path: root/libavcodec/wmavoice.c (plain)
blob: 2ec4499981f52a318ec3ad714a4a8367e5b2beba

1	/*
2	* Windows Media Audio Voice decoder.
3	* Copyright (c) 2009 Ronald S. Bultje
4	*
5	* This file is part of FFmpeg.
6	*
7	* FFmpeg is free software; you can redistribute it and/or
8	* modify it under the terms of the GNU Lesser General Public
9	* License as published by the Free Software Foundation; either
10	* version 2.1 of the License, or (at your option) any later version.
11	*
12	* FFmpeg is distributed in the hope that it will be useful,
13	* but WITHOUT ANY WARRANTY; without even the implied warranty of
14	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15	* Lesser General Public License for more details.
16	*
17	* You should have received a copy of the GNU Lesser General Public
18	* License along with FFmpeg; if not, write to the Free Software
19	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20	*/
21
22	/**
23	* @file
24	* @brief Windows Media Audio Voice compatible decoder
25	* @author Ronald S. Bultje <rsbultje@gmail.com>
26	*/
27
28	#include <math.h>
29
30	#include "libavutil/channel_layout.h"
31	#include "libavutil/float_dsp.h"
32	#include "libavutil/mem.h"
33	#include "avcodec.h"
34	#include "internal.h"
35	#include "get_bits.h"
36	#include "put_bits.h"
37	#include "wmavoice_data.h"
38	#include "celp_filters.h"
39	#include "acelp_vectors.h"
40	#include "acelp_filters.h"
41	#include "lsp.h"
42	#include "dct.h"
43	#include "rdft.h"
44	#include "sinewin.h"
45
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
58
59	/**
60	* Frame type VLC coding.
61	*/
62	static VLC frame_type_vlc;
63
64	/**
65	* Adaptive codebook types.
66	*/
67	enum {
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
72	///< window function
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
77	};
78
79	/**
80	* Fixed codebook types.
81	*/
82	enum {
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
87	///< gain values
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
92	///< pulse pairs
93	};
94
95	/**
96	* Description of frame types.
97	*/
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 }
125	};
126
127	/**
128	* WMA Voice decoding context.
129	*/
130	typedef struct WMAVoiceContext {
131	/**
132	* @name Global values specified in the stream header / extradata or used all over.
133	* @{
134	*/
135	GetBitContext gb; ///< packet bitreader. During decoder init,
136	///< it contains the extradata from the
137	///< demuxer. During decoding, it contains
138	///< packet data.
139	int8_t vbm_tree[25]; ///< converts VLC codes to frame type
140
141	int spillover_bitsize; ///< number of bits used to specify
142	///< #spillover_nbits in the packet header
143	///< = ceil(log2(ctx->block_align << 3))
144	int history_nsamples; ///< number of samples in history for signal
145	///< prediction (through ACB)
146
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
151	///< [0-11]
152	int denoise_tilt_corr; ///< Whether to apply tilt correction to the
153	///< Wiener filter coefficients (postfilter)
154	int dc_level; ///< Predicted amount of DC noise, based
155	///< on which a DC removal filter is used
156
157	int lsps; ///< number of LSPs per frame [10 or 16]
158	int lsp_q_mode; ///< defines quantizer defaults [0, 1]
159	int lsp_def_mode; ///< defines different sets of LSP defaults
160	///< [0, 1]
161
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
172	///< first block
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
176	///< conversion
177
178	/**
179	* @}
180	*
181	* @name Packet values specified in the packet header or related to a packet.
182	*
183	* A packet is considered to be a single unit of data provided to this
184	* decoder by the demuxer.
185	* @{
186	*/
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)
199
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
207	///< #sframe_cache
208	PutBitContext pb; ///< bitstream writer for #sframe_cache
209
210	/**
211	* @}
212	*
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.
217	* @{
218	*/
219	double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous
220	///< superframe
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
226
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
244	///< between blocks
245
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
255	/**
256	* @}
257	*
258	* @name Postfilter values
259	*
260	* Variables used for postfilter implementation, mostly history for
261	* smoothing and so on, and context variables for FFT/iFFT.
262	* @{
263	*/
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]
269	///< range
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)
275	///< by postfilter
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
284	///< synthesis
285	/**
286	* @}
287	*/
288	} WMAVoiceContext;
289
290	/**
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
296	* written.
297	* @return 0 on success, <0 on error.
