Directory: | ../../../ffmpeg/ |
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File: | src/libavcodec/wmavoice.c |
Date: | 2022-08-10 20:23:34 |
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Lines: | 664 | 753 | 88.2% |
Branches: | 305 | 390 | 78.2% |
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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_internal.h" | ||
33 | #include "libavutil/thread.h" | ||
34 | #include "avcodec.h" | ||
35 | #include "codec_internal.h" | ||
36 | #include "internal.h" | ||
37 | #include "get_bits.h" | ||
38 | #include "put_bits.h" | ||
39 | #include "wmavoice_data.h" | ||
40 | #include "celp_filters.h" | ||
41 | #include "acelp_vectors.h" | ||
42 | #include "acelp_filters.h" | ||
43 | #include "lsp.h" | ||
44 | #include "dct.h" | ||
45 | #include "rdft.h" | ||
46 | #include "sinewin.h" | ||
47 | |||
48 | #define MAX_BLOCKS 8 ///< maximum number of blocks per frame | ||
49 | #define MAX_LSPS 16 ///< maximum filter order | ||
50 | #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple | ||
51 | ///< of 16 for ASM input buffer alignment | ||
52 | #define MAX_FRAMES 3 ///< maximum number of frames per superframe | ||
53 | #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame | ||
54 | #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history | ||
55 | #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) | ||
56 | ///< maximum number of samples per superframe | ||
57 | #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that | ||
58 | ///< was split over two packets | ||
59 | #define VLC_NBITS 6 ///< number of bits to read per VLC iteration | ||
60 | |||
61 | /** | ||
62 | * Frame type VLC coding. | ||
63 | */ | ||
64 | static VLC frame_type_vlc; | ||
65 | |||
66 | /** | ||
67 | * Adaptive codebook types. | ||
68 | */ | ||
69 | enum { | ||
70 | ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) | ||
71 | ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which | ||
72 | ///< we interpolate to get a per-sample pitch. | ||
73 | ///< Signal is generated using an asymmetric sinc | ||
74 | ///< window function | ||
75 | ///< @note see #wmavoice_ipol1_coeffs | ||
76 | ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using | ||
77 | ///< a Hamming sinc window function | ||
78 | ///< @note see #wmavoice_ipol2_coeffs | ||
79 | }; | ||
80 | |||
81 | /** | ||
82 | * Fixed codebook types. | ||
83 | */ | ||
84 | enum { | ||
85 | FCB_TYPE_SILENCE = 0, ///< comfort noise during silence | ||
86 | ///< generated from a hardcoded (fixed) codebook | ||
87 | ///< with per-frame (low) gain values | ||
88 | FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block | ||
89 | ///< gain values | ||
90 | FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, | ||
91 | ///< used in particular for low-bitrate streams | ||
92 | FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in | ||
93 | ///< combinations of either single pulses or | ||
94 | ///< pulse pairs | ||
95 | }; | ||
96 | |||
97 | /** | ||
98 | * Description of frame types. | ||
99 | */ | ||
100 | static const struct frame_type_desc { | ||
101 | uint8_t n_blocks; ///< amount of blocks per frame (each block | ||
102 | ///< (contains 160/#n_blocks samples) | ||
103 | uint8_t log_n_blocks; ///< log2(#n_blocks) | ||
104 | uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) | ||
105 | uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) | ||
106 | uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs | ||
107 | ///< (rather than just one single pulse) | ||
108 | ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES | ||
109 | } frame_descs[17] = { | ||
110 | { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0 }, | ||
111 | { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0 }, | ||
112 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0 }, | ||
113 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, | ||
114 | { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, | ||
115 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0 }, | ||
116 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2 }, | ||
117 | { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5 }, | ||
118 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, | ||
119 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, | ||
120 | { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, | ||
121 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, | ||
122 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, | ||
123 | { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 }, | ||
124 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0 }, | ||
125 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2 }, | ||
126 | { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5 } | ||
127 | }; | ||
128 | |||
129 | /** | ||
130 | * WMA Voice decoding context. | ||
131 | */ | ||
132 | typedef struct WMAVoiceContext { | ||
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 | |||
164 | int min_pitch_val; ///< base value for pitch parsing code | ||
165 | int max_pitch_val; ///< max value + 1 for pitch parsing | ||
166 | int pitch_nbits; ///< number of bits used to specify the | ||
167 | ///< pitch value in the frame header | ||
168 | int block_pitch_nbits; ///< number of bits used to specify the | ||
169 | ///< first block's pitch value | ||
170 | int block_pitch_range; ///< range of the block pitch | ||
171 | int block_delta_pitch_nbits; ///< number of bits used to specify the | ||
172 | ///< delta pitch between this and the last | ||
173 | ///< block's pitch value, used in all but | ||
174 | ///< first block | ||
175 | int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is | ||
176 | ///< from -this to +this-1) | ||
177 | uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale | ||
178 | ///< conversion | ||
179 | |||
180 | /** | ||
181 | * @} | ||
182 | * | ||
183 | * @name Packet values specified in the packet header or related to a packet. | ||
184 | * | ||
185 | * A packet is considered to be a single unit of data provided to this | ||
186 | * decoder by the demuxer. | ||
187 | * @{ | ||
188 | */ | ||
189 | int spillover_nbits; ///< number of bits of the previous packet's | ||
190 | ///< last superframe preceding this | ||
191 | ///< packet's first full superframe (useful | ||
192 | ///< for re-synchronization also) | ||
193 | int has_residual_lsps; ///< if set, superframes contain one set of | ||
194 | ///< LSPs that cover all frames, encoded as | ||
195 | ///< independent and residual LSPs; if not | ||
196 | ///< set, each frame contains its own, fully | ||
197 | ///< independent, LSPs | ||
198 | int skip_bits_next; ///< number of bits to skip at the next call | ||
199 | ///< to #wmavoice_decode_packet() (since | ||
200 | ///< they're part of the previous superframe) | ||
201 | |||
202 | uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + AV_INPUT_BUFFER_PADDING_SIZE]; | ||
