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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 | uint16_t frame_size; ///< the amount of bits that make up the block | |
108 | ///< data (per frame) | |
109 | } frame_descs[17] = { | |
110 | { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, | |
111 | { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, | |
112 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, | |
113 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, | |
114 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, | |
115 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, | |
116 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, | |
117 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, | |
118 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, | |
119 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, | |
120 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, | |
121 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, | |
122 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, | |
123 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, | |
124 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, | |
125 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, | |
126 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } | |
127 | }; | |
128 | ||
129 | /** | |
130 | * WMA Voice decoding context. | |
131 | */ | |
132 | typedef struct { | |
133 | /** | |
134 | * @name Global values specified in the stream header / extradata or used all over. | |
135 | * @{ | |
136 | */ | |
137 | GetBitContext gb; ///< packet bitreader. During decoder init, | |
138 | ///< it contains the extradata from the | |
139 | ///< demuxer. During decoding, it contains | |
140 | ///< packet data. | |
141 | int8_t vbm_tree[25]; ///< converts VLC codes to frame type | |
142 | ||
143 | int spillover_bitsize; ///< number of bits used to specify | |
144 | ///< #spillover_nbits in the packet header | |
145 | ///< = ceil(log2(ctx->block_align << 3)) | |
146 | int history_nsamples; ///< number of samples in history for signal | |
147 | ///< prediction (through ACB) | |
148 | ||
149 | /* postfilter specific values */ | |
150 | int do_apf; ///< whether to apply the averaged | |
151 | ///< projection filter (APF) | |
152 | int denoise_strength; ///< strength of denoising in Wiener filter | |
153 | ///< [0-11] | |
154 | int denoise_tilt_corr; ///< Whether to apply tilt correction to the | |
155 | ///< Wiener filter coefficients (postfilter) | |
156 | int dc_level; ///< Predicted amount of DC noise, based | |
157 | ///< on which a DC removal filter is used | |
158 | ||
159 | int lsps; ///< number of LSPs per frame [10 or 16] | |
160 | int lsp_q_mode; ///< defines quantizer defaults [0, 1] | |
161 | int lsp_def_mode; ///< defines different sets of LSP defaults | |
162 | ///< [0, 1] | |
163 | int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded | |
164 | ///< per-frame (independent coding) | |
165 | int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded | |
166 | ///< per superframe (residual coding) | |
167 | ||
168 | int min_pitch_val; ///< base value for pitch parsing code | |
169 | int max_pitch_val; ///< max value + 1 for pitch parsing | |
170 | int pitch_nbits; ///< number of bits used to specify the | |
171 | ///< pitch value in the frame header | |
172 | int block_pitch_nbits; ///< number of bits used to specify the | |
173 | ///< first block's pitch value | |
174 | int block_pitch_range; ///< range of the block pitch | |
175 | int block_delta_pitch_nbits; ///< number of bits used to specify the | |
176 | ///< delta pitch between this and the last | |
177 | ///< block's pitch value, used in all but | |
178 | ///< first block | |
179 | int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is | |
180 | ///< from -this to +this-1) | |
181 | uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale | |
182 | ///< conversion | |
183 | ||
184 | /** | |
185 | * @} | |
186 | * | |
187 | * @name Packet values specified in the packet header or related to a packet. | |
188 | * | |
189 | * A packet is considered to be a single unit of data provided to this | |
190 | * decoder by the demuxer. | |
191 | * @{ | |
192 | */ | |
193 | int spillover_nbits; ///< number of bits of the previous packet's | |
194 | ///< last superframe preceding this | |
195 | ///< packet's first full superframe (useful | |
196 | ///< for re-synchronization also) | |
197 | int has_residual_lsps; ///< if set, superframes contain one set of | |
198 | ///< LSPs that cover all frames, encoded as | |
199 | ///< independent and residual LSPs; if not | |
200 | ///< set, each frame contains its own, fully | |
201 | ///< independent, LSPs | |
202 | int skip_bits_next; ///< number of bits to skip at the next call | |
203 | ///< to #wmavoice_decode_packet() (since | |
204 | ///< they're part of the previous superframe) | |
205 | ||
206 | uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE]; | |
207 | ///< cache for superframe data split over | |
208 | ///< multiple packets | |
209 | int sframe_cache_size; ///< set to >0 if we have data from an | |
210 | ///< (incomplete) superframe from a previous | |
211 | ///< packet that spilled over in the current | |
212 | ///< packet; specifies the amount of bits in | |
213 | ///< #sframe_cache | |
214 | PutBitContext pb; ///< bitstream writer for #sframe_cache | |
215 | ||
216 | /** | |
217 | * @} | |
218 | * | |
219 | * @name Frame and superframe values | |
220 | * Superframe and frame data - these can change from frame to frame, | |
221 | * although some of them do in that case serve as a cache / history for | |
222 | * the next frame or superframe. | |
223 | * @{ | |
224 | */ | |
225 | double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous | |
226 | ///< superframe | |
227 | int last_pitch_val; ///< pitch value of the previous frame | |
228 | int last_acb_type; ///< frame type [0-2] of the previous frame | |
229 | int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) | |
230 | ///< << 16) / #MAX_FRAMESIZE | |
231 | float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE | |
232 | ||
233 | int aw_idx_is_ext; ///< whether the AW index was encoded in | |
234 | ///< 8 bits (instead of 6) | |
235 | int aw_pulse_range; ///< the range over which #aw_pulse_set1() | |
236 | ///< can apply the pulse, relative to the | |
237 | ///< value in aw_first_pulse_off. The exact | |
238 | ///< position of the first AW-pulse is within | |
239 | ///< [pulse_off, pulse_off + this], and | |
240 | ///< depends on bitstream values; [16 or 24] | |
241 | int aw_n_pulses[2]; ///< number of AW-pulses in each block; note | |
242 | ///< that this number can be negative (in | |
243 | ///< which case it basically means "zero") | |
244 | int aw_first_pulse_off[2]; ///< index of first sample to which to | |
245 | ///< apply AW-pulses, or -0xff if unset | |
246 | int aw_next_pulse_off_cache; ///< the position (relative to start of the | |
247 | ///< second block) at which pulses should | |
248 | ///< start to be positioned, serves as a | |
249 | ///< cache for pitch-adaptive window pulses | |
250 | ///< between blocks | |
251 | ||
252 | int frame_cntr; ///< current frame index [0 - 0xFFFE]; is | |
253 | ///< only used for comfort noise in #pRNG() | |
254 | float gain_pred_err[6]; ///< cache for gain prediction | |
255 | float excitation_history[MAX_SIGNAL_HISTORY]; | |
256 | ///< cache of the signal of previous | |
257 | ///< superframes, used as a history for | |
258 | ///< signal generation | |
259 | float synth_history[MAX_LSPS]; ///< see #excitation_history | |
260 | /** | |
261 | * @} | |
262 | * | |
263 | * @name Postfilter values | |
264 | * | |
265 | * Variables used for postfilter implementation, mostly history for | |
266 | * smoothing and so on, and context variables for FFT/iFFT. | |
267 | * @{ | |
268 | */ | |
269 | RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the | |
270 | ///< postfilter (for denoise filter) | |
271 | DCTContext dct, dst; ///< contexts for phase shift (in Hilbert | |
272 | ///< transform, part of postfilter) | |
273 | float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] | |
274 | ///< range | |
275 | float postfilter_agc; ///< gain control memory, used in | |
276 | ///< #adaptive_gain_control() | |
277 | float dcf_mem[2]; ///< DC filter history | |
278 | float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; | |
279 | ///< zero filter output (i.e. excitation) | |
280 | ///< by postfilter | |
281 | float denoise_filter_cache[MAX_FRAMESIZE]; | |
282 | int denoise_filter_cache_size; ///< samples in #denoise_filter_cache | |
283 | DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80]; | |
284 | ///< aligned buffer for LPC tilting | |
285 | DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; | |
286 | ///< aligned buffer for denoise coefficients | |
287 | DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; | |
288 | ///< aligned buffer for postfilter speech | |
289 | ///< synthesis | |
290 | /** | |
291 | * @} | |
292 | */ | |
293 | } WMAVoiceContext; | |
294 | ||
295 | /** | |
296 | * Set up the variable bit mode (VBM) tree from container extradata. | |
297 | * @param gb bit I/O context. | |
298 | * The bit context (s->gb) should be loaded with byte 23-46 of the | |
299 | * container extradata (i.e. the ones containing the VBM tree). | |
300 | * @param vbm_tree pointer to array to which the decoded VBM tree will be | |
301 | * written. | |
302 | * @return 0 on success, <0 on error. | |
303 | */ | |
304 | static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) | |
305 | { | |
306 | int cntr[8] = { 0 }, n, res; | |
307 | ||
308 | memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); | |
309 | for (n = 0; n < 17; n++) { | |
310 | res = get_bits(gb, 3); | |
311 | if (cntr[res] > 3) // should be >= 3 + (res == 7)) | |
312 | return -1; | |
313 | vbm_tree[res * 3 + cntr[res]++] = n; | |
314 | } | |
315 | return 0; | |
316 | } | |
317 | ||
318 | static av_cold void wmavoice_init_static_data(AVCodec *codec) | |
319 | { | |
320 | static const uint8_t bits[] = { | |
321 | 2, 2, 2, 4, 4, 4, | |
322 | 6, 6, 6, 8, 8, 8, | |
323 | 10, 10, 10, 12, 12, 12, | |
324 | 14, 14, 14, 14 | |
325 | }; | |
326 | static const uint16_t codes[] = { | |
327 | 0x0000, 0x0001, 0x0002, // 00/01/10 | |
328 | 0x000c, 0x000d, 0x000e, // 11+00/01/10 | |