298	*/
299	static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25])
300	{
301	int cntr[8] = { 0 }, n, res;
302
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))
307	return -1;
308	vbm_tree[res * 3 + cntr[res]++] = n;
309	}
310	return 0;
311	}
312
313	static av_cold void wmavoice_init_static_data(AVCodec *codec)
314	{
315	static const uint8_t bits[] = {
316	2, 2, 2, 4, 4, 4,
317	6, 6, 6, 8, 8, 8,
318	10, 10, 10, 12, 12, 12,
319	14, 14, 14, 14
320	};
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
329	};
330
331	INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits),
332	bits, 1, 1, codes, 2, 2, 132);
333	}
334
335	static av_cold void wmavoice_flush(AVCodecContext *ctx)
336	{
337	WMAVoiceContext *s = ctx->priv_data;
338	int n;
339
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));
351
352	if (s->do_apf) {
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));
360	}
361	}
362
363	/**
364	* Set up decoder with parameters from demuxer (extradata etc.).
365	*/
366	static av_cold int wmavoice_decode_init(AVCodecContext *ctx)
367	{
368	int n, flags, pitch_range, lsp16_flag;
369	WMAVoiceContext *s = ctx->priv_data;
370
371	/**
372	* Extradata layout:
373	* - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c),
374	* - byte 19-22: flags field (annoyingly in LE; see below for known
375	* values),
376	* - byte 23-46: variable bitmode tree (really just 17 * 3 bits,
377	* rest is 0).
378	*/
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;
384	}
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;
388	}
389
390	flags = AV_RL32(ctx->extradata + 18);
391	s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align);
392	s->do_apf = flags & 0x1;
393	if (s->do_apf) {
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);
398
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];
404	}
405	}
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;
412	}
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;
418	if (lsp16_flag) {
419	s->lsps = 16;
420	} else {
421	s->lsps = 10;
422	}
423	for (n = 0; n < s->lsps; n++)
424	s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0);
425
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;
430	}
431
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;
438	}
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;
443
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;
447
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
451
452	return AVERROR(ENOSYS);
453	}
454
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;
463	}
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);
469
470	ctx->channels = 1;
471	ctx->channel_layout = AV_CH_LAYOUT_MONO;
472	ctx->sample_fmt = AV_SAMPLE_FMT_FLT;
473
474	return 0;
475	}
476
477	/**
478	* @name Postfilter functions
479	* Postfilter functions (gain control, wiener denoise filter, DC filter,
480	* kalman smoothening, plus surrounding code to wrap it)
481	* @{
482	*/
483	/**
484	* Adaptive gain control (as used in postfilter).
485	*
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(...)).
489	*
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)
497	*/
498	static void adaptive_gain_control(float *out, const float *in,
499	const float *speech_synth,
500	int size, float alpha, float *gain_mem)
501	{
502	int i;
503	float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor;
504	float mem = *gain_mem;
505
506	for (i = 0; i < size; i++) {
507	speech_energy += fabsf(speech_synth[i]);
508	postfilter_energy += fabsf(in[i]);
509	}
510	gain_scale_factor = postfilter_energy == 0.0 ? 0.0 :
511	(1.0 - alpha) * speech_energy / postfilter_energy;
512
513	for (i = 0; i < size; i++) {
514	mem = alpha * mem + gain_scale_factor;
515	out[i] = in[i] * mem;
516	}
517
518	*gain_mem = mem;
519	}
520
521	/**
522	* Kalman smoothing function.
523	*
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
528	* filter.
529	*
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
535	*
536	* @returns -1 if no smoothening took place, e.g. because no optimal
537	* fit could be found, or 0 on success.
538	*/
539	static int kalman_smoothen(WMAVoiceContext *s, int pitch,
540	const float *in, float *out, int size)
541	{
542	int n;
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;
547
548	/* find best fitting point in history */
549	do {
550	dot = avpriv_scalarproduct_float_c(in, ptr, size);
551	if (dot > optimal_gain) {
552	optimal_gain = dot;
553	best_hist_ptr = ptr;
554	}
555	} while (--ptr >= end);
556
557	if (optimal_gain <= 0)
558	return -1;
559	dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
560	if (dot <= 0) // would be 1.0
561	return -1;
562
563	if (optimal_gain <= dot) {
564	dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000
565	} else
566	dot = 0.625;
567
568	/* actual smoothing */
569	for (n = 0; n < size; n++)
570	out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]);
571
572	return 0;
573	}
574
575	/**
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
580	*
581	* @param lpcs LPC coefficients
582	* @param n_lpcs Size of LPC buffer
583	* @returns the tilt factor
584	*/
585	static float tilt_factor(const float *lpcs, int n_lpcs)
586	{
587	float rh0, rh1;
588
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);
591
592	return rh1 / rh0;
593	}
594
595	/**
596	* Derive denoise filter coefficients (in real domain) from the LPCs.