203 | ///< cache for superframe data split over | ||
204 | ///< multiple packets | ||
205 | int sframe_cache_size; ///< set to >0 if we have data from an | ||
206 | ///< (incomplete) superframe from a previous | ||
207 | ///< packet that spilled over in the current | ||
208 | ///< packet; specifies the amount of bits in | ||
209 | ///< #sframe_cache | ||
210 | PutBitContext pb; ///< bitstream writer for #sframe_cache | ||
211 | |||
212 | /** | ||
213 | * @} | ||
214 | * | ||
215 | * @name Frame and superframe values | ||
216 | * Superframe and frame data - these can change from frame to frame, | ||
217 | * although some of them do in that case serve as a cache / history for | ||
218 | * the next frame or superframe. | ||
219 | * @{ | ||
220 | */ | ||
221 | double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous | ||
222 | ///< superframe | ||
223 | int last_pitch_val; ///< pitch value of the previous frame | ||
224 | int last_acb_type; ///< frame type [0-2] of the previous frame | ||
225 | int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) | ||
226 | ///< << 16) / #MAX_FRAMESIZE | ||
227 | float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE | ||
228 | |||
229 | int aw_idx_is_ext; ///< whether the AW index was encoded in | ||
230 | ///< 8 bits (instead of 6) | ||
231 | int aw_pulse_range; ///< the range over which #aw_pulse_set1() | ||
232 | ///< can apply the pulse, relative to the | ||
233 | ///< value in aw_first_pulse_off. The exact | ||
234 | ///< position of the first AW-pulse is within | ||
235 | ///< [pulse_off, pulse_off + this], and | ||
236 | ///< depends on bitstream values; [16 or 24] | ||
237 | int aw_n_pulses[2]; ///< number of AW-pulses in each block; note | ||
238 | ///< that this number can be negative (in | ||
239 | ///< which case it basically means "zero") | ||
240 | int aw_first_pulse_off[2]; ///< index of first sample to which to | ||
241 | ///< apply AW-pulses, or -0xff if unset | ||
242 | int aw_next_pulse_off_cache; ///< the position (relative to start of the | ||
243 | ///< second block) at which pulses should | ||
244 | ///< start to be positioned, serves as a | ||
245 | ///< cache for pitch-adaptive window pulses | ||
246 | ///< between blocks | ||
247 | |||
248 | int frame_cntr; ///< current frame index [0 - 0xFFFE]; is | ||
249 | ///< only used for comfort noise in #pRNG() | ||
250 | int nb_superframes; ///< number of superframes in current packet | ||
251 | float gain_pred_err[6]; ///< cache for gain prediction | ||
252 | float excitation_history[MAX_SIGNAL_HISTORY]; | ||
253 | ///< cache of the signal of previous | ||
254 | ///< superframes, used as a history for | ||
255 | ///< signal generation | ||
256 | float synth_history[MAX_LSPS]; ///< see #excitation_history | ||
257 | /** | ||
258 | * @} | ||
259 | * | ||
260 | * @name Postfilter values | ||
261 | * | ||
262 | * Variables used for postfilter implementation, mostly history for | ||
263 | * smoothing and so on, and context variables for FFT/iFFT. | ||
264 | * @{ | ||
265 | */ | ||
266 | RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the | ||
267 | ///< postfilter (for denoise filter) | ||
268 | DCTContext dct, dst; ///< contexts for phase shift (in Hilbert | ||
269 | ///< transform, part of postfilter) | ||
270 | float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] | ||
271 | ///< range | ||
272 | float postfilter_agc; ///< gain control memory, used in | ||
273 | ///< #adaptive_gain_control() | ||
274 | float dcf_mem[2]; ///< DC filter history | ||
275 | float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; | ||
276 | ///< zero filter output (i.e. excitation) | ||
277 | ///< by postfilter | ||
278 | float denoise_filter_cache[MAX_FRAMESIZE]; | ||
279 | int denoise_filter_cache_size; ///< samples in #denoise_filter_cache | ||
280 | DECLARE_ALIGNED(32, float, tilted_lpcs_pf)[0x80]; | ||
281 | ///< aligned buffer for LPC tilting | ||
282 | DECLARE_ALIGNED(32, float, denoise_coeffs_pf)[0x80]; | ||
283 | ///< aligned buffer for denoise coefficients | ||
284 | DECLARE_ALIGNED(32, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; | ||
285 | ///< aligned buffer for postfilter speech | ||
286 | ///< synthesis | ||
287 | /** | ||
288 | * @} | ||
289 | */ | ||
290 | } WMAVoiceContext; | ||
291 | |||
292 | /** | ||
293 | * Set up the variable bit mode (VBM) tree from container extradata. | ||
294 | * @param gb bit I/O context. | ||
295 | * The bit context (s->gb) should be loaded with byte 23-46 of the | ||
296 | * container extradata (i.e. the ones containing the VBM tree). | ||
297 | * @param vbm_tree pointer to array to which the decoded VBM tree will be | ||
298 | * written. | ||
299 | * @return 0 on success, <0 on error. | ||
300 | */ | ||
301 | 8 | static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) | |
302 | { | ||
303 | 8 | int cntr[8] = { 0 }, n, res; | |
304 | |||
305 | 8 | memset(vbm_tree, 0xff, sizeof(vbm_tree[0]) * 25); | |
306 |
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144 | for (n = 0; n < 17; n++) { |
307 | 136 | res = get_bits(gb, 3); | |
308 |
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136 | if (cntr[res] > 3) // should be >= 3 + (res == 7)) |
309 | ✗ | return -1; | |
310 | 136 | vbm_tree[res * 3 + cntr[res]++] = n; | |
311 | } | ||
312 | 8 | return 0; | |
313 | } | ||
314 | |||
315 | 5 | static av_cold void wmavoice_init_static_data(void) | |
316 | { | ||
317 | static const uint8_t bits[] = { | ||
318 | 2, 2, 2, 4, 4, 4, | ||
319 | 6, 6, 6, 8, 8, 8, | ||
320 | 10, 10, 10, 12, 12, 12, | ||
321 | 14, 14, 14, 14 | ||
322 | }; | ||
323 | static const uint16_t codes[] = { | ||
324 | 0x0000, 0x0001, 0x0002, // 00/01/10 | ||
325 | 0x000c, 0x000d, 0x000e, // 11+00/01/10 | ||
326 | 0x003c, 0x003d, 0x003e, // 1111+00/01/10 | ||
327 | 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 | ||
328 | 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 | ||
329 | 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 | ||
330 | 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx | ||
331 | }; | ||
332 | |||
333 | 5 | INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), | |
334 | bits, 1, 1, codes, 2, 2, 132); | ||
335 | 5 | } | |
336 | |||
337 | ✗ | static av_cold void wmavoice_flush(AVCodecContext *ctx) | |
338 | { | ||
339 | ✗ | WMAVoiceContext *s = ctx->priv_data; | |
340 | int n; | ||
341 | |||
342 | ✗ | s->postfilter_agc = 0; | |
343 | ✗ | s->sframe_cache_size = 0; | |
344 | ✗ | s->skip_bits_next = 0; | |
345 | ✗ | for (n = 0; n < s->lsps; n++) | |
346 | ✗ | s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
347 | ✗ | memset(s->excitation_history, 0, | |
348 | sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); | ||
349 | ✗ | memset(s->synth_history, 0, | |
350 | sizeof(*s->synth_history) * MAX_LSPS); | ||
351 | ✗ | memset(s->gain_pred_err, 0, | |
352 | sizeof(s->gain_pred_err)); | ||
353 | |||