329 | 0x003c, 0x003d, 0x003e, // 1111+00/01/10 | |
330 | 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 | |
331 | 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 | |
332 | 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 | |
333 | 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx | |
334 | }; | |
335 | ||
336 | INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), | |
337 | bits, 1, 1, codes, 2, 2, 132); | |
338 | } | |
339 | ||
340 | /** | |
341 | * Set up decoder with parameters from demuxer (extradata etc.). | |
342 | */ | |
343 | static av_cold int wmavoice_decode_init(AVCodecContext *ctx) | |
344 | { | |
345 | int n, flags, pitch_range, lsp16_flag; | |
346 | WMAVoiceContext *s = ctx->priv_data; | |
347 | ||
348 | /** | |
349 | * Extradata layout: | |
350 | * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), | |
351 | * - byte 19-22: flags field (annoyingly in LE; see below for known | |
352 | * values), | |
353 | * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, | |
354 | * rest is 0). | |
355 | */ | |
356 | if (ctx->extradata_size != 46) { | |
357 | av_log(ctx, AV_LOG_ERROR, | |
358 | "Invalid extradata size %d (should be 46)\n", | |
359 | ctx->extradata_size); | |
360 | return AVERROR_INVALIDDATA; | |
361 | } | |
362 | flags = AV_RL32(ctx->extradata + 18); | |
363 | s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); | |
364 | s->do_apf = flags & 0x1; | |
365 | if (s->do_apf) { | |
366 | ff_rdft_init(&s->rdft, 7, DFT_R2C); | |
367 | ff_rdft_init(&s->irdft, 7, IDFT_C2R); | |
368 | ff_dct_init(&s->dct, 6, DCT_I); | |
369 | ff_dct_init(&s->dst, 6, DST_I); | |
370 | ||
371 | ff_sine_window_init(s->cos, 256); | |
372 | memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); | |
373 | for (n = 0; n < 255; n++) { | |
374 | s->sin[n] = -s->sin[510 - n]; | |
375 | s->cos[510 - n] = s->cos[n]; | |
376 | } | |
377 | } | |
378 | s->denoise_strength = (flags >> 2) & 0xF; | |
379 | if (s->denoise_strength >= 12) { | |
380 | av_log(ctx, AV_LOG_ERROR, | |
381 | "Invalid denoise filter strength %d (max=11)\n", | |
382 | s->denoise_strength); | |
383 | return AVERROR_INVALIDDATA; | |
384 | } | |
385 | s->denoise_tilt_corr = !!(flags & 0x40); | |
386 | s->dc_level = (flags >> 7) & 0xF; | |
387 | s->lsp_q_mode = !!(flags & 0x2000); | |
388 | s->lsp_def_mode = !!(flags & 0x4000); | |
389 | lsp16_flag = flags & 0x1000; | |
390 | if (lsp16_flag) { | |
391 | s->lsps = 16; | |
392 | s->frame_lsp_bitsize = 34; | |
393 | s->sframe_lsp_bitsize = 60; | |
394 | } else { | |
395 | s->lsps = 10; | |
396 | s->frame_lsp_bitsize = 24; | |
397 | s->sframe_lsp_bitsize = 48; | |
398 | } | |
399 | for (n = 0; n < s->lsps; n++) | |
400 | s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
401 | ||
402 | init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); | |
403 | if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { | |
404 | av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); | |
405 | return AVERROR_INVALIDDATA; | |
406 | } | |
407 | ||
408 | s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; | |
409 | s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; | |
410 | pitch_range = s->max_pitch_val - s->min_pitch_val; | |
411 | if (pitch_range <= 0) { | |
412 | av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n"); | |
413 | return AVERROR_INVALIDDATA; | |
414 | } | |
415 | s->pitch_nbits = av_ceil_log2(pitch_range); | |
416 | s->last_pitch_val = 40; | |
417 | s->last_acb_type = ACB_TYPE_NONE; | |
418 | s->history_nsamples = s->max_pitch_val + 8; | |
419 | ||
420 | if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { | |
421 | int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, | |
422 | max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; | |
423 | ||
424 | av_log(ctx, AV_LOG_ERROR, | |
425 | "Unsupported samplerate %d (min=%d, max=%d)\n", | |
426 | ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz | |
427 | ||
428 | return AVERROR(ENOSYS); | |
429 | } | |
430 | ||
431 | s->block_conv_table[0] = s->min_pitch_val; | |
432 | s->block_conv_table[1] = (pitch_range * 25) >> 6; | |
433 | s->block_conv_table[2] = (pitch_range * 44) >> 6; | |
434 | s->block_conv_table[3] = s->max_pitch_val - 1; | |
435 | s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; | |
436 | if (s->block_delta_pitch_hrange <= 0) { | |
437 | av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n"); | |
438 | return AVERROR_INVALIDDATA; | |
439 | } | |
440 | s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); | |
441 | s->block_pitch_range = s->block_conv_table[2] + | |
442 | s->block_conv_table[3] + 1 + | |
443 | 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); | |
444 | s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); | |
445 | ||
446 | ctx->channels = 1; | |
447 | ctx->channel_layout = AV_CH_LAYOUT_MONO; | |
448 | ctx->sample_fmt = AV_SAMPLE_FMT_FLT; | |
449 | ||
450 | return 0; | |
451 | } | |
452 | ||
453 | /** | |
454 | * @name Postfilter functions | |
455 | * Postfilter functions (gain control, wiener denoise filter, DC filter, | |
456 | * kalman smoothening, plus surrounding code to wrap it) | |
457 | * @{ | |
458 | */ | |
459 | /** | |
460 | * Adaptive gain control (as used in postfilter). | |
461 | * | |
462 | * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except | |
463 | * that the energy here is calculated using sum(abs(...)), whereas the | |
464 | * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). | |
465 | * | |
466 | * @param out output buffer for filtered samples | |
467 | * @param in input buffer containing the samples as they are after the | |
468 | * postfilter steps so far | |
469 | * @param speech_synth input buffer containing speech synth before postfilter | |
470 | * @param size input buffer size | |
471 | * @param alpha exponential filter factor | |
472 | * @param gain_mem pointer to filter memory (single float) | |
473 | */ | |
474 | static void adaptive_gain_control(float *out, const float *in, | |
475 | const float *speech_synth, | |
476 | int size, float alpha, float *gain_mem) | |
477 | { | |
478 | int i; | |
479 | float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; | |
480 | float mem = *gain_mem; | |
481 | ||
482 | for (i = 0; i < size; i++) { | |
483 | speech_energy += fabsf(speech_synth[i]); | |
484 | postfilter_energy += fabsf(in[i]); | |
485 | } | |
486 | gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; | |
487 | ||
488 | for (i = 0; i < size; i++) { | |
489 | mem = alpha * mem + gain_scale_factor; | |
490 | out[i] = in[i] * mem; | |
491 | } | |
492 | ||
493 | *gain_mem = mem; | |
494 | } | |
495 | ||
496 | /** | |
497 | * Kalman smoothing function. | |
498 | * | |
499 | * This function looks back pitch +/- 3 samples back into history to find | |
500 | * the best fitting curve (that one giving the optimal gain of the two | |
501 | * signals, i.e. the highest dot product between the two), and then | |
502 | * uses that signal history to smoothen the output of the speech synthesis | |
503 | * filter. | |
504 | * | |
505 | * @param s WMA Voice decoding context | |
506 | * @param pitch pitch of the speech signal | |
507 | * @param in input speech signal | |
508 | * @param out output pointer for smoothened signal | |
509 | * @param size input/output buffer size | |
510 | * | |
511 | * @returns -1 if no smoothening took place, e.g. because no optimal | |
512 | * fit could be found, or 0 on success. | |
513 | */ | |
514 | static int kalman_smoothen(WMAVoiceContext *s, int pitch, | |
515 | const float *in, float *out, int size) | |
516 | { | |
517 | int n; | |
518 | float optimal_gain = 0, dot; | |
519 | const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], | |
520 | *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], | |
521 | *best_hist_ptr = NULL; | |
522 | ||
523 | /* find best fitting point in history */ | |
524 | do { | |
525 | dot = avpriv_scalarproduct_float_c(in, ptr, size); | |
526 | if (dot > optimal_gain) { | |
527 | optimal_gain = dot; | |
528 | best_hist_ptr = ptr; | |
529 | } | |
530 | } while (--ptr >= end); | |
531 | ||
532 | if (optimal_gain <= 0) | |
533 | return -1; | |
534 | dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size); | |
535 | if (dot <= 0) // would be 1.0 | |
536 | return -1; | |
537 | ||
538 | if (optimal_gain <= dot) { | |
539 | dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 | |
540 | } else | |
541 | dot = 0.625; | |
542 | ||
543 | /* actual smoothing */ | |
544 | for (n = 0; n < size; n++) | |
545 | out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); | |
546 | ||
547 | return 0; | |
548 | } | |
549 | ||
550 | /** | |
551 | * Get the tilt factor of a formant filter from its transfer function | |
552 | * @see #tilt_factor() in amrnbdec.c, which does essentially the same, | |
553 | * but somehow (??) it does a speech synthesis filter in the | |
554 | * middle, which is missing here | |
555 | * | |
556 | * @param lpcs LPC coefficients | |
557 | * @param n_lpcs Size of LPC buffer | |
558 | * @returns the tilt factor | |
559 | */ | |
560 | static float tilt_factor(const float *lpcs, int n_lpcs) | |
561 | { | |
562 | float rh0, rh1; | |
563 | ||
564 | rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs); | |
565 | rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1); | |
566 | ||
567 | return rh1 / rh0; | |
568 | } | |
569 | ||
570 | /** | |
571 | * Derive denoise filter coefficients (in real domain) from the LPCs. | |
572 | */ | |
573 | static void calc_input_response(WMAVoiceContext *s, float *lpcs, | |
574 | int fcb_type, float *coeffs, int remainder) | |
575 | { | |
576 | float last_coeff, min = 15.0, max = -15.0; | |
577 | float irange, angle_mul, gain_mul, range, sq; | |
578 | int n, idx; | |
579 | ||
580 | /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ | |
581 | s->rdft.rdft_calc(&s->rdft, lpcs); | |
582 | #define log_range(var, assign) do { \ | |
583 | float tmp = log10f(assign); var = tmp; \ | |
584 | max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ | |
585 | } while (0) | |
586 | log_range(last_coeff, lpcs[1] * lpcs[1]); | |
587 | for (n = 1; n < 64; n++) | |
588 | log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + | |
589 | lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); | |
590 | log_range(lpcs[0], lpcs[0] * lpcs[0]); | |
591 | #undef log_range | |