597	*/
598	static void calc_input_response(WMAVoiceContext *s, float *lpcs,
599	int fcb_type, float *coeffs, int remainder)
600	{
601	float last_coeff, min = 15.0, max = -15.0;
602	float irange, angle_mul, gain_mul, range, sq;
603	int n, idx;
604
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); \
610	} while (0)
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]);
616	#undef log_range
617	range = max - min;
618	lpcs[64] = last_coeff;
619
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) :
627	(5.0 / 14.7));
628	angle_mul = gain_mul * (8.0 * M_LN10 / M_PI);
629	for (n = 0; n <= 64; n++) {
630	float pwr;
631
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;
635
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);
641	} else
642	coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)];
643	}
644
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);
651
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];
657	for (n = 63;; n--) {
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];
661
662	if (!--n) break;
663
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];
667	}
668	coeffs[1] = last_coeff;
669
670	/* move into real domain */
671	s->irdft.rdft_calc(&s->irdft, coeffs);
672
673	/* tilt correction and normalize scale */
674	memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder));
675	if (s->denoise_tilt_corr) {
676	float tilt_mem = 0;
677
678	coeffs[remainder - 1] = 0;
679	ff_tilt_compensation(&tilt_mem,
680	-1.8 * tilt_factor(coeffs, remainder - 1),
681	coeffs, remainder);
682	}
683	sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
684	remainder));
685	for (n = 0; n < remainder; n++)
686	coeffs[n] *= sq;
687	}
688
689	/**
690	* This function applies a Wiener filter on the (noisy) speech signal as
691	* a means to denoise it.
692	*
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
698	* of 4-8kHz);
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).
707	*
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
714	*/
715	static void wiener_denoise(WMAVoiceContext *s, int fcb_type,
716	float *synth_pf, int size,
717	const float *lpcs)
718	{
719	int remainder, lim, n;
720
721	if (fcb_type != FCB_TYPE_SILENCE) {
722	float *tilted_lpcs = s->tilted_lpcs_pf,
723	*coeffs = s->denoise_coeffs_pf, tilt_mem = 0;
724
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);
731
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);
738
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];
750	}
751	s->irdft.rdft_calc(&s->irdft, synth_pf);
752	}
753
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);
762	}
763
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;
773	}
774	}
775	}
776
777	/**
778	* Averaging projection filter, the postfilter used in WMAVoice.
779	*
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
786	* - DC filter
787	*
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
796	*/
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)
801	{
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;
805
806	av_assert0(size <= MAX_FRAMESIZE / 2);
807
808	/* generate excitation from input signal */
809	ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps);
810
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;
814
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);
820
821	wiener_denoise(s, fcb_type, synth_pf, size, lpcs);
822
823	adaptive_gain_control(samples, synth_pf, synth, size, 0.99,
824	&s->postfilter_agc);
825
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);
834	}
835	}
836	/**
837	* @}
838	*/
839
840	/**
841	* Dequantize LSPs
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
850	*/
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,
855	const double *mul_q,
856	const double *base_q)
857	{
858	int n, m;
859
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];
864
865	for (m = 0; m < num; m++)
866	lsps[m] += base + mul * t_off[m];
867
868	table += sizes[n] * num;
869	}
870	}
871
872	/**
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.
877	* @{
878	*/
879	/**
880	* Parse 10 independently-coded LSPs.
881	*/
882	static void dequant_lsp10i(GetBitContext *gb, double *lsps)
883	{
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
888	};
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
892	};
893	uint16_t v[4];
894
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);
899
900	dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i,
901	mul_lsf, base_lsf);
902	}
903
904	/**
905	* Parse 10 independently-coded LSPs, and then derive the tables to
906	* generate LSPs for the other frames from them (residual coding).
907	*/
908	static void dequant_lsp10r(GetBitContext *gb,
909	double *i_lsps, const double *old,
910	double *a1, double *a2, int q_mode)
911	{
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
915	};
916	static const double base_lsf[3] = {
917	M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2
918	};
919	const float (*ipol_tab)[2][10] = q_mode ?
