
1740 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2];
1742 for (n = 0; n < s->lsps; n++)
1743 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1745 if (s->lsps == 10) {
1746 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1747 } else /* s->lsps == 16 */
1748 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode);
1750 for (n = 0; n < s->lsps; n++) {
1751 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]);
1752 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]);
1753 lsps[2][n] += mean_lsf[n];
1755 for (n = 0; n < 3; n++)
1756 stabilize_lsps(lsps[n], s->lsps);
1759 /* get output buffer */
1760 frame->nb_samples = MAX_SFRAMESIZE;
1761 if ((res = ff_get_buffer(ctx, frame, 0)) < 0)
1763 frame->nb_samples = n_samples;
1764 samples = (float *)frame->data[0];
1766 /* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1767 for (n = 0; n < 3; n++) {
1768 if (!s->has_residual_lsps) {
1771 if (s->lsps == 10) {
1772 dequant_lsp10i(gb, lsps[n]);
1773 } else /* s->lsps == 16 */
1774 dequant_lsp16i(gb, lsps[n]);
1776 for (m = 0; m < s->lsps; m++)
1777 lsps[n][m] += mean_lsf[m];
1778 stabilize_lsps(lsps[n], s->lsps);
1781 if ((res = synth_frame(ctx, gb, n,
1782 &samples[n * MAX_FRAMESIZE],
1783 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1],
1784 &excitation[s->history_nsamples + n * MAX_FRAMESIZE],
1785 &synth[s->lsps + n * MAX_FRAMESIZE]))) {
1791 /* Statistics? FIXME - we don't check for length, a slight overrun
1792 * will be caught by internal buffer padding, and anything else
1793 * will be skipped, not read. */
1794 if (get_bits1(gb)) {
1795 res = get_bits(gb, 4);
1796 skip_bits(gb, 10 * (res + 1));
1799 if (get_bits_left(gb) < 0) {
1800 wmavoice_flush(ctx);
1801 return AVERROR_INVALIDDATA;
1806 /* Update history */
1807 memcpy(s->prev_lsps, lsps[2],
1808 s->lsps * sizeof(*s->prev_lsps));
1809 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE],
1810 s->lsps * sizeof(*synth));
1811 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE],
1812 s->history_nsamples * sizeof(*excitation));
1814 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1815 s->history_nsamples * sizeof(*s->zero_exc_pf));
1821 * Parse the packet header at the start of each packet (input data to this
1824 * @param s WMA Voice decoding context private data
1825 * @return <0 on error, nb_superframes on success.
1827 static int parse_packet_header(WMAVoiceContext *s)
1829 GetBitContext *gb = &s->gb;
1830 unsigned int res, n_superframes = 0;
1832 skip_bits(gb, 4); // packet sequence number
1833 s->has_residual_lsps = get_bits1(gb);
1835 res = get_bits(gb, 6); // number of superframes per packet
1836 // (minus first one if there is spillover)
1837 n_superframes += res;
1838 } while (res == 0x3F);
1839 s->spillover_nbits = get_bits(gb, s->spillover_bitsize);
1841 return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA;
1845 * Copy (unaligned) bits from gb/data/size to pb.
1847 * @param pb target buffer to copy bits into
1848 * @param data source buffer to copy bits from
1849 * @param size size of the source data, in bytes
1850 * @param gb bit I/O context specifying the current position in the source.
1851 * data. This function might use this to align the bit position to
1852 * a whole-byte boundary before calling #avpriv_copy_bits() on aligned
1854 * @param nbits the amount of bits to copy from source to target
1856 * @note after calling this function, the current position in the input bit
1857 * I/O context is undefined.