354 | ✗ | if (s->do_apf) { | |
355 | ✗ | memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, | |
356 | ✗ | sizeof(*s->synth_filter_out_buf) * s->lsps); | |
357 | ✗ | memset(s->dcf_mem, 0, | |
358 | sizeof(*s->dcf_mem) * 2); | ||
359 | ✗ | memset(s->zero_exc_pf, 0, | |
360 | ✗ | sizeof(*s->zero_exc_pf) * s->history_nsamples); | |
361 | ✗ | memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); | |
362 | } | ||
363 | } | ||
364 | |||
365 | /** | ||
366 | * Set up decoder with parameters from demuxer (extradata etc.). | ||
367 | */ | ||
368 | 8 | static av_cold int wmavoice_decode_init(AVCodecContext *ctx) | |
369 | { | ||
370 | static AVOnce init_static_once = AV_ONCE_INIT; | ||
371 | int n, flags, pitch_range, lsp16_flag, ret; | ||
372 | 8 | WMAVoiceContext *s = ctx->priv_data; | |
373 | |||
374 | 8 | ff_thread_once(&init_static_once, wmavoice_init_static_data); | |
375 | |||
376 | /** | ||
377 | * Extradata layout: | ||
378 | * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), | ||
379 | * - byte 19-22: flags field (annoyingly in LE; see below for known | ||
380 | * values), | ||
381 | * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, | ||
382 | * rest is 0). | ||
383 | */ | ||
384 |
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8 | if (ctx->extradata_size != 46) { |
385 | ✗ | av_log(ctx, AV_LOG_ERROR, | |
386 | "Invalid extradata size %d (should be 46)\n", | ||
387 | ctx->extradata_size); | ||
388 | ✗ | return AVERROR_INVALIDDATA; | |
389 | } | ||
390 |
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8 | if (ctx->block_align <= 0 || ctx->block_align > (1<<22)) { |
391 | ✗ | av_log(ctx, AV_LOG_ERROR, "Invalid block alignment %d.\n", ctx->block_align); | |
392 | ✗ | return AVERROR_INVALIDDATA; | |
393 | } | ||
394 | |||
395 | 8 | flags = AV_RL32(ctx->extradata + 18); | |
396 | 8 | s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); | |
397 | 8 | s->do_apf = flags & 0x1; | |
398 |
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8 | if (s->do_apf) { |
399 |
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16 | if ((ret = ff_rdft_init(&s->rdft, 7, DFT_R2C)) < 0 || |
400 |
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16 | (ret = ff_rdft_init(&s->irdft, 7, IDFT_C2R)) < 0 || |
401 |
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16 | (ret = ff_dct_init (&s->dct, 6, DCT_I)) < 0 || |
402 | 8 | (ret = ff_dct_init (&s->dst, 6, DST_I)) < 0) | |
403 | ✗ | return ret; | |
404 | |||
405 | 8 | ff_sine_window_init(s->cos, 256); | |
406 | 8 | memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); | |
407 |
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2048 | for (n = 0; n < 255; n++) { |
408 | 2040 | s->sin[n] = -s->sin[510 - n]; | |
409 | 2040 | s->cos[510 - n] = s->cos[n]; | |
410 | } | ||
411 | } | ||
412 | 8 | s->denoise_strength = (flags >> 2) & 0xF; | |
413 |
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8 | if (s->denoise_strength >= 12) { |
414 | ✗ | av_log(ctx, AV_LOG_ERROR, | |
415 | "Invalid denoise filter strength %d (max=11)\n", | ||
416 | s->denoise_strength); | ||
417 | ✗ | return AVERROR_INVALIDDATA; | |
418 | } | ||
419 | 8 | s->denoise_tilt_corr = !!(flags & 0x40); | |
420 | 8 | s->dc_level = (flags >> 7) & 0xF; | |
421 | 8 | s->lsp_q_mode = !!(flags & 0x2000); | |
422 | 8 | s->lsp_def_mode = !!(flags & 0x4000); | |
423 | 8 | lsp16_flag = flags & 0x1000; | |
424 |
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8 | if (lsp16_flag) { |
425 | 4 | s->lsps = 16; | |
426 | } else { | ||
427 | 4 | s->lsps = 10; | |
428 | } | ||
429 |
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112 | for (n = 0; n < s->lsps; n++) |
430 | 104 | s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
431 | |||
432 | 8 | init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); | |
433 |
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8 | if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { |
434 | ✗ | av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); | |
435 | ✗ | return AVERROR_INVALIDDATA; | |
436 | } | ||
437 | |||
438 |
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8 | if (ctx->sample_rate >= INT_MAX / (256 * 37)) |
439 | ✗ | return AVERROR_INVALIDDATA; | |
440 | |||
441 | 8 | s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; | |
442 | 8 | s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; | |
443 | 8 | pitch_range = s->max_pitch_val - s->min_pitch_val; | |
444 |
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8 | if (pitch_range <= 0) { |
445 | ✗ | av_log(ctx, AV_LOG_ERROR, "Invalid pitch range; broken extradata?\n"); | |
446 | ✗ | return AVERROR_INVALIDDATA; | |
447 | } | ||
448 | 8 | s->pitch_nbits = av_ceil_log2(pitch_range); | |
449 | 8 | s->last_pitch_val = 40; | |
450 | 8 | s->last_acb_type = ACB_TYPE_NONE; | |
451 | 8 | s->history_nsamples = s->max_pitch_val + 8; | |
452 | |||
453 |
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8 | if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { |
454 | ✗ | int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, | |
455 | ✗ | max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; | |
456 | |||
457 | ✗ | av_log(ctx, AV_LOG_ERROR, | |
458 | "Unsupported samplerate %d (min=%d, max=%d)\n", | ||
459 | ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz | ||
460 | |||
461 | ✗ | return AVERROR(ENOSYS); | |
462 | } | ||
463 | |||
464 | 8 | s->block_conv_table[0] = s->min_pitch_val; | |
465 | 8 | s->block_conv_table[1] = (pitch_range * 25) >> 6; | |
466 | 8 | s->block_conv_table[2] = (pitch_range * 44) >> 6; | |
467 | 8 | s->block_conv_table[3] = s->max_pitch_val - 1; | |
468 | 8 | s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; | |
469 |
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8 | if (s->block_delta_pitch_hrange <= 0) { |
470 | ✗ | av_log(ctx, AV_LOG_ERROR, "Invalid delta pitch hrange; broken extradata?\n"); | |
471 | ✗ | return AVERROR_INVALIDDATA; | |
472 | } | ||
473 | 8 | s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); | |
474 | 8 | s->block_pitch_range = s->block_conv_table[2] + | |
475 | 8 | s->block_conv_table[3] + 1 + | |
476 | 8 | 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); | |
477 | 8 | s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); | |
478 | |||
479 | 8 | av_channel_layout_uninit(&ctx->ch_layout); | |
480 | 8 | ctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO; | |
481 | 8 | ctx->sample_fmt = AV_SAMPLE_FMT_FLT; | |
482 | |||
483 | 8 | return 0; | |
484 | } | ||
485 | |||
486 | /** | ||
487 | * @name Postfilter functions | ||
488 | * Postfilter functions (gain control, wiener denoise filter, DC filter, | ||
489 | * kalman smoothening, plus surrounding code to wrap it) | ||
490 | * @{ | ||
491 | */ | ||
492 | /** | ||
493 | * Adaptive gain control (as used in postfilter). | ||
494 | * | ||
495 | * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except | ||