920	wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a;
921	uint16_t interpol, v[3];
922	int n;
923
924	dequant_lsp10i(gb, i_lsps);
925
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);
930
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];
935	}
936
937	dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r,
938	mul_lsf, base_lsf);
939	}
940
941	/**
942	* Parse 16 independently-coded LSPs.
943	*/
944	static void dequant_lsp16i(GetBitContext *gb, double *lsps)
945	{
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,
950	3.1899104283e-3
951	};
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,
955	M_PI * -1.29816e-1
956	};
957	uint16_t v[5];
958
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);
964
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]);
971	}
972
973	/**
974	* Parse 16 independently-coded LSPs, and then derive the tables to
975	* generate LSPs for the other frames from them (residual coding).
976	*/
977	static void dequant_lsp16r(GetBitContext *gb,
978	double *i_lsps, const double *old,
979	double *a1, double *a2, int q_mode)
980	{
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
984	};
985	static const double base_lsf[3] = {
986	M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2
987	};
988	const float (*ipol_tab)[2][16] = q_mode ?
989	wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a;
990	uint16_t interpol, v[3];
991	int n;
992
993	dequant_lsp16i(gb, i_lsps);
994
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);
999
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];
1004	}
1005
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]);
1012	}
1013
1014	/**
1015	* @}
1016	* @name Pitch-adaptive window coding functions
1017	* The next few functions are for pitch-adaptive window coding.
1018	* @{
1019	*/
1020	/**
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
1026	*/
1027	static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb,
1028	const int *pitch)
1029	{
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
1039	};
1040	int bits, offset;
1041
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);
1047	}
1048
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;
1058
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];
1068	}
1069	}
1070
1071	/**
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
1078	*/
1079	static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb,
1080	int block_idx, AMRFixed *fcb)
1081	{
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;
1093
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;
1098
1099	/* find range per pulse */
1100	if (s->aw_n_pulses[0] > 0) {
1101	if (block_idx == 0) {
1102	range = 32;
1103	} else /* block_idx = 1 */ {
1104	range = 8;
1105	if (s->aw_n_pulses[block_idx] > 0)
1106	pulse_off = s->aw_next_pulse_off_cache;
1107	}
1108	} else
1109	range = 16;
1110	pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0;
1111
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);
1128	} else
1129	*use_mask_ptr &= 0xFFFF >> excl_range;
1130	}
1131
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;
1142	else return -1;
1143	idx -= av_log2_16bit(use_mask[idx >> 4]);
1144	}
1145	if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) {
1146	use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15));
1147	n++;
1148	start_off = idx;
1149	}
1150	}
1151
1152	fcb->x[fcb->n] = start_off;
1153	fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0;
1154	fcb->n++;
1155
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;
1159	return 0;
1160	}
1161
1162	/**
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
1168	*/
1169	static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb,
1170	int block_idx, AMRFixed *fcb)
1171	{
1172	int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx));
1173	float v;
1174
1175	if (s->aw_n_pulses[block_idx] > 0) {
1176	int n, v_mask, i_mask, sh, n_pulses;
1177
1178	if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each
1179	n_pulses = 3;
1180	v_mask = 8;
1181	i_mask = 7;
1182	sh = 4;
1183	} else { // 4 pulses, 1:sign + 2:index each
1184	n_pulses = 4;
1185	v_mask = 4;
1186	i_mask = 3;
1187	sh = 3;
1188	}
1189
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)
1197	fcb->n++;
1198	}
1199	} else {
1200	int num2 = (val & 0x1FF) >> 1, delta, idx;
1201
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;
1207
1208	fcb->no_repeat_mask |= 3 << fcb->n;
1209	fcb->x[fcb->n] = idx - delta;
1210	fcb->y[fcb->n] = v;
1211	fcb->x[fcb->n + 1] = idx;
1212	fcb->y[fcb->n + 1] = (val & 1) ? -v : v;
1213	fcb->n += 2;
1214	}
1215	}
1216
1217	/**
1218	* @}
1219	*
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).
1223	*
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.
1229	*/
1230	static int pRNG(int frame_cntr, int block_num, int block_size)
1231	{
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
1252	};
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);
1260	}
1261
1262	/**
1263	* Parse hardcoded signal for a single block.