592 | range = max - min; | |
593 | lpcs[64] = last_coeff; | |
594 | ||
595 | /* Now, use this spectrum to pick out these frequencies with higher | |
596 | * (relative) power/energy (which we then take to be "not noise"), | |
597 | * and set up a table (still in lpc[]) of (relative) gains per frequency. | |
598 | * These frequencies will be maintained, while others ("noise") will be | |
599 | * decreased in the filter output. */ | |
600 | irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] | |
601 | gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : | |
602 | (5.0 / 14.7)); | |
603 | angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); | |
604 | for (n = 0; n <= 64; n++) { | |
605 | float pwr; | |
606 | ||
607 | idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1); | |
608 | pwr = wmavoice_denoise_power_table[s->denoise_strength][idx]; | |
609 | lpcs[n] = angle_mul * pwr; | |
610 | ||
611 | /* 70.57 =~ 1/log10(1.0331663) */ | |
612 | idx = (pwr * gain_mul - 0.0295) * 70.570526123; | |
613 | if (idx > 127) { // fall back if index falls outside table range | |
614 | coeffs[n] = wmavoice_energy_table[127] * | |
615 | powf(1.0331663, idx - 127); | |
616 | } else | |
617 | coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; | |
618 | } | |
619 | ||
620 | /* calculate the Hilbert transform of the gains, which we do (since this | |
621 | * is a sine input) by doing a phase shift (in theory, H(sin())=cos()). | |
622 | * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the | |
623 | * "moment" of the LPCs in this filter. */ | |
624 | s->dct.dct_calc(&s->dct, lpcs); | |
625 | s->dst.dct_calc(&s->dst, lpcs); | |
626 | ||
627 | /* Split out the coefficient indexes into phase/magnitude pairs */ | |
628 | idx = 255 + av_clip(lpcs[64], -255, 255); | |
629 | coeffs[0] = coeffs[0] * s->cos[idx]; | |
630 | idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); | |
631 | last_coeff = coeffs[64] * s->cos[idx]; | |
632 | for (n = 63;; n--) { | |
633 | idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); | |
634 | coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; | |
635 | coeffs[n * 2] = coeffs[n] * s->cos[idx]; | |
636 | ||
637 | if (!--n) break; | |
638 | ||
639 | idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); | |
640 | coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; | |
641 | coeffs[n * 2] = coeffs[n] * s->cos[idx]; | |
642 | } | |
643 | coeffs[1] = last_coeff; | |
644 | ||
645 | /* move into real domain */ | |
646 | s->irdft.rdft_calc(&s->irdft, coeffs); | |
647 | ||
648 | /* tilt correction and normalize scale */ | |
649 | memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); | |
650 | if (s->denoise_tilt_corr) { | |
651 | float tilt_mem = 0; | |
652 | ||
653 | coeffs[remainder - 1] = 0; | |
654 | ff_tilt_compensation(&tilt_mem, | |
655 | -1.8 * tilt_factor(coeffs, remainder - 1), | |
656 | coeffs, remainder); | |
657 | } | |
658 | sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs, | |
659 | remainder)); | |
660 | for (n = 0; n < remainder; n++) | |
661 | coeffs[n] *= sq; | |
662 | } | |
663 | ||
664 | /** | |
665 | * This function applies a Wiener filter on the (noisy) speech signal as | |
666 | * a means to denoise it. | |
667 | * | |
668 | * - take RDFT of LPCs to get the power spectrum of the noise + speech; | |
669 | * - using this power spectrum, calculate (for each frequency) the Wiener | |
670 | * filter gain, which depends on the frequency power and desired level | |
671 | * of noise subtraction (when set too high, this leads to artifacts) | |
672 | * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse | |
673 | * of 4-8kHz); | |
674 | * - by doing a phase shift, calculate the Hilbert transform of this array | |
675 | * of per-frequency filter-gains to get the filtering coefficients; | |
676 | * - smoothen/normalize/de-tilt these filter coefficients as desired; | |
677 | * - take RDFT of noisy sound, apply the coefficients and take its IRDFT | |
678 | * to get the denoised speech signal; | |
679 | * - the leftover (i.e. output of the IRDFT on denoised speech data beyond | |
680 | * the frame boundary) are saved and applied to subsequent frames by an | |
681 | * overlap-add method (otherwise you get clicking-artifacts). | |
682 | * | |
683 | * @param s WMA Voice decoding context | |
684 | * @param fcb_type Frame (codebook) type | |
685 | * @param synth_pf input: the noisy speech signal, output: denoised speech | |
686 | * data; should be 16-byte aligned (for ASM purposes) | |
687 | * @param size size of the speech data | |
688 | * @param lpcs LPCs used to synthesize this frame's speech data | |
689 | */ | |
690 | static void wiener_denoise(WMAVoiceContext *s, int fcb_type, | |
691 | float *synth_pf, int size, | |
692 | const float *lpcs) | |
693 | { | |
694 | int remainder, lim, n; | |
695 | ||
696 | if (fcb_type != FCB_TYPE_SILENCE) { | |
697 | float *tilted_lpcs = s->tilted_lpcs_pf, | |
698 | *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; | |
699 | ||
700 | tilted_lpcs[0] = 1.0; | |
701 | memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); | |
702 | memset(&tilted_lpcs[s->lsps + 1], 0, | |
703 | sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); | |
704 | ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), | |
705 | tilted_lpcs, s->lsps + 2); | |
706 | ||
707 | /* The IRDFT output (127 samples for 7-bit filter) beyond the frame | |
708 | * size is applied to the next frame. All input beyond this is zero, | |
709 | * and thus all output beyond this will go towards zero, hence we can | |
710 | * limit to min(size-1, 127-size) as a performance consideration. */ | |
711 | remainder = FFMIN(127 - size, size - 1); | |
712 | calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); | |
713 | ||
714 | /* apply coefficients (in frequency spectrum domain), i.e. complex | |
715 | * number multiplication */ | |
716 | memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); | |
717 | s->rdft.rdft_calc(&s->rdft, synth_pf); | |
718 | s->rdft.rdft_calc(&s->rdft, coeffs); | |
719 | synth_pf[0] *= coeffs[0]; | |
720 | synth_pf[1] *= coeffs[1]; | |
721 | for (n = 1; n < 64; n++) { | |
722 | float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; | |
723 | synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; | |
724 | synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; | |
725 | } | |
726 | s->irdft.rdft_calc(&s->irdft, synth_pf); | |
727 | } | |
728 | ||
729 | /* merge filter output with the history of previous runs */ | |
730 | if (s->denoise_filter_cache_size) { | |
731 | lim = FFMIN(s->denoise_filter_cache_size, size); | |
732 | for (n = 0; n < lim; n++) | |
733 | synth_pf[n] += s->denoise_filter_cache[n]; | |
734 | s->denoise_filter_cache_size -= lim; | |
735 | memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], | |
736 | sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); | |
737 | } | |
738 | ||
739 | /* move remainder of filter output into a cache for future runs */ | |
740 | if (fcb_type != FCB_TYPE_SILENCE) { | |
741 | lim = FFMIN(remainder, s->denoise_filter_cache_size); | |
742 | for (n = 0; n < lim; n++) | |
743 | s->denoise_filter_cache[n] += synth_pf[size + n]; | |
744 | if (lim < remainder) { | |
745 | memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], | |
746 | sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); | |
747 | s->denoise_filter_cache_size = remainder; | |
748 | } | |
749 | } | |
750 | } | |
751 | ||
752 | /** | |
753 | * Averaging projection filter, the postfilter used in WMAVoice. | |
754 | * | |
755 | * This uses the following steps: | |
756 | * - A zero-synthesis filter (generate excitation from synth signal) | |
757 | * - Kalman smoothing on excitation, based on pitch | |
758 | * - Re-synthesized smoothened output | |
759 | * - Iterative Wiener denoise filter | |
760 | * - Adaptive gain filter | |
761 | * - DC filter | |
762 | * | |
763 | * @param s WMAVoice decoding context | |
764 | * @param synth Speech synthesis output (before postfilter) | |
765 | * @param samples Output buffer for filtered samples | |
766 | * @param size Buffer size of synth & samples | |
767 | * @param lpcs Generated LPCs used for speech synthesis | |
768 | * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned) | |
769 | * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) | |
770 | * @param pitch Pitch of the input signal | |
771 | */ | |
772 | static void postfilter(WMAVoiceContext *s, const float *synth, | |
773 | float *samples, int size, | |
774 | const float *lpcs, float *zero_exc_pf, | |
775 | int fcb_type, int pitch) | |
776 | { | |
777 | float synth_filter_in_buf[MAX_FRAMESIZE / 2], | |
778 | *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], | |
779 | *synth_filter_in = zero_exc_pf; | |
780 | ||
781 | av_assert0(size <= MAX_FRAMESIZE / 2); | |
782 | ||
783 | /* generate excitation from input signal */ | |
784 | ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); | |
785 | ||
786 | if (fcb_type >= FCB_TYPE_AW_PULSES && | |
787 | !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) | |
788 | synth_filter_in = synth_filter_in_buf; | |
789 | ||
790 | /* re-synthesize speech after smoothening, and keep history */ | |
791 | ff_celp_lp_synthesis_filterf(synth_pf, lpcs, | |
792 | synth_filter_in, size, s->lsps); | |
793 | memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], | |
794 | sizeof(synth_pf[0]) * s->lsps); | |
795 | ||
796 | wiener_denoise(s, fcb_type, synth_pf, size, lpcs); | |
797 | ||
798 | adaptive_gain_control(samples, synth_pf, synth, size, 0.99, | |
799 | &s->postfilter_agc); | |
800 | ||
801 | if (s->dc_level > 8) { | |
802 | /* remove ultra-low frequency DC noise / highpass filter; | |
803 | * coefficients are identical to those used in SIPR decoding, | |
804 | * and very closely resemble those used in AMR-NB decoding. */ | |
805 | ff_acelp_apply_order_2_transfer_function(samples, samples, | |
806 | (const float[2]) { -1.99997, 1.0 }, | |
807 | (const float[2]) { -1.9330735188, 0.93589198496 }, | |
808 | 0.93980580475, s->dcf_mem, size); | |
809 | } | |
810 | } | |
811 | /** | |
812 | * @} | |
813 | */ | |
814 | ||
815 | /** | |
816 | * Dequantize LSPs | |
817 | * @param lsps output pointer to the array that will hold the LSPs | |
818 | * @param num number of LSPs to be dequantized | |
819 | * @param values quantized values, contains n_stages values | |