496 | * that the energy here is calculated using sum(abs(...)), whereas the | ||
497 | * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). | ||
498 | * | ||
499 | * @param out output buffer for filtered samples | ||
500 | * @param in input buffer containing the samples as they are after the | ||
501 | * postfilter steps so far | ||
502 | * @param speech_synth input buffer containing speech synth before postfilter | ||
503 | * @param size input buffer size | ||
504 | * @param alpha exponential filter factor | ||
505 | * @param gain_mem pointer to filter memory (single float) | ||
506 | */ | ||
507 | 6612 | static void adaptive_gain_control(float *out, const float *in, | |
508 | const float *speech_synth, | ||
509 | int size, float alpha, float *gain_mem) | ||
510 | { | ||
511 | int i; | ||
512 | 6612 | float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; | |
513 | 6612 | float mem = *gain_mem; | |
514 | |||
515 |
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535572 | for (i = 0; i < size; i++) { |
516 | 528960 | speech_energy += fabsf(speech_synth[i]); | |
517 | 528960 | postfilter_energy += fabsf(in[i]); | |
518 | } | ||
519 |
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6612 | gain_scale_factor = postfilter_energy == 0.0 ? 0.0 : |
520 | 6612 | (1.0 - alpha) * speech_energy / postfilter_energy; | |
521 | |||
522 |
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535572 | for (i = 0; i < size; i++) { |
523 | 528960 | mem = alpha * mem + gain_scale_factor; | |
524 | 528960 | out[i] = in[i] * mem; | |
525 | } | ||
526 | |||
527 | 6612 | *gain_mem = mem; | |
528 | 6612 | } | |
529 | |||
530 | /** | ||
531 | * Kalman smoothing function. | ||
532 | * | ||
533 | * This function looks back pitch +/- 3 samples back into history to find | ||
534 | * the best fitting curve (that one giving the optimal gain of the two | ||
535 | * signals, i.e. the highest dot product between the two), and then | ||
536 | * uses that signal history to smoothen the output of the speech synthesis | ||
537 | * filter. | ||
538 | * | ||
539 | * @param s WMA Voice decoding context | ||
540 | * @param pitch pitch of the speech signal | ||
541 | * @param in input speech signal | ||
542 | * @param out output pointer for smoothened signal | ||
543 | * @param size input/output buffer size | ||
544 | * | ||
545 | * @returns -1 if no smoothening took place, e.g. because no optimal | ||
546 | * fit could be found, or 0 on success. | ||
547 | */ | ||
548 | 5070 | static int kalman_smoothen(WMAVoiceContext *s, int pitch, | |
549 | const float *in, float *out, int size) | ||
550 | { | ||
551 | int n; | ||
552 | 5070 | float optimal_gain = 0, dot; | |
553 |
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5070 | const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], |
554 | 5070 | *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], | |
555 | 5070 | *best_hist_ptr = NULL; | |
556 | |||
557 | /* find best fitting point in history */ | ||
558 | do { | ||
559 | 35388 | dot = avpriv_scalarproduct_float_c(in, ptr, size); | |
560 |
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35388 | if (dot > optimal_gain) { |
561 | 12328 | optimal_gain = dot; | |
562 | 12328 | best_hist_ptr = ptr; | |
563 | } | ||
564 |
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35388 | } while (--ptr >= end); |
565 | |||
566 |
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5070 | if (optimal_gain <= 0) |
567 | 26 | return -1; | |
568 | 5044 | dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size); | |
569 |
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5044 | if (dot <= 0) // would be 1.0 |
570 | ✗ | return -1; | |
571 | |||
572 |
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5044 | if (optimal_gain <= dot) { |
573 | 4872 | dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 | |
574 | } else | ||
575 | 172 | dot = 0.625; | |
576 | |||
577 | /* actual smoothing */ | ||
578 |
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408564 | for (n = 0; n < size; n++) |
579 | 403520 | out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); | |
580 | |||
581 | 5044 | return 0; | |
582 | } | ||
583 | |||
584 | /** | ||
585 | * Get the tilt factor of a formant filter from its transfer function | ||
586 | * @see #tilt_factor() in amrnbdec.c, which does essentially the same, | ||
587 | * but somehow (??) it does a speech synthesis filter in the | ||
588 | * middle, which is missing here | ||
589 | * | ||
590 | * @param lpcs LPC coefficients | ||
591 | * @param n_lpcs Size of LPC buffer | ||
592 | * @returns the tilt factor | ||
593 | */ | ||
594 | 7098 | static float tilt_factor(const float *lpcs, int n_lpcs) | |
595 | { | ||
596 | float rh0, rh1; | ||
597 | |||
598 | 7098 | rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs); | |
599 | 7098 | rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1); | |
600 | |||
601 | 7098 | return rh1 / rh0; | |
602 | } | ||
603 | |||
604 | /** | ||
605 | * Derive denoise filter coefficients (in real domain) from the LPCs. | ||
606 | */ | ||
607 | 5614 | static void calc_input_response(WMAVoiceContext *s, float *lpcs, | |
608 | int fcb_type, float *coeffs, int remainder) | ||
609 | { | ||
610 | 5614 | float last_coeff, min = 15.0, max = -15.0; | |
611 | float irange, angle_mul, gain_mul, range, sq; | ||
612 | int n, idx; | ||
613 | |||
614 | /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ | ||
615 | 5614 | s->rdft.rdft_calc(&s->rdft, lpcs); | |
616 | #define log_range(var, assign) do { \ | ||
617 | float tmp = log10f(assign); var = tmp; \ | ||
618 | max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ | ||
619 | } while (0) | ||
620 |
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5614 | log_range(last_coeff, lpcs[1] * lpcs[1]); |
621 |
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359296 | for (n = 1; n < 64; n++) |
622 |
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353682 | log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + |
623 | lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); | ||
624 |
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5614 | log_range(lpcs[0], lpcs[0] * lpcs[0]); |
625 | #undef log_range | ||
626 | 5614 | range = max - min; | |
627 | 5614 | lpcs[64] = last_coeff; | |
628 | |||
629 | /* Now, use this spectrum to pick out these frequencies with higher | ||
630 | * (relative) power/energy (which we then take to be "not noise"), | ||
631 | * and set up a table (still in lpc[]) of (relative) gains per frequency. | ||
632 | * These frequencies will be maintained, while others ("noise") will be | ||
633 | * decreased in the filter output. */ | ||
634 | 5614 | irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] | |
635 |
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5614 | gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : |
636 | (5.0 / 14.7)); | ||