1264	* @note see #synth_block().
1265	*/
1266	static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb,
1267	int block_idx, int size,
1268	const struct frame_type_desc *frame_desc,
1269	float *excitation)
1270	{
1271	float gain;
1272	int n, r_idx;
1273
1274	av_assert0(size <= MAX_FRAMESIZE);
1275
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)];
1283	}
1284
1285	/* Clear gain prediction parameters */
1286	memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err));
1287
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;
1291	}
1292
1293	/**
1294	* Parse FCB/ACB signal for a single block.
1295	* @note see #synth_block().
1296	*/
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,
1301	float *excitation)
1302	{
1303	static const float gain_coeff[6] = {
1304	0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458
1305	};
1306	float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain;
1307	int n, idx, gain_weight;
1308	AMRFixed fcb;
1309
1310	av_assert0(size <= MAX_FRAMESIZE / 2);
1311	memset(pulses, 0, sizeof(*pulses) * size);
1312
1313	fcb.pitch_lag = block_pitch_sh2 >> 2;
1314	fcb.pitch_fac = 1.0;
1315	fcb.no_repeat_mask = 0;
1316	fcb.n = 0;
1317
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);
1327
1328	for (n = 0; n < size; n++)
1329	excitation[n] =
1330	wmavoice_std_codebook[r_idx + n] * s->silence_gain;
1331	skip_bits(gb, 7 + 1);
1332	return;
1333	}
1334	} else /* FCB_TYPE_EXC_PULSES */ {
1335	int offset_nbits = 5 - frame_desc->log_n_blocks;
1336
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++) {
1341	float sign;
1342	int pos1, pos2;
1343
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;
1352	}
1353	}
1354	}
1355	ff_set_fixed_vector(pulses, &fcb, 1.0, size);
1356
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,
1361	gain_coeff, 6) -
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) */);
1367
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;
1373
1374	/* Calculation of adaptive codebook */
1375	if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) {
1376	int len;
1377	for (n = 0; n < size; n += len) {
1378	int next_idx_sh16;
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;
1388	} else
1389	next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF;
1390	len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8,
1391	1, size - n);
1392	} else
1393	len = size;
1394
1395	ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch],
1396	wmavoice_ipol1_coeffs, 17,
1397	idx, 9, len);
1398	}
1399	} else /* ACB_TYPE_HAMMING */ {
1400	int block_pitch = block_pitch_sh2 >> 2;
1401	idx = block_pitch_sh2 & 3;
1402	if (idx) {
1403	ff_acelp_interpolatef(excitation, &excitation[-block_pitch],
1404	wmavoice_ipol2_coeffs, 4,
1405	idx, 8, size);
1406	} else
1407	av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch,
1408	sizeof(float) * size);
1409	}
1410
1411	/* Interpolate ACB/FCB and use as excitation signal */
1412	ff_weighted_vector_sumf(excitation, excitation, pulses,
1413	acb_gain, fcb_gain, size);
1414	}
1415
1416	/**
1417	* Parse data in a single block.
1418	*
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.
1430	*/
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)
1437	{
1438	double i_lsps[MAX_LSPS];
1439	float lpcs[MAX_LSPS];
1440	float fac;
1441	int n;
1442
1443	if (frame_desc->acb_type == ACB_TYPE_NONE)
1444	synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation);
1445	else
1446	synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2,
1447	frame_desc, excitation);
1448
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);
1454
1455	/* Speech synthesis */
1456	ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps);
1457	}
1458
1459	/**
1460	* Synthesize output samples for a single frame.
1461	*
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
1466	* samples
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.