820 | * @param sizes range (i.e. max value) of each quantized value | |
821 | * @param n_stages number of dequantization runs | |
822 | * @param table dequantization table to be used | |
823 | * @param mul_q LSF multiplier | |
824 | * @param base_q base (lowest) LSF values | |
825 | */ | |
826 | static void dequant_lsps(double *lsps, int num, | |
827 | const uint16_t *values, | |
828 | const uint16_t *sizes, | |
829 | int n_stages, const uint8_t *table, | |
830 | const double *mul_q, | |
831 | const double *base_q) | |
832 | { | |
833 | int n, m; | |
834 | ||
835 | memset(lsps, 0, num * sizeof(*lsps)); | |
836 | for (n = 0; n < n_stages; n++) { | |
837 | const uint8_t *t_off = &table[values[n] * num]; | |
838 | double base = base_q[n], mul = mul_q[n]; | |
839 | ||
840 | for (m = 0; m < num; m++) | |
841 | lsps[m] += base + mul * t_off[m]; | |
842 | ||
843 | table += sizes[n] * num; | |
844 | } | |
845 | } | |
846 | ||
847 | /** | |
848 | * @name LSP dequantization routines | |
849 | * LSP dequantization routines, for 10/16LSPs and independent/residual coding. | |
850 | * @note we assume enough bits are available, caller should check. | |
851 | * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; | |
852 | * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. | |
853 | * @{ | |
854 | */ | |
855 | /** | |
856 | * Parse 10 independently-coded LSPs. | |
857 | */ | |
858 | static void dequant_lsp10i(GetBitContext *gb, double *lsps) | |
859 | { | |
860 | static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; | |
861 | static const double mul_lsf[4] = { | |
862 | 5.2187144800e-3, 1.4626986422e-3, | |
863 | 9.6179549166e-4, 1.1325736225e-3 | |
864 | }; | |
865 | static const double base_lsf[4] = { | |
866 | M_PI * -2.15522e-1, M_PI * -6.1646e-2, | |
867 | M_PI * -3.3486e-2, M_PI * -5.7408e-2 | |
868 | }; | |
869 | uint16_t v[4]; | |
870 | ||
871 | v[0] = get_bits(gb, 8); | |
872 | v[1] = get_bits(gb, 6); | |
873 | v[2] = get_bits(gb, 5); | |
874 | v[3] = get_bits(gb, 5); | |
875 | ||
876 | dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, | |
877 | mul_lsf, base_lsf); | |
878 | } | |
879 | ||
880 | /** | |
881 | * Parse 10 independently-coded LSPs, and then derive the tables to | |
882 | * generate LSPs for the other frames from them (residual coding). | |
883 | */ | |
884 | static void dequant_lsp10r(GetBitContext *gb, | |
885 | double *i_lsps, const double *old, | |
886 | double *a1, double *a2, int q_mode) | |
887 | { | |
888 | static const uint16_t vec_sizes[3] = { 128, 64, 64 }; | |
889 | static const double mul_lsf[3] = { | |
890 | 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 | |
891 | }; | |
892 | static const double base_lsf[3] = { | |
893 | M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 | |
894 | }; | |
895 | const float (*ipol_tab)[2][10] = q_mode ? | |
896 | wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; | |
897 | uint16_t interpol, v[3]; | |
898 | int n; | |
899 | ||
900 | dequant_lsp10i(gb, i_lsps); | |
901 | ||
902 | interpol = get_bits(gb, 5); | |
903 | v[0] = get_bits(gb, 7); | |
904 | v[1] = get_bits(gb, 6); | |
905 | v[2] = get_bits(gb, 6); | |
906 | ||
907 | for (n = 0; n < 10; n++) { | |
908 | double delta = old[n] - i_lsps[n]; | |
909 | a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
910 | a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
911 | } | |
912 | ||
913 | dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, | |
914 | mul_lsf, base_lsf); | |
915 | } | |
916 | ||
917 | /** | |
918 | * Parse 16 independently-coded LSPs. | |
919 | */ | |
920 | static void dequant_lsp16i(GetBitContext *gb, double *lsps) | |
921 | { | |
922 | static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; | |
923 | static const double mul_lsf[5] = { | |
924 | 3.3439586280e-3, 6.9908173703e-4, | |
925 | 3.3216608306e-3, 1.0334960326e-3, | |
926 | 3.1899104283e-3 | |
927 | }; | |
928 | static const double base_lsf[5] = { | |
929 | M_PI * -1.27576e-1, M_PI * -2.4292e-2, | |
930 | M_PI * -1.28094e-1, M_PI * -3.2128e-2, | |
931 | M_PI * -1.29816e-1 | |
932 | }; | |
933 | uint16_t v[5]; | |
934 | ||
935 | v[0] = get_bits(gb, 8); | |
936 | v[1] = get_bits(gb, 6); | |
937 | v[2] = get_bits(gb, 7); | |
938 | v[3] = get_bits(gb, 6); | |
939 | v[4] = get_bits(gb, 7); | |
940 | ||
941 | dequant_lsps( lsps, 5, v, vec_sizes, 2, | |
942 | wmavoice_dq_lsp16i1, mul_lsf, base_lsf); | |
943 | dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, | |
944 | wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); | |
945 | dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, | |
946 | wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); | |
947 | } | |
948 | ||
949 | /** | |
950 | * Parse 16 independently-coded LSPs, and then derive the tables to | |
951 | * generate LSPs for the other frames from them (residual coding). | |
952 | */ | |
953 | static void dequant_lsp16r(GetBitContext *gb, | |
954 | double *i_lsps, const double *old, | |
955 | double *a1, double *a2, int q_mode) | |
956 | { | |
957 | static const uint16_t vec_sizes[3] = { 128, 128, 128 }; | |
958 | static const double mul_lsf[3] = { | |
959 | 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 | |
960 | }; | |
961 | static const double base_lsf[3] = { | |
962 | M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 | |
963 | }; | |
964 | const float (*ipol_tab)[2][16] = q_mode ? | |
965 | wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; | |
966 | uint16_t interpol, v[3]; | |
967 | int n; | |
968 | ||
969 | dequant_lsp16i(gb, i_lsps); | |
970 | ||
971 | interpol = get_bits(gb, 5); | |
972 | v[0] = get_bits(gb, 7); | |
973 | v[1] = get_bits(gb, 7); | |
974 | v[2] = get_bits(gb, 7); | |
975 | ||
976 | for (n = 0; n < 16; n++) { | |
977 | double delta = old[n] - i_lsps[n]; | |
978 | a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
979 | a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
980 | } | |
981 | ||
982 | dequant_lsps( a2, 10, v, vec_sizes, 1, | |
983 | wmavoice_dq_lsp16r1, mul_lsf, base_lsf); | |
984 | dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, | |
985 | wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); | |
986 | dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, | |
987 | wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); | |
988 | } | |
989 | ||
990 | /** | |
991 | * @} | |
992 | * @name Pitch-adaptive window coding functions | |
993 | * The next few functions are for pitch-adaptive window coding. | |
994 | * @{ | |
995 | */ | |
996 | /** | |
997 | * Parse the offset of the first pitch-adaptive window pulses, and | |
998 | * the distribution of pulses between the two blocks in this frame. | |
999 | * @param s WMA Voice decoding context private data | |
1000 | * @param gb bit I/O context | |
1001 | * @param pitch pitch for each block in this frame | |
1002 | */ | |
1003 | static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, | |
1004 | const int *pitch) | |
1005 | { | |
1006 | static const int16_t start_offset[94] = { | |
1007 | -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, | |
1008 | 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, | |
1009 | 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, | |
1010 | 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, | |
1011 | 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, | |
1012 | 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, | |
1013 | 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, | |
1014 | 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 | |
1015 | }; | |
1016 | int bits, offset; | |
1017 | ||
1018 | /* position of pulse */ | |
1019 | s->aw_idx_is_ext = 0; | |
1020 | if ((bits = get_bits(gb, 6)) >= 54) { | |
1021 | s->aw_idx_is_ext = 1; | |
1022 | bits += (bits - 54) * 3 + get_bits(gb, 2); | |
1023 | } | |
1024 | ||
1025 | /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count | |
1026 | * the distribution of the pulses in each block contained in this frame. */ | |
1027 | s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; | |
1028 | for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; | |
1029 | s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; | |
1030 | s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; | |
1031 | offset += s->aw_n_pulses[0] * pitch[0]; | |
1032 | s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; | |
1033 | s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; | |
1034 | ||
1035 | /* if continuing from a position before the block, reset position to | |
1036 | * start of block (when corrected for the range over which it can be | |
1037 | * spread in aw_pulse_set1()). */ | |
1038 | if (start_offset[bits] < MAX_FRAMESIZE / 2) { | |
1039 | while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) | |
1040 | s->aw_first_pulse_off[1] -= pitch[1]; | |
1041 | if (start_offset[bits] < 0) | |
1042 | while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) | |
1043 | s->aw_first_pulse_off[0] -= pitch[0]; | |
1044 | } | |
1045 | } | |
1046 | ||
1047 | /** | |
1048 | * Apply second set of pitch-adaptive window pulses. | |
1049 | * @param s WMA Voice decoding context private data | |
1050 | * @param gb bit I/O context | |
1051 | * @param block_idx block index in frame [0, 1] | |
1052 | * @param fcb structure containing fixed codebook vector info | |
1053 | * @return -1 on error, 0 otherwise | |
1054 | */ | |
1055 | static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, | |
1056 | int block_idx, AMRFixed *fcb) | |
1057 | { | |
1058 | uint16_t use_mask_mem[9]; // only 5 are used, rest is padding | |
1059 | uint16_t *use_mask = use_mask_mem + 2; | |
1060 | /* in this function, idx is the index in the 80-bit (+ padding) use_mask | |
1061 | * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits | |
1062 | * of idx are the position of the bit within a particular item in the | |
1063 | * array (0 being the most significant bit, and 15 being the least | |
1064 | * significant bit), and the remainder (>> 4) is the index in the | |
1065 | * use_mask[]-array. This is faster and uses less memory than using a | |
1066 | * 80-byte/80-int array. */ | |
1067 | int pulse_off = s->aw_first_pulse_off[block_idx], | |
1068 | pulse_start, n, idx, range, aidx, start_off = 0; | |
1069 | ||
1070 | /* set offset of first pulse to within this block */ | |