637 | 5614 | angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); | |
638 |
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370524 | for (n = 0; n <= 64; n++) { |
639 | float pwr; | ||
640 | |||
641 | 364910 | idx = lrint((max - lpcs[n]) * irange - 1); | |
642 | 364910 | idx = FFMAX(0, idx); | |
643 | 364910 | pwr = wmavoice_denoise_power_table[s->denoise_strength][idx]; | |
644 | 364910 | lpcs[n] = angle_mul * pwr; | |
645 | |||
646 | /* 70.57 =~ 1/log10(1.0331663) */ | ||
647 | 364910 | idx = av_clipf((pwr * gain_mul - 0.0295) * 70.570526123, 0, INT_MAX / 2); | |
648 | |||
649 |
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364910 | if (idx > 127) { // fall back if index falls outside table range |
650 | 8557 | coeffs[n] = wmavoice_energy_table[127] * | |
651 | 8557 | powf(1.0331663, idx - 127); | |
652 | } else | ||
653 | 356353 | coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; | |
654 | } | ||
655 | |||
656 | /* calculate the Hilbert transform of the gains, which we do (since this | ||
657 | * is a sine input) by doing a phase shift (in theory, H(sin())=cos()). | ||
658 | * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the | ||
659 | * "moment" of the LPCs in this filter. */ | ||
660 | 5614 | s->dct.dct_calc(&s->dct, lpcs); | |
661 | 5614 | s->dst.dct_calc(&s->dst, lpcs); | |
662 | |||
663 | /* Split out the coefficient indexes into phase/magnitude pairs */ | ||
664 | 5614 | idx = 255 + av_clip(lpcs[64], -255, 255); | |
665 | 5614 | coeffs[0] = coeffs[0] * s->cos[idx]; | |
666 | 5614 | idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); | |
667 | 5614 | last_coeff = coeffs[64] * s->cos[idx]; | |
668 | 5614 | for (n = 63;; n--) { | |
669 | 179648 | idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); | |
670 | 179648 | coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; | |
671 | 179648 | coeffs[n * 2] = coeffs[n] * s->cos[idx]; | |
672 | |||
673 |
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179648 | if (!--n) break; |
674 | |||
675 | 174034 | idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); | |
676 | 174034 | coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; | |
677 | 174034 | coeffs[n * 2] = coeffs[n] * s->cos[idx]; | |
678 | } | ||
679 | 5614 | coeffs[1] = last_coeff; | |
680 | |||
681 | /* move into real domain */ | ||
682 | 5614 | s->irdft.rdft_calc(&s->irdft, coeffs); | |
683 | |||
684 | /* tilt correction and normalize scale */ | ||
685 | 5614 | memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); | |
686 |
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5614 | if (s->denoise_tilt_corr) { |
687 | 1484 | float tilt_mem = 0; | |
688 | |||
689 | 1484 | coeffs[remainder - 1] = 0; | |
690 | 1484 | ff_tilt_compensation(&tilt_mem, | |
691 | 1484 | -1.8 * tilt_factor(coeffs, remainder - 1), | |
692 | coeffs, remainder); | ||
693 | } | ||
694 | 5614 | sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs, | |
695 | remainder)); | ||
696 |
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269472 | for (n = 0; n < remainder; n++) |
697 | 263858 | coeffs[n] *= sq; | |
698 | 5614 | } | |
699 | |||
700 | /** | ||
701 | * This function applies a Wiener filter on the (noisy) speech signal as | ||
702 | * a means to denoise it. | ||
703 | * | ||
704 | * - take RDFT of LPCs to get the power spectrum of the noise + speech; | ||
705 | * - using this power spectrum, calculate (for each frequency) the Wiener | ||
706 | * filter gain, which depends on the frequency power and desired level | ||
707 | * of noise subtraction (when set too high, this leads to artifacts) | ||
708 | * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse | ||
709 | * of 4-8kHz); | ||
710 | * - by doing a phase shift, calculate the Hilbert transform of this array | ||
711 | * of per-frequency filter-gains to get the filtering coefficients; | ||
712 | * - smoothen/normalize/de-tilt these filter coefficients as desired; | ||
713 | * - take RDFT of noisy sound, apply the coefficients and take its IRDFT | ||
714 | * to get the denoised speech signal; | ||
715 | * - the leftover (i.e. output of the IRDFT on denoised speech data beyond | ||
716 | * the frame boundary) are saved and applied to subsequent frames by an | ||
717 | * overlap-add method (otherwise you get clicking-artifacts). | ||
718 | * | ||
719 | * @param s WMA Voice decoding context | ||
720 | * @param fcb_type Frame (codebook) type | ||
721 | * @param synth_pf input: the noisy speech signal, output: denoised speech | ||
722 | * data; should be 16-byte aligned (for ASM purposes) | ||
723 | * @param size size of the speech data | ||
724 | * @param lpcs LPCs used to synthesize this frame's speech data | ||
725 | */ | ||
726 | 6612 | static void wiener_denoise(WMAVoiceContext *s, int fcb_type, | |
727 | float *synth_pf, int size, | ||
728 | const float *lpcs) | ||
729 | { | ||
730 | int remainder, lim, n; | ||
731 | |||
732 |
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6612 | if (fcb_type != FCB_TYPE_SILENCE) { |
733 | 5614 | float *tilted_lpcs = s->tilted_lpcs_pf, | |
734 | 5614 | *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; | |
735 | |||
736 | 5614 | tilted_lpcs[0] = 1.0; | |
737 | 5614 | memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); | |
738 | 5614 | memset(&tilted_lpcs[s->lsps + 1], 0, | |
739 | 5614 | sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); | |
740 | 5614 | ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), | |
741 | 5614 | tilted_lpcs, s->lsps + 2); | |
742 | |||
743 | /* The IRDFT output (127 samples for 7-bit filter) beyond the frame | ||
744 | * size is applied to the next frame. All input beyond this is zero, | ||
745 | * and thus all output beyond this will go towards zero, hence we can | ||
746 | * limit to min(size-1, 127-size) as a performance consideration. */ | ||
747 |
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5614 | remainder = FFMIN(127 - size, size - 1); |
748 | 5614 | calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); | |
749 | |||
750 | /* apply coefficients (in frequency spectrum domain), i.e. complex | ||
751 | * number multiplication */ | ||
752 | 5614 | memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); | |
753 | 5614 | s->rdft.rdft_calc(&s->rdft, synth_pf); | |
754 | 5614 | s->rdft.rdft_calc(&s->rdft, coeffs); | |
755 | 5614 | synth_pf[0] *= coeffs[0]; | |
756 | 5614 | synth_pf[1] *= coeffs[1]; | |
757 |
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359296 | for (n = 1; n < 64; n++) { |
758 | 353682 | float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; | |
759 | 353682 | synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; | |
760 | 353682 | synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; | |
761 | } | ||
762 | 5614 | s->irdft.rdft_calc(&s->irdft, synth_pf); | |
763 | } | ||
764 | |||
765 | /* merge filter output with the history of previous runs */ | ||