1472	*/
1473	static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx,
1474	float *samples,
1475	const double *lsps, const double *prev_lsps,
1476	float *excitation, float *synth)
1477	{
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);
1481
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;
1484
1485	if (bd_idx < 0) {
1486	av_log(ctx, AV_LOG_ERROR,
1487	"Invalid frame type VLC code, skipping\n");
1488	return AVERROR_INVALIDDATA;
1489	}
1490
1491	block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks;
1492
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;
1507
1508	/* pitch per block */
1509	for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1510	int fac = n * 2 + 1;
1511
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;
1515	}
1516
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;
1520	}
1521
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)];
1526	break;
1527	case FCB_TYPE_AW_PULSES:
1528	aw_parse_coords(s, gb, pitch);
1529	break;
1530	}
1531
1532	for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) {
1533	int bl_pitch_sh2;
1534
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. */
1543	int block_pitch,
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;
1547
1548	if (n == 0) {
1549	block_pitch = get_bits(gb, s->block_pitch_nbits);
1550	} else
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);
1558
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;
1562	} else {
1563	block_pitch -= t1;
1564	if (block_pitch < t2) {
1565	bl_pitch_sh2 =
1566	(s->block_conv_table[1] << 2) + (block_pitch << 1);
1567	} else {
1568	block_pitch -= t2;
1569	if (block_pitch < t3) {
1570	bl_pitch_sh2 =
1571	(s->block_conv_table[2] + block_pitch) << 2;
1572	} else
1573	bl_pitch_sh2 = s->block_conv_table[3] << 2;
1574	}
1575	}
1576	pitch[n] = bl_pitch_sh2 >> 2;
1577	break;
1578	}
1579
1580	case ACB_TYPE_ASYMMETRIC: {
1581	bl_pitch_sh2 = pitch[n] << 2;
1582	break;
1583	}
1584
1585	default: // ACB_TYPE_NONE has no pitch
1586	bl_pitch_sh2 = 0;
1587	break;
1588	}
1589
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]);
1594	}
1595
1596	/* Averaging projection filter, if applicable. Else, just copy samples
1597	* from synthesis buffer */
1598	if (s->do_apf) {
1599	double i_lsps[MAX_LSPS];
1600	float lpcs[MAX_LSPS];
1601
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]);
1608
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]);
1615	} else
1616	memcpy(samples, synth, 160 * sizeof(synth[0]));
1617
1618	/* Cache values for next frame */
1619	s->frame_cntr++;
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) {
1623	case ACB_TYPE_NONE:
1624	s->last_pitch_val = 0;
1625	break;
1626	case ACB_TYPE_ASYMMETRIC:
1627	s->last_pitch_val = cur_pitch_val;
1628	break;
1629	case ACB_TYPE_HAMMING:
1630	s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1];
1631	break;
1632	}
1633
1634	return 0;
1635	}
1636
1637	/**
1638	* Ensure minimum value for first item, maximum value for last value,
1639	* proper spacing between each value and proper ordering.
1640	*
1641	* @param lsps array of LSPs
1642	* @param num size of LSP array
1643	*
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.
1648	*/
1649	static void stabilize_lsps(double *lsps, int num)
1650	{
1651	int n, m, l;
1652
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);
1660
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];
1670	}
1671	lsps[l + 1] = tmp;
1672	}
1673	break;
1674	}
1675	}
1676	}
1677
1678	/**
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
1681	* in s->gb.
1682	*
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.
1690	*
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
1694	*/
1695	static int synth_superframe(AVCodecContext *ctx, AVFrame *frame,
1696	int *got_frame_ptr)
1697	{
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];
1706	float *samples;
1707
1708	memcpy(synth, s->synth_history,
1709	s->lsps * sizeof(*synth));
1710	memcpy(excitation, s->excitation_history,
1711	s->history_nsamples * sizeof(*excitation));
1712
1713	if (s->sframe_cache_size > 0) {
1714	gb = &s_gb;
1715	init_get_bits(gb, s->sframe_cache, s->sframe_cache_size);
1716	s->sframe_cache_size = 0;
1717	}
1718
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
1722	* the wild yet. */
1723	if (!get_bits1(gb)) {
1724	avpriv_request_sample(ctx, "WMAPro-in-WMAVoice");
1725	return AVERROR_PATCHWELCOME;
1726	}
1727
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;
1735	}
1736	}
1737
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];
1741
1742	for (n = 0; n < s->lsps; n++)
1743	prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n];
1744
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);
1749
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];
1754	}
1755	for (n = 0; n < 3; n++)
1756	stabilize_lsps(lsps[n], s->lsps);
1757	}
1758
1759	/* get output buffer */
1760	frame->nb_samples = MAX_SFRAMESIZE;
1761	if ((res = ff_get_buffer(ctx, frame, 0)) < 0)
1762	return res;
1763	frame->nb_samples = n_samples;
1764	samples = (float *)frame->data[0];
1765
1766	/* Parse frames, optionally preceded by per-frame (independent) LSPs. */
1767	for (n = 0; n < 3; n++) {
1768	if (!s->has_residual_lsps) {
1769	int m;
1770
1771	if (s->lsps == 10) {
1772	dequant_lsp10i(gb, lsps[n]);
1773	} else /* s->lsps == 16 */
1774	dequant_lsp16i(gb, lsps[n]);
1775
1776	for (m = 0; m < s->lsps; m++)
1777	lsps[n][m] += mean_lsf[m];
1778	stabilize_lsps(lsps[n], s->lsps);
1779	}
1780
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]))) {
1786	*got_frame_ptr = 0;
1787	return res;
1788	}
1789	}
1790
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));
1797	}
1798
1799	if (get_bits_left(gb) < 0) {
1800	wmavoice_flush(ctx);
1801	return AVERROR_INVALIDDATA;
1802	}
1803
1804	*got_frame_ptr = 1;
1805
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));
1813	if (s->do_apf)
1814	memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE],
1815	s->history_nsamples * sizeof(*s->zero_exc_pf));
1816
1817	return 0;
1818	}
1819
1820	/**
1821	* Parse the packet header at the start of each packet (input data to this
1822	* decoder).