1859 static void copy_bits(PutBitContext *pb,
1860 const uint8_t *data, int size,
1861 GetBitContext *gb, int nbits)
1863 int rmn_bytes, rmn_bits;
1865 rmn_bits = rmn_bytes = get_bits_left(gb);
1866 if (rmn_bits < nbits)
1868 if (nbits > pb->size_in_bits - put_bits_count(pb))
1870 rmn_bits &= 7; rmn_bytes >>= 3;
1871 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0)
1872 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits));
1873 avpriv_copy_bits(pb, data + size - rmn_bytes,
1874 FFMIN(nbits - rmn_bits, rmn_bytes << 3));
1878 * Packet decoding: a packet is anything that the (ASF) demuxer contains,
1879 * and we expect that the demuxer / application provides it to us as such
1880 * (else you'll probably get garbage as output). Every packet has a size of
1881 * ctx->block_align bytes, starts with a packet header (see
1882 * #parse_packet_header()), and then a series of superframes. Superframe
1883 * boundaries may exceed packets, i.e. superframes can split data over
1884 * multiple (two) packets.
1886 * For more information about frames, see #synth_superframe().
1888 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1889 int *got_frame_ptr, AVPacket *avpkt)
1891 WMAVoiceContext *s = ctx->priv_data;
1892 GetBitContext *gb = &s->gb;
1895 /* Packets are sometimes a multiple of ctx->block_align, with a packet
1896 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer
1897 * feeds us ASF packets, which may concatenate multiple "codec" packets
1898 * in a single "muxer" packet, so we artificially emulate that by
1899 * capping the packet size at ctx->block_align. */
1900 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align);
1901 init_get_bits(&s->gb, avpkt->data, size << 3);
1903 /* size == ctx->block_align is used to indicate whether we are dealing with
1904 * a new packet or a packet of which we already read the packet header
1906 if (!(size % ctx->block_align)) { // new packet header
1908 s->spillover_nbits = 0;
1909 s->nb_superframes = 0;
1911 if ((res = parse_packet_header(s)) < 0)
1913 s->nb_superframes = res;
1916 /* If the packet header specifies a s->spillover_nbits, then we want
1917 * to push out all data of the previous packet (+ spillover) before
1918 * continuing to parse new superframes in the current packet. */
1919 if (s->sframe_cache_size > 0) {
1920 int cnt = get_bits_count(gb);
1921 if (cnt + s->spillover_nbits > avpkt->size * 8) {
1922 s->spillover_nbits = avpkt->size * 8 - cnt;
1924 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits);
1925 flush_put_bits(&s->pb);
1926 s->sframe_cache_size += s->spillover_nbits;
1927 if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 &&
1929 cnt += s->spillover_nbits;
1930 s->skip_bits_next = cnt & 7;
1934 skip_bits_long (gb, s->spillover_nbits - cnt +
1935 get_bits_count(gb)); // resync
1936 } else if (s->spillover_nbits) {
1937 skip_bits_long(gb, s->spillover_nbits); // resync
1939 } else if (s->skip_bits_next)
1940 skip_bits(gb, s->skip_bits_next);
1942 /* Try parsing superframes in current packet */
1943 s->sframe_cache_size = 0;
1944 s->skip_bits_next = 0;
1945 pos = get_bits_left(gb);
1946 if (s->nb_superframes-- == 0) {
1949 } else if (s->nb_superframes > 0) {
1950 if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
1952 } else if (*got_frame_ptr) {
1953 int cnt = get_bits_count(gb);
1954 s->skip_bits_next = cnt & 7;
1958 } else if ((s->sframe_cache_size = pos) > 0) {
1959 /* ... cache it for spillover in next packet */
1960 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE);
1961 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size);
1962 // FIXME bad - just copy bytes as whole and add use the
1963 // skip_bits_next field
1969 static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
1971 WMAVoiceContext *s = ctx->priv_data;
1974 ff_rdft_end(&s->rdft);
1975 ff_rdft_end(&s->irdft);
1976 ff_dct_end(&s->dct);
1977 ff_dct_end(&s->dst);
1983 AVCodec ff_wmavoice_decoder = {
1985 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"),
1986 .type = AVMEDIA_TYPE_AUDIO,
1987 .id = AV_CODEC_ID_WMAVOICE,
1988 .priv_data_size = sizeof(WMAVoiceContext),
1989 .init = wmavoice_decode_init,
1990 .init_static_data = wmavoice_init_static_data,
1991 .close = wmavoice_decode_end,
1992 .decode = wmavoice_decode_packet,
1993 .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
1994 .flush = wmavoice_flush,