766 |
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6612 | if (s->denoise_filter_cache_size) { |
767 | 5612 | lim = FFMIN(s->denoise_filter_cache_size, size); | |
768 |
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269376 | for (n = 0; n < lim; n++) |
769 | 263764 | synth_pf[n] += s->denoise_filter_cache[n]; | |
770 | 5612 | s->denoise_filter_cache_size -= lim; | |
771 | 5612 | memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], | |
772 | 5612 | sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); | |
773 | } | ||
774 | |||
775 | /* move remainder of filter output into a cache for future runs */ | ||
776 |
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6612 | if (fcb_type != FCB_TYPE_SILENCE) { |
777 | 5614 | lim = FFMIN(remainder, s->denoise_filter_cache_size); | |
778 |
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5614 | for (n = 0; n < lim; n++) |
779 | ✗ | s->denoise_filter_cache[n] += synth_pf[size + n]; | |
780 |
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5614 | if (lim < remainder) { |
781 | 5614 | memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], | |
782 | 5614 | sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); | |
783 | 5614 | s->denoise_filter_cache_size = remainder; | |
784 | } | ||
785 | } | ||
786 | 6612 | } | |
787 | |||
788 | /** | ||
789 | * Averaging projection filter, the postfilter used in WMAVoice. | ||
790 | * | ||
791 | * This uses the following steps: | ||
792 | * - A zero-synthesis filter (generate excitation from synth signal) | ||
793 | * - Kalman smoothing on excitation, based on pitch | ||
794 | * - Re-synthesized smoothened output | ||
795 | * - Iterative Wiener denoise filter | ||
796 | * - Adaptive gain filter | ||
797 | * - DC filter | ||
798 | * | ||
799 | * @param s WMAVoice decoding context | ||
800 | * @param synth Speech synthesis output (before postfilter) | ||
801 | * @param samples Output buffer for filtered samples | ||
802 | * @param size Buffer size of synth & samples | ||
803 | * @param lpcs Generated LPCs used for speech synthesis | ||
804 | * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned) | ||
805 | * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) | ||
806 | * @param pitch Pitch of the input signal | ||
807 | */ | ||
808 | 6612 | static void postfilter(WMAVoiceContext *s, const float *synth, | |
809 | float *samples, int size, | ||
810 | const float *lpcs, float *zero_exc_pf, | ||
811 | int fcb_type, int pitch) | ||
812 | { | ||
813 | float synth_filter_in_buf[MAX_FRAMESIZE / 2], | ||
814 | 6612 | *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], | |
815 | 6612 | *synth_filter_in = zero_exc_pf; | |
816 | |||
817 |
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6612 | av_assert0(size <= MAX_FRAMESIZE / 2); |
818 | |||
819 | /* generate excitation from input signal */ | ||
820 | 6612 | ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); | |
821 | |||
822 |
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11682 | if (fcb_type >= FCB_TYPE_AW_PULSES && |
823 | 5070 | !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) | |
824 | 5044 | synth_filter_in = synth_filter_in_buf; | |
825 | |||
826 | /* re-synthesize speech after smoothening, and keep history */ | ||
827 | 6612 | ff_celp_lp_synthesis_filterf(synth_pf, lpcs, | |
828 | synth_filter_in, size, s->lsps); | ||
829 | 6612 | memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], | |
830 | 6612 | sizeof(synth_pf[0]) * s->lsps); | |
831 | |||
832 | 6612 | wiener_denoise(s, fcb_type, synth_pf, size, lpcs); | |
833 | |||
834 | 6612 | adaptive_gain_control(samples, synth_pf, synth, size, 0.99, | |
835 | &s->postfilter_agc); | ||
836 | |||
837 |
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6612 | if (s->dc_level > 8) { |
838 | /* remove ultra-low frequency DC noise / highpass filter; | ||
839 | * coefficients are identical to those used in SIPR decoding, | ||
840 | * and very closely resemble those used in AMR-NB decoding. */ | ||
841 | ✗ | ff_acelp_apply_order_2_transfer_function(samples, samples, | |
842 | ✗ | (const float[2]) { -1.99997, 1.0 }, | |
843 | ✗ | (const float[2]) { -1.9330735188, 0.93589198496 }, | |
844 | ✗ | 0.93980580475, s->dcf_mem, size); | |
845 | } | ||
846 | 6612 | } | |
847 | /** | ||
848 | * @} | ||
849 | */ | ||
850 | |||
851 | /** | ||
852 | * Dequantize LSPs | ||
853 | * @param lsps output pointer to the array that will hold the LSPs | ||
854 | * @param num number of LSPs to be dequantized | ||
855 | * @param values quantized values, contains n_stages values | ||
856 | * @param sizes range (i.e. max value) of each quantized value | ||
857 | * @param n_stages number of dequantization runs | ||
858 | * @param table dequantization table to be used | ||
859 | * @param mul_q LSF multiplier | ||
860 | * @param base_q base (lowest) LSF values | ||
861 | */ | ||
862 | 4404 | static void dequant_lsps(double *lsps, int num, | |
863 | const uint16_t *values, | ||
864 | const uint16_t *sizes, | ||
865 | int n_stages, const uint8_t *table, | ||
866 | const double *mul_q, | ||
867 | const double *base_q) | ||
868 | { | ||
869 | int n, m; | ||
870 | |||
871 | 4404 | memset(lsps, 0, num * sizeof(*lsps)); | |
872 |
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12668 | for (n = 0; n < n_stages; n++) { |
873 | 8264 | const uint8_t *t_off = &table[values[n] * num]; | |
874 | 8264 | double base = base_q[n], mul = mul_q[n]; | |
875 | |||
876 |
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95364 | for (m = 0; m < num; m++) |
877 | 87100 | lsps[m] += base + mul * t_off[m]; | |
878 | |||
879 | 8264 | table += sizes[n] * num; | |
880 | } | ||
881 | 4404 | } | |
882 | |||
883 | /** | ||
884 | * @name LSP dequantization routines | ||
885 | * LSP dequantization routines, for 10/16LSPs and independent/residual coding. | ||
886 | * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; | ||
887 | * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. | ||
888 | * @{ | ||
889 | */ | ||
890 | /** | ||
891 | * Parse 10 independently-coded LSPs. | ||
892 | */ | ||
893 | 552 | static void dequant_lsp10i(GetBitContext *gb, double *lsps) | |
894 | { | ||
895 | static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; | ||
896 | static const double mul_lsf[4] = { | ||
897 | 5.2187144800e-3, 1.4626986422e-3, | ||
898 | 9.6179549166e-4, 1.1325736225e-3 | ||
899 | }; | ||
900 | static const double base_lsf[4] = { | ||
901 | M_PI * -2.15522e-1, M_PI * -6.1646e-2, | ||
902 | M_PI * -3.3486e-2, M_PI * -5.7408e-2 | ||
903 | }; | ||
904 | uint16_t v[4]; | ||
905 | |||
906 | 552 | v[0] = get_bits(gb, 8); | |
907 | 552 | v[1] = get_bits(gb, 6); | |
908 | 552 | v[2] = get_bits(gb, 5); | |
909 | 552 | v[3] = get_bits(gb, 5); | |
910 | |||
911 | 552 | dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, | |
912 | mul_lsf, base_lsf); | ||
913 | 552 | } | |
914 | |||
915 | /** | ||