1071 | if (s->aw_n_pulses[block_idx] > 0) | |
1072 | while (pulse_off + s->aw_pulse_range < 1) | |
1073 | pulse_off += fcb->pitch_lag; | |
1074 | ||
1075 | /* find range per pulse */ | |
1076 | if (s->aw_n_pulses[0] > 0) { | |
1077 | if (block_idx == 0) { | |
1078 | range = 32; | |
1079 | } else /* block_idx = 1 */ { | |
1080 | range = 8; | |
1081 | if (s->aw_n_pulses[block_idx] > 0) | |
1082 | pulse_off = s->aw_next_pulse_off_cache; | |
1083 | } | |
1084 | } else | |
1085 | range = 16; | |
1086 | pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; | |
1087 | ||
1088 | /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, | |
1089 | * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus | |
1090 | * we exclude that range from being pulsed again in this function. */ | |
1091 | memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0])); | |
1092 | memset( use_mask, -1, 5 * sizeof(use_mask[0])); | |
1093 | memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); | |
1094 | if (s->aw_n_pulses[block_idx] > 0) | |
1095 | for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { | |
1096 | int excl_range = s->aw_pulse_range; // always 16 or 24 | |
1097 | uint16_t *use_mask_ptr = &use_mask[idx >> 4]; | |
1098 | int first_sh = 16 - (idx & 15); | |
1099 | *use_mask_ptr++ &= 0xFFFFu << first_sh; | |
1100 | excl_range -= first_sh; | |
1101 | if (excl_range >= 16) { | |
1102 | *use_mask_ptr++ = 0; | |
1103 | *use_mask_ptr &= 0xFFFF >> (excl_range - 16); | |
1104 | } else | |
1105 | *use_mask_ptr &= 0xFFFF >> excl_range; | |
1106 | } | |
1107 | ||
1108 | /* find the 'aidx'th offset that is not excluded */ | |
1109 | aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); | |
1110 | for (n = 0; n <= aidx; pulse_start++) { | |
1111 | for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; | |
1112 | if (idx >= MAX_FRAMESIZE / 2) { // find from zero | |
1113 | if (use_mask[0]) idx = 0x0F; | |
1114 | else if (use_mask[1]) idx = 0x1F; | |
1115 | else if (use_mask[2]) idx = 0x2F; | |
1116 | else if (use_mask[3]) idx = 0x3F; | |
1117 | else if (use_mask[4]) idx = 0x4F; | |
1118 | else return -1; | |
1119 | idx -= av_log2_16bit(use_mask[idx >> 4]); | |
1120 | } | |
1121 | if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { | |
1122 | use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); | |
1123 | n++; | |
1124 | start_off = idx; | |
1125 | } | |
1126 | } | |
1127 | ||
1128 | fcb->x[fcb->n] = start_off; | |
1129 | fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; | |
1130 | fcb->n++; | |
1131 | ||
1132 | /* set offset for next block, relative to start of that block */ | |
1133 | n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; | |
1134 | s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; | |
1135 | return 0; | |
1136 | } | |
1137 | ||
1138 | /** | |
1139 | * Apply first set of pitch-adaptive window pulses. | |
1140 | * @param s WMA Voice decoding context private data | |
1141 | * @param gb bit I/O context | |
1142 | * @param block_idx block index in frame [0, 1] | |
1143 | * @param fcb storage location for fixed codebook pulse info | |
1144 | */ | |
1145 | static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, | |
1146 | int block_idx, AMRFixed *fcb) | |
1147 | { | |
1148 | int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); | |
1149 | float v; | |
1150 | ||
1151 | if (s->aw_n_pulses[block_idx] > 0) { | |
1152 | int n, v_mask, i_mask, sh, n_pulses; | |
1153 | ||
1154 | if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each | |
1155 | n_pulses = 3; | |
1156 | v_mask = 8; | |
1157 | i_mask = 7; | |
1158 | sh = 4; | |
1159 | } else { // 4 pulses, 1:sign + 2:index each | |
1160 | n_pulses = 4; | |
1161 | v_mask = 4; | |
1162 | i_mask = 3; | |
1163 | sh = 3; | |
1164 | } | |
1165 | ||
1166 | for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { | |
1167 | fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; | |
1168 | fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + | |
1169 | s->aw_first_pulse_off[block_idx]; | |
1170 | while (fcb->x[fcb->n] < 0) | |
1171 | fcb->x[fcb->n] += fcb->pitch_lag; | |
1172 | if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) | |
1173 | fcb->n++; | |
1174 | } | |
1175 | } else { | |
1176 | int num2 = (val & 0x1FF) >> 1, delta, idx; | |
1177 | ||
1178 | if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } | |
1179 | else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } | |
1180 | else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } | |
1181 | else { delta = 7; idx = num2 + 1 - 3 * 75; } | |
1182 | v = (val & 0x200) ? -1.0 : 1.0; | |
1183 | ||
1184 | fcb->no_repeat_mask |= 3 << fcb->n; | |
1185 | fcb->x[fcb->n] = idx - delta; | |
1186 | fcb->y[fcb->n] = v; | |
1187 | fcb->x[fcb->n + 1] = idx; | |
1188 | fcb->y[fcb->n + 1] = (val & 1) ? -v : v; | |
1189 | fcb->n += 2; | |
1190 | } | |
1191 | } | |
1192 | ||
1193 | /** | |
1194 | * @} | |
1195 | * | |
1196 | * Generate a random number from frame_cntr and block_idx, which will lief | |
1197 | * in the range [0, 1000 - block_size] (so it can be used as an index in a | |
1198 | * table of size 1000 of which you want to read block_size entries). | |
1199 | * | |
1200 | * @param frame_cntr current frame number | |
1201 | * @param block_num current block index | |
1202 | * @param block_size amount of entries we want to read from a table | |
1203 | * that has 1000 entries | |
1204 | * @return a (non-)random number in the [0, 1000 - block_size] range. | |
1205 | */ | |
1206 | static int pRNG(int frame_cntr, int block_num, int block_size) | |
1207 | { | |
1208 | /* array to simplify the calculation of z: | |
1209 | * y = (x % 9) * 5 + 6; | |
1210 | * z = (49995 * x) / y; | |
1211 | * Since y only has 9 values, we can remove the division by using a | |
1212 | * LUT and using FASTDIV-style divisions. For each of the 9 values | |
1213 | * of y, we can rewrite z as: | |
1214 | * z = x * (49995 / y) + x * ((49995 % y) / y) | |
1215 | * In this table, each col represents one possible value of y, the | |
1216 | * first number is 49995 / y, and the second is the FASTDIV variant | |
1217 | * of 49995 % y / y. */ | |
1218 | static const unsigned int div_tbl[9][2] = { | |
1219 | { 8332, 3 * 715827883U }, // y = 6 | |
1220 | { 4545, 0 * 390451573U }, // y = 11 | |
1221 | { 3124, 11 * 268435456U }, // y = 16 | |
1222 | { 2380, 15 * 204522253U }, // y = 21 | |
1223 | { 1922, 23 * 165191050U }, // y = 26 | |
1224 | { 1612, 23 * 138547333U }, // y = 31 | |
1225 | { 1388, 27 * 119304648U }, // y = 36 | |
1226 | { 1219, 16 * 104755300U }, // y = 41 | |
1227 | { 1086, 39 * 93368855U } // y = 46 | |
1228 | }; | |
1229 | unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; | |
1230 | if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, | |
1231 | // so this is effectively a modulo (%) | |
1232 | y = x - 9 * MULH(477218589, x); // x % 9 | |
1233 | z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); | |
1234 | // z = x * 49995 / (y * 5 + 6) | |
1235 | return z % (1000 - block_size); | |
1236 | } | |
1237 | ||
1238 | /** | |
1239 | * Parse hardcoded signal for a single block. | |
1240 | * @note see #synth_block(). | |
1241 | */ | |
1242 | static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, | |
1243 | int block_idx, int size, | |
1244 | const struct frame_type_desc *frame_desc, | |
1245 | float *excitation) | |
1246 | { | |
1247 | float gain; | |
1248 | int n, r_idx; | |
1249 | ||
1250 | av_assert0(size <= MAX_FRAMESIZE); | |
1251 | ||
1252 | /* Set the offset from which we start reading wmavoice_std_codebook */ | |
1253 | if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1254 | r_idx = pRNG(s->frame_cntr, block_idx, size); | |
1255 | gain = s->silence_gain; | |
1256 | } else /* FCB_TYPE_HARDCODED */ { | |
1257 | r_idx = get_bits(gb, 8); | |
1258 | gain = wmavoice_gain_universal[get_bits(gb, 6)]; | |
1259 | } | |
1260 | ||
1261 | /* Clear gain prediction parameters */ | |
1262 | memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); | |
1263 | ||
1264 | /* Apply gain to hardcoded codebook and use that as excitation signal */ | |
1265 | for (n = 0; n < size; n++) | |
1266 | excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; | |
1267 | } | |
1268 | ||
1269 | /** | |
1270 | * Parse FCB/ACB signal for a single block. | |
1271 | * @note see #synth_block(). | |
1272 | */ | |
1273 | static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, | |
1274 | int block_idx, int size, | |
1275 | int block_pitch_sh2, | |
1276 | const struct frame_type_desc *frame_desc, | |
1277 | float *excitation) | |
1278 | { | |
1279 | static const float gain_coeff[6] = { | |
1280 | 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 | |
1281 | }; | |
1282 | float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; | |
1283 | int n, idx, gain_weight; | |
1284 | AMRFixed fcb; | |
1285 | ||
1286 | av_assert0(size <= MAX_FRAMESIZE / 2); | |
1287 | memset(pulses, 0, sizeof(*pulses) * size); | |
1288 | ||
1289 | fcb.pitch_lag = block_pitch_sh2 >> 2; | |
1290 | fcb.pitch_fac = 1.0; | |
1291 | fcb.no_repeat_mask = 0; | |
1292 | fcb.n = 0; | |
1293 | ||
1294 | /* For the other frame types, this is where we apply the innovation | |
1295 | * (fixed) codebook pulses of the speech signal. */ | |
1296 | if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1297 | aw_pulse_set1(s, gb, block_idx, &fcb); | |
1298 | if (aw_pulse_set2(s, gb, block_idx, &fcb)) { | |
1299 | /* Conceal the block with silence and return. | |
1300 | * Skip the correct amount of bits to read the next | |
1301 | * block from the correct offset. */ | |
1302 | int r_idx = pRNG(s->frame_cntr, block_idx, size); | |
1303 | ||
1304 | for (n = 0; n < size; n++) | |
1305 | excitation[n] = | |
1306 | wmavoice_std_codebook[r_idx + n] * s->silence_gain; | |
1307 | skip_bits(gb, 7 + 1); | |
1308 | return; | |
1309 | } | |
1310 | } else /* FCB_TYPE_EXC_PULSES */ { | |
1311 | int offset_nbits = 5 - frame_desc->log_n_blocks; | |
1312 | ||
1313 | fcb.no_repeat_mask = -1; | |
1314 | /* similar to ff_decode_10_pulses_35bits(), but with single pulses | |
1315 | * (instead of double) for a subset of pulses */ | |
1316 | for (n = 0; n < 5; n++) { | |
1317 | float sign; | |
1318 | int pos1, pos2; | |
1319 | ||
1320 | sign = get_bits1(gb) ? 1.0 : -1.0; | |
1321 | pos1 = get_bits(gb, offset_nbits); | |
1322 | fcb.x[fcb.n] = n + 5 * pos1; | |