1823	*
1824	* @param s WMA Voice decoding context private data
1825	* @return <0 on error, nb_superframes on success.
1826	*/
1827	static int parse_packet_header(WMAVoiceContext *s)
1828	{
1829	GetBitContext *gb = &s->gb;
1830	unsigned int res, n_superframes = 0;
1831
1832	skip_bits(gb, 4); // packet sequence number
1833	s->has_residual_lsps = get_bits1(gb);
1834	do {
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);
1840
1841	return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA;
1842	}
1843
1844	/**
1845	* Copy (unaligned) bits from gb/data/size to pb.
1846	*
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
1853	* source data
1854	* @param nbits the amount of bits to copy from source to target
1855	*
1856	* @note after calling this function, the current position in the input bit
1857	* I/O context is undefined.
1858	*/
1859	static void copy_bits(PutBitContext *pb,
1860	const uint8_t *data, int size,
1861	GetBitContext *gb, int nbits)
1862	{
1863	int rmn_bytes, rmn_bits;
1864
1865	rmn_bits = rmn_bytes = get_bits_left(gb);
1866	if (rmn_bits < nbits)
1867	return;
1868	if (nbits > pb->size_in_bits - put_bits_count(pb))
1869	return;
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));
1875	}
1876
1877	/**
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.
1885	*
1886	* For more information about frames, see #synth_superframe().
1887	*/
1888	static int wmavoice_decode_packet(AVCodecContext *ctx, void *data,
1889	int *got_frame_ptr, AVPacket *avpkt)
1890	{
1891	WMAVoiceContext *s = ctx->priv_data;
1892	GetBitContext *gb = &s->gb;
1893	int size, res, pos;
1894
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);
1902
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
1905	* previously. */
1906	if (!(size % ctx->block_align)) { // new packet header
1907	if (!size) {
1908	s->spillover_nbits = 0;
1909	s->nb_superframes = 0;
1910	} else {
1911	if ((res = parse_packet_header(s)) < 0)
1912	return res;
1913	s->nb_superframes = res;
1914	}
1915
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;
1923	}
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 &&
1928	*got_frame_ptr) {
1929	cnt += s->spillover_nbits;
1930	s->skip_bits_next = cnt & 7;
1931	res = cnt >> 3;
1932	return res;
1933	} else
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
1938	}
1939	} else if (s->skip_bits_next)
1940	skip_bits(gb, s->skip_bits_next);
1941
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) {
1947	*got_frame_ptr = 0;
1948	return size;
1949	} else if (s->nb_superframes > 0) {
1950	if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) {
1951	return res;
1952	} else if (*got_frame_ptr) {
1953	int cnt = get_bits_count(gb);
1954	s->skip_bits_next = cnt & 7;
1955	res = cnt >> 3;
1956	return res;
1957	}
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
1964	}
1965
1966	return size;
1967	}
1968
1969	static av_cold int wmavoice_decode_end(AVCodecContext *ctx)
1970	{
1971	WMAVoiceContext *s = ctx->priv_data;
1972
1973	if (s->do_apf) {
1974	ff_rdft_end(&s->rdft);
1975	ff_rdft_end(&s->irdft);
1976	ff_dct_end(&s->dct);
1977	ff_dct_end(&s->dst);
1978	}
1979
1980	return 0;
1981	}
1982
1983	AVCodec ff_wmavoice_decoder = {
1984	.name = "wmavoice",
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,
1995	};
1996