916 | * Parse 10 independently-coded LSPs, and then derive the tables to | ||
917 | * generate LSPs for the other frames from them (residual coding). | ||
918 | */ | ||
919 | 552 | static void dequant_lsp10r(GetBitContext *gb, | |
920 | double *i_lsps, const double *old, | ||
921 | double *a1, double *a2, int q_mode) | ||
922 | { | ||
923 | static const uint16_t vec_sizes[3] = { 128, 64, 64 }; | ||
924 | static const double mul_lsf[3] = { | ||
925 | 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 | ||
926 | }; | ||
927 | static const double base_lsf[3] = { | ||
928 | M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 | ||
929 | }; | ||
930 | 552 | const float (*ipol_tab)[2][10] = q_mode ? | |
931 |
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552 | wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; |
932 | uint16_t interpol, v[3]; | ||
933 | int n; | ||
934 | |||
935 | 552 | dequant_lsp10i(gb, i_lsps); | |
936 | |||
937 | 552 | interpol = get_bits(gb, 5); | |
938 | 552 | v[0] = get_bits(gb, 7); | |
939 | 552 | v[1] = get_bits(gb, 6); | |
940 | 552 | v[2] = get_bits(gb, 6); | |
941 | |||
942 |
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6072 | for (n = 0; n < 10; n++) { |
943 | 5520 | double delta = old[n] - i_lsps[n]; | |
944 | 5520 | a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
945 | 5520 | a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
946 | } | ||
947 | |||
948 | 552 | dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, | |
949 | mul_lsf, base_lsf); | ||
950 | 552 | } | |
951 | |||
952 | /** | ||
953 | * Parse 16 independently-coded LSPs. | ||
954 | */ | ||
955 | 550 | static void dequant_lsp16i(GetBitContext *gb, double *lsps) | |
956 | { | ||
957 | static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; | ||
958 | static const double mul_lsf[5] = { | ||
959 | 3.3439586280e-3, 6.9908173703e-4, | ||
960 | 3.3216608306e-3, 1.0334960326e-3, | ||
961 | 3.1899104283e-3 | ||
962 | }; | ||
963 | static const double base_lsf[5] = { | ||
964 | M_PI * -1.27576e-1, M_PI * -2.4292e-2, | ||
965 | M_PI * -1.28094e-1, M_PI * -3.2128e-2, | ||
966 | M_PI * -1.29816e-1 | ||
967 | }; | ||
968 | uint16_t v[5]; | ||
969 | |||
970 | 550 | v[0] = get_bits(gb, 8); | |
971 | 550 | v[1] = get_bits(gb, 6); | |
972 | 550 | v[2] = get_bits(gb, 7); | |
973 | 550 | v[3] = get_bits(gb, 6); | |
974 | 550 | v[4] = get_bits(gb, 7); | |
975 | |||
976 | 550 | dequant_lsps( lsps, 5, v, vec_sizes, 2, | |
977 | wmavoice_dq_lsp16i1, mul_lsf, base_lsf); | ||
978 | 550 | dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, | |
979 | wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); | ||
980 | 550 | dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, | |
981 | wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); | ||
982 | 550 | } | |
983 | |||
984 | /** | ||
985 | * Parse 16 independently-coded LSPs, and then derive the tables to | ||
986 | * generate LSPs for the other frames from them (residual coding). | ||
987 | */ | ||
988 | 550 | static void dequant_lsp16r(GetBitContext *gb, | |
989 | double *i_lsps, const double *old, | ||
990 | double *a1, double *a2, int q_mode) | ||
991 | { | ||
992 | static const uint16_t vec_sizes[3] = { 128, 128, 128 }; | ||
993 | static const double mul_lsf[3] = { | ||
994 | 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 | ||
995 | }; | ||
996 | static const double base_lsf[3] = { | ||
997 | M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 | ||
998 | }; | ||
999 | 550 | const float (*ipol_tab)[2][16] = q_mode ? | |
1000 |
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550 | wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; |
1001 | uint16_t interpol, v[3]; | ||
1002 | int n; | ||
1003 | |||
1004 | 550 | dequant_lsp16i(gb, i_lsps); | |
1005 | |||
1006 | 550 | interpol = get_bits(gb, 5); | |
1007 | 550 | v[0] = get_bits(gb, 7); | |
1008 | 550 | v[1] = get_bits(gb, 7); | |
1009 | 550 | v[2] = get_bits(gb, 7); | |
1010 | |||
1011 |
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9350 | for (n = 0; n < 16; n++) { |
1012 | 8800 | double delta = old[n] - i_lsps[n]; | |
1013 | 8800 | a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
1014 | 8800 | a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
1015 | } | ||
1016 | |||
1017 | 550 | dequant_lsps( a2, 10, v, vec_sizes, 1, | |
1018 | wmavoice_dq_lsp16r1, mul_lsf, base_lsf); | ||
1019 | 550 | dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, | |
1020 | wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); | ||
1021 | 550 | dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, | |
1022 | wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); | ||
1023 | 550 | } | |
1024 | |||
1025 | /** | ||
1026 | * @} | ||
1027 | * @name Pitch-adaptive window coding functions | ||
1028 | * The next few functions are for pitch-adaptive window coding. | ||
1029 | * @{ | ||
1030 | */ | ||
1031 | /** | ||
1032 | * Parse the offset of the first pitch-adaptive window pulses, and | ||
1033 | * the distribution of pulses between the two blocks in this frame. | ||
1034 | * @param s WMA Voice decoding context private data | ||
1035 | * @param gb bit I/O context | ||
1036 | * @param pitch pitch for each block in this frame | ||
1037 | */ | ||
1038 | 341 | static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, | |
1039 | const int *pitch) | ||
1040 | { | ||
1041 | static const int16_t start_offset[94] = { | ||
1042 | -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, | ||
1043 | 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, | ||
1044 | 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, | ||
1045 | 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, | ||
1046 | 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, | ||
1047 | 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, | ||
1048 | 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, | ||
1049 | 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 | ||
1050 | }; | ||
1051 | int bits, offset; | ||
1052 | |||
1053 | /* position of pulse */ | ||
1054 | 341 | s->aw_idx_is_ext = 0; | |
1055 |
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341 | if ((bits = get_bits(gb, 6)) >= 54) { |
1056 | 10 | s->aw_idx_is_ext = 1; | |
1057 | 10 | bits += (bits - 54) * 3 + get_bits(gb, 2); | |
1058 | } | ||
1059 | |||
1060 | /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count | ||
1061 | * the distribution of the pulses in each block contained in this frame. */ | ||
1062 |
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341 | s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; |
1063 |
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391 | for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; |
1064 | 341 | s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; | |
1065 | 341 | s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; | |