1323 | fcb.y[fcb.n++] = sign; | |
1324 | if (n < frame_desc->dbl_pulses) { | |
1325 | pos2 = get_bits(gb, offset_nbits); | |
1326 | fcb.x[fcb.n] = n + 5 * pos2; | |
1327 | fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; | |
1328 | } | |
1329 | } | |
1330 | } | |
1331 | ff_set_fixed_vector(pulses, &fcb, 1.0, size); | |
1332 | ||
1333 | /* Calculate gain for adaptive & fixed codebook signal. | |
1334 | * see ff_amr_set_fixed_gain(). */ | |
1335 | idx = get_bits(gb, 7); | |
1336 | fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, | |
1337 | gain_coeff, 6) - | |
1338 | 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); | |
1339 | acb_gain = wmavoice_gain_codebook_acb[idx]; | |
1340 | pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], | |
1341 | -2.9957322736 /* log(0.05) */, | |
1342 | 1.6094379124 /* log(5.0) */); | |
1343 | ||
1344 | gain_weight = 8 >> frame_desc->log_n_blocks; | |
1345 | memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, | |
1346 | sizeof(*s->gain_pred_err) * (6 - gain_weight)); | |
1347 | for (n = 0; n < gain_weight; n++) | |
1348 | s->gain_pred_err[n] = pred_err; | |
1349 | ||
1350 | /* Calculation of adaptive codebook */ | |
1351 | if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1352 | int len; | |
1353 | for (n = 0; n < size; n += len) { | |
1354 | int next_idx_sh16; | |
1355 | int abs_idx = block_idx * size + n; | |
1356 | int pitch_sh16 = (s->last_pitch_val << 16) + | |
1357 | s->pitch_diff_sh16 * abs_idx; | |
1358 | int pitch = (pitch_sh16 + 0x6FFF) >> 16; | |
1359 | int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; | |
1360 | idx = idx_sh16 >> 16; | |
1361 | if (s->pitch_diff_sh16) { | |
1362 | if (s->pitch_diff_sh16 > 0) { | |
1363 | next_idx_sh16 = (idx_sh16) &~ 0xFFFF; | |
1364 | } else | |
1365 | next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; | |
1366 | len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, | |
1367 | 1, size - n); | |
1368 | } else | |
1369 | len = size; | |
1370 | ||
1371 | ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], | |
1372 | wmavoice_ipol1_coeffs, 17, | |
1373 | idx, 9, len); | |
1374 | } | |
1375 | } else /* ACB_TYPE_HAMMING */ { | |
1376 | int block_pitch = block_pitch_sh2 >> 2; | |
1377 | idx = block_pitch_sh2 & 3; | |
1378 | if (idx) { | |
1379 | ff_acelp_interpolatef(excitation, &excitation[-block_pitch], | |
1380 | wmavoice_ipol2_coeffs, 4, | |
1381 | idx, 8, size); | |
1382 | } else | |
1383 | av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, | |
1384 | sizeof(float) * size); | |
1385 | } | |
1386 | ||
1387 | /* Interpolate ACB/FCB and use as excitation signal */ | |
1388 | ff_weighted_vector_sumf(excitation, excitation, pulses, | |
1389 | acb_gain, fcb_gain, size); | |
1390 | } | |
1391 | ||
1392 | /** | |
1393 | * Parse data in a single block. | |
1394 | * @note we assume enough bits are available, caller should check. | |
1395 | * | |
1396 | * @param s WMA Voice decoding context private data | |
1397 | * @param gb bit I/O context | |
1398 | * @param block_idx index of the to-be-read block | |
1399 | * @param size amount of samples to be read in this block | |
1400 | * @param block_pitch_sh2 pitch for this block << 2 | |
1401 | * @param lsps LSPs for (the end of) this frame | |
1402 | * @param prev_lsps LSPs for the last frame | |
1403 | * @param frame_desc frame type descriptor | |
1404 | * @param excitation target memory for the ACB+FCB interpolated signal | |
1405 | * @param synth target memory for the speech synthesis filter output | |
1406 | * @return 0 on success, <0 on error. | |
1407 | */ | |
1408 | static void synth_block(WMAVoiceContext *s, GetBitContext *gb, | |
1409 | int block_idx, int size, | |
1410 | int block_pitch_sh2, | |
1411 | const double *lsps, const double *prev_lsps, | |
1412 | const struct frame_type_desc *frame_desc, | |
1413 | float *excitation, float *synth) | |
1414 | { | |
1415 | double i_lsps[MAX_LSPS]; | |
1416 | float lpcs[MAX_LSPS]; | |
1417 | float fac; | |
1418 | int n; | |
1419 | ||
1420 | if (frame_desc->acb_type == ACB_TYPE_NONE) | |
1421 | synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); | |
1422 | else | |
1423 | synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, | |
1424 | frame_desc, excitation); | |
1425 | ||
1426 | /* convert interpolated LSPs to LPCs */ | |
1427 | fac = (block_idx + 0.5) / frame_desc->n_blocks; | |
1428 | for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1429 | i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); | |
1430 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1431 | ||
1432 | /* Speech synthesis */ | |
1433 | ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); | |
1434 | } | |
1435 | ||
1436 | /** | |
1437 | * Synthesize output samples for a single frame. | |
1438 | * @note we assume enough bits are available, caller should check. | |
1439 | * | |
1440 | * @param ctx WMA Voice decoder context | |
1441 | * @param gb bit I/O context (s->gb or one for cross-packet superframes) | |
1442 | * @param frame_idx Frame number within superframe [0-2] | |
1443 | * @param samples pointer to output sample buffer, has space for at least 160 | |
1444 | * samples | |
1445 | * @param lsps LSP array | |
1446 | * @param prev_lsps array of previous frame's LSPs | |
1447 | * @param excitation target buffer for excitation signal | |
1448 | * @param synth target buffer for synthesized speech data | |
1449 | * @return 0 on success, <0 on error. | |
1450 | */ | |
1451 | static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, | |
1452 | float *samples, | |
1453 | const double *lsps, const double *prev_lsps, | |
1454 | float *excitation, float *synth) | |
1455 | { | |
1456 | WMAVoiceContext *s = ctx->priv_data; | |
1457 | int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val); | |
1458 | int pitch[MAX_BLOCKS], av_uninit(last_block_pitch); | |
1459 | ||
1460 | /* Parse frame type ("frame header"), see frame_descs */ | |
1461 | int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; | |
1462 | ||
1463 | if (bd_idx < 0) { | |
1464 | av_log(ctx, AV_LOG_ERROR, | |
1465 | "Invalid frame type VLC code, skipping\n"); | |
1466 | return AVERROR_INVALIDDATA; | |
1467 | } | |
1468 | ||
1469 | block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; | |
1470 | ||
1471 | /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ | |
1472 | if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { | |
1473 | /* Pitch is provided per frame, which is interpreted as the pitch of | |
1474 | * the last sample of the last block of this frame. We can interpolate | |
1475 | * the pitch of other blocks (and even pitch-per-sample) by gradually | |
1476 | * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ | |
1477 | n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; | |
1478 | log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; | |
1479 | cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); | |
1480 | cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); | |
1481 | if (s->last_acb_type == ACB_TYPE_NONE || | |
1482 | 20 * abs(cur_pitch_val - s->last_pitch_val) > | |
1483 | (cur_pitch_val + s->last_pitch_val)) | |
1484 | s->last_pitch_val = cur_pitch_val; | |
1485 | ||
1486 | /* pitch per block */ | |
1487 | for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1488 | int fac = n * 2 + 1; | |
1489 | ||
1490 | pitch[n] = (MUL16(fac, cur_pitch_val) + | |
1491 | MUL16((n_blocks_x2 - fac), s->last_pitch_val) + | |
1492 | frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; | |
1493 | } | |
1494 | ||
1495 | /* "pitch-diff-per-sample" for calculation of pitch per sample */ | |
1496 | s->pitch_diff_sh16 = | |
1497 | ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; | |
1498 | } | |
1499 | ||
1500 | /* Global gain (if silence) and pitch-adaptive window coordinates */ | |
1501 | switch (frame_descs[bd_idx].fcb_type) { | |
1502 | case FCB_TYPE_SILENCE: | |
1503 | s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; | |
1504 | break; | |
1505 | case FCB_TYPE_AW_PULSES: | |
1506 | aw_parse_coords(s, gb, pitch); | |
1507 | break; | |
1508 | } | |
1509 | ||
1510 | for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1511 | int bl_pitch_sh2; | |
1512 | ||
1513 | /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ | |
1514 | switch (frame_descs[bd_idx].acb_type) { | |
1515 | case ACB_TYPE_HAMMING: { | |
1516 | /* Pitch is given per block. Per-block pitches are encoded as an | |
1517 | * absolute value for the first block, and then delta values | |
1518 | * relative to this value) for all subsequent blocks. The scale of | |
1519 | * this pitch value is semi-logaritmic compared to its use in the | |
1520 | * decoder, so we convert it to normal scale also. */ | |
1521 | int block_pitch, | |
1522 | t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, | |
1523 | t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, | |
1524 | t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; | |
1525 | ||
1526 | if (n == 0) { | |
1527 | block_pitch = get_bits(gb, s->block_pitch_nbits); | |
1528 | } else | |
1529 | block_pitch = last_block_pitch - s->block_delta_pitch_hrange + | |
1530 | get_bits(gb, s->block_delta_pitch_nbits); | |
1531 | /* Convert last_ so that any next delta is within _range */ | |
1532 | last_block_pitch = av_clip(block_pitch, | |
1533 | s->block_delta_pitch_hrange, | |
1534 | s->block_pitch_range - | |
1535 | s->block_delta_pitch_hrange); | |
1536 | ||
1537 | /* Convert semi-log-style scale back to normal scale */ | |
1538 | if (block_pitch < t1) { | |
1539 | bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; | |
1540 | } else { | |
1541 | block_pitch -= t1; | |
1542 | if (block_pitch < t2) { | |
1543 | bl_pitch_sh2 = | |
1544 | (s->block_conv_table[1] << 2) + (block_pitch << 1); | |
1545 | } else { | |
1546 | block_pitch -= t2; | |
1547 | if (block_pitch < t3) { | |
1548 | bl_pitch_sh2 = | |
1549 | (s->block_conv_table[2] + block_pitch) << 2; | |
1550 | } else | |
1551 | bl_pitch_sh2 = s->block_conv_table[3] << 2; | |
1552 | } | |
1553 | } | |
1554 | pitch[n] = bl_pitch_sh2 >> 2; | |
1555 | break; | |
1556 | } | |
1557 | ||
1558 | case ACB_TYPE_ASYMMETRIC: { | |
1559 | bl_pitch_sh2 = pitch[n] << 2; | |
1560 | break; | |
1561 | } | |
1562 | ||
1563 | default: // ACB_TYPE_NONE has no pitch | |
1564 | bl_pitch_sh2 = 0; | |
1565 | break; | |
1566 | } | |
1567 | ||
1568 | synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, | |
1569 | lsps, prev_lsps, &frame_descs[bd_idx], | |
1570 | &excitation[n * block_nsamples], | |
1571 | &synth[n * block_nsamples]); | |
1572 | } | |
1573 | ||
1574 | /* Averaging projection filter, if applicable. Else, just copy samples | |
1575 | * from synthesis buffer */ | |
1576 | if (s->do_apf) { | |
1577 | double i_lsps[MAX_LSPS]; | |
1578 | float lpcs[MAX_LSPS]; | |
1579 | ||
1580 | for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1581 | i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); | |
1582 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1583 | postfilter(s, synth, samples, 80, lpcs, | |
1584 | &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], | |
1585 | frame_descs[bd_idx].fcb_type, pitch[0]); | |
1586 | ||
1587 | for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1588 | i_lsps[n] = cos(lsps[n]); | |
1589 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1590 | postfilter(s, &synth[80], &samples[80], 80, lpcs, | |
1591 | &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], | |
1592 | frame_descs[bd_idx].fcb_type, pitch[0]); | |
1593 | } else | |
1594 | memcpy(samples, synth, 160 * sizeof(synth[0])); | |
1595 | ||
1596 | /* Cache values for next frame */ | |
1597 | s->frame_cntr++; | |
1598 | if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) | |
1599 | s->last_acb_type = frame_descs[bd_idx].acb_type; | |
1600 | switch (frame_descs[bd_idx].acb_type) { | |
1601 | case ACB_TYPE_NONE: | |
1602 | s->last_pitch_val = 0; | |
1603 | break; | |
1604 | case ACB_TYPE_ASYMMETRIC: | |
1605 | s->last_pitch_val = cur_pitch_val; | |
1606 | break; | |
1607 | case ACB_TYPE_HAMMING: | |
1608 | s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; | |
1609 | break; | |
1610 | } | |
1611 | ||
1612 | return 0; | |
1613 | } | |
1614 | ||
1615 | /** | |
1616 | * Ensure minimum value for first item, maximum value for last value, | |
1617 | * proper spacing between each value and proper ordering. | |
1618 | * | |
1619 | * @param lsps array of LSPs | |
1620 | * @param num size of LSP array | |
1621 | * | |
1622 | * @note basically a double version of #ff_acelp_reorder_lsf(), might be | |
1623 | * useful to put in a generic location later on. Parts are also | |
1624 | * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), | |
1625 | * which is in float. | |
1626 | */ | |
1627 | static void stabilize_lsps(double *lsps, int num) | |
1628 | { | |
1629 | int n, m, l; | |
1630 | ||
1631 | /* set minimum value for first, maximum value for last and minimum | |
1632 | * spacing between LSF values. | |
1633 | * Very similar to ff_set_min_dist_lsf(), but in double. */ | |
1634 | lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); | |
1635 | for (n = 1; n < num; n++) | |
1636 | lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); | |
1637 | lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); | |
1638 | ||
1639 | /* reorder (looks like one-time / non-recursed bubblesort). | |
1640 | * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ | |
1641 | for (n = 1; n < num; n++) { | |
1642 | if (lsps[n] < lsps[n - 1]) { | |
1643 | for (m = 1; m < num; m++) { | |
1644 | double tmp = lsps[m]; | |
1645 | for (l = m - 1; l >= 0; l--) { | |
1646 | if (lsps[l] <= tmp) break; | |
1647 | lsps[l + 1] = lsps[l]; | |
1648 | } | |
1649 | lsps[l + 1] = tmp; | |
1650 | } | |
1651 | break; | |
1652 | } | |
1653 | } | |
1654 | } | |
1655 | ||
1656 | /** | |
1657 | * Test if there's enough bits to read 1 superframe. | |
1658 | * | |
1659 | * @param orig_gb bit I/O context used for reading. This function | |
1660 | * does not modify the state of the bitreader; it | |
1661 | * only uses it to copy the current stream position | |
1662 | * @param s WMA Voice decoding context private data | |
1663 | * @return < 0 on error, 1 on not enough bits or 0 if OK. | |
1664 | */ | |
1665 | static int check_bits_for_superframe(GetBitContext *orig_gb, | |
1666 | WMAVoiceContext *s) | |
1667 | { | |
1668 | GetBitContext s_gb, *gb = &s_gb; | |
1669 | int n, need_bits, bd_idx; | |
1670 | const struct frame_type_desc *frame_desc; | |
1671 | ||
1672 | /* initialize a copy */ | |
1673 | init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); | |
1674 | skip_bits_long(gb, get_bits_count(orig_gb)); | |
1675 | av_assert1(get_bits_left(gb) == get_bits_left(orig_gb)); | |
1676 | ||
1677 | /* superframe header */ | |
1678 | if (get_bits_left(gb) < 14) | |
1679 | return 1; | |
1680 | if (!get_bits1(gb)) | |
1681 | return AVERROR(ENOSYS); // WMAPro-in-WMAVoice superframe | |
1682 | if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe | |
1683 | if (s->has_residual_lsps) { // residual LSPs (for all frames) | |
1684 | if (get_bits_left(gb) < s->sframe_lsp_bitsize) | |
1685 | return 1; | |
1686 | skip_bits_long(gb, s->sframe_lsp_bitsize); | |
1687 | } | |
1688 | ||
1689 | /* frames */ | |
1690 | for (n = 0; n < MAX_FRAMES; n++) { | |
1691 | int aw_idx_is_ext = 0; | |
1692 | ||
1693 | if (!s->has_residual_lsps) { // independent LSPs (per-frame) | |
1694 | if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; | |
1695 | skip_bits_long(gb, s->frame_lsp_bitsize); | |
1696 | } | |
1697 | bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; | |
1698 | if (bd_idx < 0) | |
1699 | return AVERROR_INVALIDDATA; // invalid frame type VLC code | |
1700 | frame_desc = &frame_descs[bd_idx]; | |
1701 | if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1702 | if (get_bits_left(gb) < s->pitch_nbits) | |
1703 | return 1; | |
1704 | skip_bits_long(gb, s->pitch_nbits); | |
1705 | } | |
1706 | if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1707 | skip_bits(gb, 8); | |
1708 | } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1709 | int tmp = get_bits(gb, 6); | |
1710 | if (tmp >= 0x36) { | |
1711 | skip_bits(gb, 2); | |
1712 | aw_idx_is_ext = 1; | |
1713 | } | |
1714 | } | |
1715 | ||
1716 | /* blocks */ | |
1717 | if (frame_desc->acb_type == ACB_TYPE_HAMMING) { | |
1718 | need_bits = s->block_pitch_nbits + | |
1719 | (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; | |
1720 | } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1721 | need_bits = 2 * !aw_idx_is_ext; | |
1722 | } else | |
1723 | need_bits = 0; | |
1724 | need_bits += frame_desc->frame_size; | |
1725 | if (get_bits_left(gb) < need_bits) | |
1726 | return 1; | |
1727 | skip_bits_long(gb, need_bits); | |
1728 | } | |
1729 | ||
1730 | return 0; | |
1731 | } | |
1732 | ||
1733 | /** | |
1734 | * Synthesize output samples for a single superframe. If we have any data | |
1735 | * cached in s->sframe_cache, that will be used instead of whatever is loaded | |
1736 | * in s->gb. | |
1737 | * | |
1738 | * WMA Voice superframes contain 3 frames, each containing 160 audio samples, | |
1739 | * to give a total of 480 samples per frame. See #synth_frame() for frame | |
1740 | * parsing. In addition to 3 frames, superframes can also contain the LSPs | |
1741 | * (if these are globally specified for all frames (residually); they can | |
1742 | * also be specified individually per-frame. See the s->has_residual_lsps | |
1743 | * option), and can specify the number of samples encoded in this superframe | |
1744 | * (if less than 480), usually used to prevent blanks at track boundaries. | |
1745 | * | |
1746 | * @param ctx WMA Voice decoder context | |
1747 | * @return 0 on success, <0 on error or 1 if there was not enough data to | |
1748 | * fully parse the superframe | |
1749 | */ | |
1750 | static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, | |
1751 | int *got_frame_ptr) | |
1752 | { | |
1753 | WMAVoiceContext *s = ctx->priv_data; | |
1754 | GetBitContext *gb = &s->gb, s_gb; | |
1755 | int n, res, n_samples = 480; | |
1756 | double lsps[MAX_FRAMES][MAX_LSPS]; | |
1757 | const double *mean_lsf = s->lsps == 16 ? | |
1758 | wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; | |
1759 | float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; | |
1760 | float synth[MAX_LSPS + MAX_SFRAMESIZE]; | |
1761 | float *samples; | |
1762 | ||
1763 | memcpy(synth, s->synth_history, | |
1764 | s->lsps * sizeof(*synth)); | |
1765 | memcpy(excitation, s->excitation_history, | |
1766 | s->history_nsamples * sizeof(*excitation)); | |
1767 | ||
1768 | if (s->sframe_cache_size > 0) { | |
1769 | gb = &s_gb; | |
1770 | init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); | |
1771 | s->sframe_cache_size = 0; | |
1772 | } | |
1773 | ||
1774 | if ((res = check_bits_for_superframe(gb, s)) == 1) { | |
1775 | *got_frame_ptr = 0; | |
1776 | return 1; | |
1777 | } else if (res < 0) | |
1778 | return res; | |
1779 | ||
1780 | /* First bit is speech/music bit, it differentiates between WMAVoice | |
1781 | * speech samples (the actual codec) and WMAVoice music samples, which | |
1782 | * are really WMAPro-in-WMAVoice-superframes. I've never seen those in | |
1783 | * the wild yet. */ | |
1784 | if (!get_bits1(gb)) { | |
1785 | avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); | |
1786 | return AVERROR_PATCHWELCOME; | |
1787 | } | |
1788 | ||
1789 | /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ | |
1790 | if (get_bits1(gb)) { | |
1791 | if ((n_samples = get_bits(gb, 12)) > 480) { | |
1792 | av_log(ctx, AV_LOG_ERROR, | |
1793 | "Superframe encodes >480 samples (%d), not allowed\n", | |
1794 | n_samples); | |
1795 | return AVERROR_INVALIDDATA; | |
1796 | } | |
1797 | } | |
1798 | /* Parse LSPs, if global for the superframe (can also be per-frame). */ | |
1799 | if (s->has_residual_lsps) { | |
1800 | double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; | |
1801 | ||
1802 | for (n = 0; n < s->lsps; n++) | |
1803 | prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; | |
1804 | ||
1805 | if (s->lsps == 10) { | |
1806 | dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1807 | } else /* s->lsps == 16 */ | |
1808 | dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1809 | ||
1810 | for (n = 0; n < s->lsps; n++) { | |
1811 | lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); | |
1812 | lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); | |
1813 | lsps[2][n] += mean_lsf[n]; | |
1814 | } | |
1815 | for (n = 0; n < 3; n++) | |
1816 | stabilize_lsps(lsps[n], s->lsps); | |
1817 | } | |
1818 | ||
1819 | /* get output buffer */ | |
1820 | frame->nb_samples = 480; | |
1821 | if ((res = ff_get_buffer(ctx, frame, 0)) < 0) | |
1822 | return res; | |
1823 | frame->nb_samples = n_samples; | |
1824 | samples = (float *)frame->data[0]; | |
1825 | ||
1826 | /* Parse frames, optionally preceded by per-frame (independent) LSPs. */ | |
1827 | for (n = 0; n < 3; n++) { | |
1828 | if (!s->has_residual_lsps) { | |
1829 | int m; | |
1830 | ||
1831 | if (s->lsps == 10) { | |
1832 | dequant_lsp10i(gb, lsps[n]); | |
1833 | } else /* s->lsps == 16 */ | |
1834 | dequant_lsp16i(gb, lsps[n]); | |
1835 | ||
1836 | for (m = 0; m < s->lsps; m++) | |
1837 | lsps[n][m] += mean_lsf[m]; | |
1838 | stabilize_lsps(lsps[n], s->lsps); | |
1839 | } | |
1840 | ||
1841 | if ((res = synth_frame(ctx, gb, n, | |
1842 | &samples[n * MAX_FRAMESIZE], | |
1843 | lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], | |
1844 | &excitation[s->history_nsamples + n * MAX_FRAMESIZE], | |
1845 | &synth[s->lsps + n * MAX_FRAMESIZE]))) { | |
1846 | *got_frame_ptr = 0; | |
1847 | return res; | |
1848 | } | |
1849 | } | |
1850 | ||
1851 | /* Statistics? FIXME - we don't check for length, a slight overrun | |
1852 | * will be caught by internal buffer padding, and anything else | |
1853 | * will be skipped, not read. */ | |
1854 | if (get_bits1(gb)) { | |
1855 | res = get_bits(gb, 4); | |
1856 | skip_bits(gb, 10 * (res + 1)); | |
1857 | } | |
1858 | ||
1859 | *got_frame_ptr = 1; | |
1860 | ||
1861 | /* Update history */ | |
1862 | memcpy(s->prev_lsps, lsps[2], | |
1863 | s->lsps * sizeof(*s->prev_lsps)); | |
1864 | memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], | |
1865 | s->lsps * sizeof(*synth)); | |
1866 | memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], | |
1867 | s->history_nsamples * sizeof(*excitation)); | |
1868 | if (s->do_apf) | |
1869 | memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], | |
1870 | s->history_nsamples * sizeof(*s->zero_exc_pf)); | |
1871 | ||
1872 | return 0; | |
1873 | } | |
1874 | ||
1875 | /** | |
1876 | * Parse the packet header at the start of each packet (input data to this | |
1877 | * decoder). | |
1878 | * | |
1879 | * @param s WMA Voice decoding context private data | |
1880 | * @return 1 if not enough bits were available, or 0 on success. | |
1881 | */ | |
1882 | static int parse_packet_header(WMAVoiceContext *s) | |
1883 | { | |
1884 | GetBitContext *gb = &s->gb; | |
1885 | unsigned int res; | |
1886 | ||
1887 | if (get_bits_left(gb) < 11) | |
1888 | return 1; | |
1889 | skip_bits(gb, 4); // packet sequence number | |
1890 | s->has_residual_lsps = get_bits1(gb); | |
1891 | do { | |
1892 | res = get_bits(gb, 6); // number of superframes per packet | |
1893 | // (minus first one if there is spillover) | |
1894 | if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) | |
1895 | return 1; | |
1896 | } while (res == 0x3F); | |
1897 | s->spillover_nbits = get_bits(gb, s->spillover_bitsize); | |
1898 | ||
1899 | return 0; | |
1900 | } | |
1901 | ||
1902 | /** | |
1903 | * Copy (unaligned) bits from gb/data/size to pb. | |
1904 | * | |
1905 | * @param pb target buffer to copy bits into | |
1906 | * @param data source buffer to copy bits from | |
1907 | * @param size size of the source data, in bytes | |
1908 | * @param gb bit I/O context specifying the current position in the source. | |
1909 | * data. This function might use this to align the bit position to | |
1910 | * a whole-byte boundary before calling #avpriv_copy_bits() on aligned | |
1911 | * source data | |
1912 | * @param nbits the amount of bits to copy from source to target | |
1913 | * | |
1914 | * @note after calling this function, the current position in the input bit | |
1915 | * I/O context is undefined. | |
1916 | */ | |
1917 | static void copy_bits(PutBitContext *pb, | |
1918 | const uint8_t *data, int size, | |
1919 | GetBitContext *gb, int nbits) | |
1920 | { | |
1921 | int rmn_bytes, rmn_bits; | |
1922 | ||
1923 | rmn_bits = rmn_bytes = get_bits_left(gb); | |
1924 | if (rmn_bits < nbits) | |
1925 | return; | |
1926 | if (nbits > pb->size_in_bits - put_bits_count(pb)) | |
1927 | return; | |
1928 | rmn_bits &= 7; rmn_bytes >>= 3; | |
1929 | if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) | |
1930 | put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); | |
1931 | avpriv_copy_bits(pb, data + size - rmn_bytes, | |
1932 | FFMIN(nbits - rmn_bits, rmn_bytes << 3)); | |
1933 | } | |
1934 | ||
1935 | /** | |
1936 | * Packet decoding: a packet is anything that the (ASF) demuxer contains, | |
1937 | * and we expect that the demuxer / application provides it to us as such | |
1938 | * (else you'll probably get garbage as output). Every packet has a size of | |
1939 | * ctx->block_align bytes, starts with a packet header (see | |
1940 | * #parse_packet_header()), and then a series of superframes. Superframe | |
1941 | * boundaries may exceed packets, i.e. superframes can split data over | |
1942 | * multiple (two) packets. | |
1943 | * | |
1944 | * For more information about frames, see #synth_superframe(). | |
1945 | */ | |
1946 | static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, | |
1947 | int *got_frame_ptr, AVPacket *avpkt) | |
1948 | { | |
1949 | WMAVoiceContext *s = ctx->priv_data; | |
1950 | GetBitContext *gb = &s->gb; | |
1951 | int size, res, pos; | |
1952 | ||
1953 | /* Packets are sometimes a multiple of ctx->block_align, with a packet | |
1954 | * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer | |
1955 | * feeds us ASF packets, which may concatenate multiple "codec" packets | |
1956 | * in a single "muxer" packet, so we artificially emulate that by | |
1957 | * capping the packet size at ctx->block_align. */ | |
1958 | for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); | |
1959 | if (!size) { | |
1960 | *got_frame_ptr = 0; | |
1961 | return 0; | |
1962 | } | |
1963 | init_get_bits(&s->gb, avpkt->data, size << 3); | |
1964 | ||
1965 | /* size == ctx->block_align is used to indicate whether we are dealing with | |
1966 | * a new packet or a packet of which we already read the packet header | |
1967 | * previously. */ | |
1968 | if (size == ctx->block_align) { // new packet header | |
1969 | if ((res = parse_packet_header(s)) < 0) | |
1970 | return res; | |
1971 | ||
1972 | /* If the packet header specifies a s->spillover_nbits, then we want | |
1973 | * to push out all data of the previous packet (+ spillover) before | |
1974 | * continuing to parse new superframes in the current packet. */ | |
1975 | if (s->spillover_nbits > 0) { | |
1976 | if (s->sframe_cache_size > 0) { | |
1977 | int cnt = get_bits_count(gb); | |
1978 | copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); | |
1979 | flush_put_bits(&s->pb); | |
1980 | s->sframe_cache_size += s->spillover_nbits; | |
1981 | if ((res = synth_superframe(ctx, data, got_frame_ptr)) == 0 && | |
1982 | *got_frame_ptr) { | |
1983 | cnt += s->spillover_nbits; | |
1984 | s->skip_bits_next = cnt & 7; | |
1985 | return cnt >> 3; | |
1986 | } else | |
1987 | skip_bits_long (gb, s->spillover_nbits - cnt + | |
1988 | get_bits_count(gb)); // resync | |
1989 | } else | |
1990 | skip_bits_long(gb, s->spillover_nbits); // resync | |
1991 | } | |
1992 | } else if (s->skip_bits_next) | |
1993 | skip_bits(gb, s->skip_bits_next); | |
1994 | ||
1995 | /* Try parsing superframes in current packet */ | |
1996 | s->sframe_cache_size = 0; | |
1997 | s->skip_bits_next = 0; | |
1998 | pos = get_bits_left(gb); | |
1999 | if ((res = synth_superframe(ctx, data, got_frame_ptr)) < 0) { | |
2000 | return res; | |
2001 | } else if (*got_frame_ptr) { | |
2002 | int cnt = get_bits_count(gb); | |
2003 | s->skip_bits_next = cnt & 7; | |
2004 | return cnt >> 3; | |
2005 | } else if ((s->sframe_cache_size = pos) > 0) { | |
2006 | /* rewind bit reader to start of last (incomplete) superframe... */ | |
2007 | init_get_bits(gb, avpkt->data, size << 3); | |
2008 | skip_bits_long(gb, (size << 3) - pos); | |
2009 | av_assert1(get_bits_left(gb) == pos); | |
2010 | ||
2011 | /* ...and cache it for spillover in next packet */ | |
2012 | init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); | |
2013 | copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); | |
2014 | // FIXME bad - just copy bytes as whole and add use the | |
2015 | // skip_bits_next field | |
2016 | } | |
2017 | ||
2018 | return size; | |
2019 | } | |
2020 | ||
2021 | static av_cold int wmavoice_decode_end(AVCodecContext *ctx) | |
2022 | { | |
2023 | WMAVoiceContext *s = ctx->priv_data; | |
2024 | ||
2025 | if (s->do_apf) { | |
2026 | ff_rdft_end(&s->rdft); | |
2027 | ff_rdft_end(&s->irdft); | |
2028 | ff_dct_end(&s->dct); | |
2029 | ff_dct_end(&s->dst); | |
2030 | } | |
2031 | ||
2032 | return 0; | |
2033 | } | |
2034 | ||
2035 | static av_cold void wmavoice_flush(AVCodecContext *ctx) | |
2036 | { | |
2037 | WMAVoiceContext *s = ctx->priv_data; | |
2038 | int n; | |
2039 | ||
2040 | s->postfilter_agc = 0; | |
2041 | s->sframe_cache_size = 0; | |
2042 | s->skip_bits_next = 0; | |
2043 | for (n = 0; n < s->lsps; n++) | |
2044 | s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
2045 | memset(s->excitation_history, 0, | |
2046 | sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); | |
2047 | memset(s->synth_history, 0, | |
2048 | sizeof(*s->synth_history) * MAX_LSPS); | |
2049 | memset(s->gain_pred_err, 0, | |
2050 | sizeof(s->gain_pred_err)); | |
2051 | ||
2052 | if (s->do_apf) { | |
2053 | memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, | |
2054 | sizeof(*s->synth_filter_out_buf) * s->lsps); | |
2055 | memset(s->dcf_mem, 0, | |
2056 | sizeof(*s->dcf_mem) * 2); | |
2057 | memset(s->zero_exc_pf, 0, | |
2058 | sizeof(*s->zero_exc_pf) * s->history_nsamples); | |
2059 | memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); | |
2060 | } | |
2061 | } | |
2062 | ||
2063 | AVCodec ff_wmavoice_decoder = { | |
2064 | .name = "wmavoice", | |
2065 | .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), | |
2066 | .type = AVMEDIA_TYPE_AUDIO, | |
2067 | .id = AV_CODEC_ID_WMAVOICE, | |
2068 | .priv_data_size = sizeof(WMAVoiceContext), | |
2069 | .init = wmavoice_decode_init, | |
2070 | .init_static_data = wmavoice_init_static_data, | |
2071 | .close = wmavoice_decode_end, | |
2072 | .decode = wmavoice_decode_packet, | |
2073 | .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1, | |
2074 | .flush = wmavoice_flush, | |
2075 | }; |