1066 | 341 | offset += s->aw_n_pulses[0] * pitch[0]; | |
1067 | 341 | s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; | |
1068 | 341 | s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; | |
1069 | |||
1070 | /* if continuing from a position before the block, reset position to | ||
1071 | * start of block (when corrected for the range over which it can be | ||
1072 | * spread in aw_pulse_set1()). */ | ||
1073 |
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341 | if (start_offset[bits] < MAX_FRAMESIZE / 2) { |
1074 |
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387 | while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) |
1075 | 56 | s->aw_first_pulse_off[1] -= pitch[1]; | |
1076 |
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331 | if (start_offset[bits] < 0) |
1077 |
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100 | while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) |
1078 | 50 | s->aw_first_pulse_off[0] -= pitch[0]; | |
1079 | } | ||
1080 | 341 | } | |
1081 | |||
1082 | /** | ||
1083 | * Apply second set of pitch-adaptive window pulses. | ||
1084 | * @param s WMA Voice decoding context private data | ||
1085 | * @param gb bit I/O context | ||
1086 | * @param block_idx block index in frame [0, 1] | ||
1087 | * @param fcb structure containing fixed codebook vector info | ||
1088 | * @return -1 on error, 0 otherwise | ||
1089 | */ | ||
1090 | 682 | static int aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, | |
1091 | int block_idx, AMRFixed *fcb) | ||
1092 | { | ||
1093 | uint16_t use_mask_mem[9]; // only 5 are used, rest is padding | ||
1094 | 682 | uint16_t *use_mask = use_mask_mem + 2; | |
1095 | /* in this function, idx is the index in the 80-bit (+ padding) use_mask | ||
1096 | * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits | ||
1097 | * of idx are the position of the bit within a particular item in the | ||
1098 | * array (0 being the most significant bit, and 15 being the least | ||
1099 | * significant bit), and the remainder (>> 4) is the index in the | ||
1100 | * use_mask[]-array. This is faster and uses less memory than using a | ||
1101 | * 80-byte/80-int array. */ | ||
1102 | 682 | int pulse_off = s->aw_first_pulse_off[block_idx], | |
1103 | 682 | pulse_start, n, idx, range, aidx, start_off = 0; | |
1104 | |||
1105 | /* set offset of first pulse to within this block */ | ||
1106 |
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682 | if (s->aw_n_pulses[block_idx] > 0) |
1107 |
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657 | while (pulse_off + s->aw_pulse_range < 1) |
1108 | ✗ | pulse_off += fcb->pitch_lag; | |
1109 | |||
1110 | /* find range per pulse */ | ||
1111 |
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682 | if (s->aw_n_pulses[0] > 0) { |
1112 |
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646 | if (block_idx == 0) { |
1113 | 323 | range = 32; | |
1114 | } else /* block_idx = 1 */ { | ||
1115 | 323 | range = 8; | |
1116 |
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323 | if (s->aw_n_pulses[block_idx] > 0) |
1117 | 316 | pulse_off = s->aw_next_pulse_off_cache; | |
1118 | } | ||
1119 | } else | ||
1120 | 36 | range = 16; | |
1121 |
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682 | pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; |
1122 | |||
1123 | /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, | ||
1124 | * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus | ||
1125 | * we exclude that range from being pulsed again in this function. */ | ||
1126 | 682 | memset(&use_mask[-2], 0, 2 * sizeof(use_mask[0])); | |
1127 | 682 | memset( use_mask, -1, 5 * sizeof(use_mask[0])); | |
1128 | 682 | memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); | |
1129 |
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682 | if (s->aw_n_pulses[block_idx] > 0) |
1130 |
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1568 | for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { |
1131 | 911 | int excl_range = s->aw_pulse_range; // always 16 or 24 | |
1132 | 911 | uint16_t *use_mask_ptr = &use_mask[idx >> 4]; | |
1133 | 911 | int first_sh = 16 - (idx & 15); | |
1134 | 911 | *use_mask_ptr++ &= 0xFFFFu << first_sh; | |
1135 | 911 | excl_range -= first_sh; | |
1136 |
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911 | if (excl_range >= 16) { |
1137 | 468 | *use_mask_ptr++ = 0; | |
1138 | 468 | *use_mask_ptr &= 0xFFFF >> (excl_range - 16); | |
1139 | } else | ||
1140 | 443 | *use_mask_ptr &= 0xFFFF >> excl_range; | |
1141 | } | ||
1142 | |||
1143 | /* find the 'aidx'th offset that is not excluded */ | ||
1144 |
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682 | aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); |
1145 |
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16825 | for (n = 0; n <= aidx; pulse_start++) { |
1146 |
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18458 | for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; |
1147 |
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16143 | if (idx >= MAX_FRAMESIZE / 2) { // find from zero |
1148 |
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538 | if (use_mask[0]) idx = 0x0F; |
1149 |
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123 | else if (use_mask[1]) idx = 0x1F; |
1150 |
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18 | else if (use_mask[2]) idx = 0x2F; |
1151 | ✗ | else if (use_mask[3]) idx = 0x3F; | |
1152 | ✗ | else if (use_mask[4]) idx = 0x4F; | |
1153 | ✗ | else return -1; | |
1154 | 538 | idx -= av_log2_16bit(use_mask[idx >> 4]); | |
1155 | } | ||
1156 |
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16143 | if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { |
1157 | 7465 | use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); | |
1158 | 7465 | n++; | |
1159 | 7465 | start_off = idx; | |
1160 | } | ||
1161 | } | ||
1162 | |||
1163 | 682 | fcb->x[fcb->n] = start_off; | |
1164 |
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|
682 | fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; |
1165 | 682 | fcb->n++; | |
1166 | |||
1167 | /* set offset for next block, relative to start of that block */ | ||
1168 | 682 | n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; | |
1169 |
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682 | s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; |
1170 | 682 | return 0; | |
1171 | } | ||
1172 | |||
1173 | /** | ||
1174 | * Apply first set of pitch-adaptive window pulses. | ||
1175 | * @param s WMA Voice decoding context private data | ||
1176 | * @param gb bit I/O context | ||
1177 | * @param block_idx block index in frame [0, 1] | ||
1178 | * @param fcb storage location for fixed codebook pulse info | ||
1179 | */ | ||
1180 | 682 | static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, | |
1181 | int block_idx, AMRFixed *fcb) | ||
1182 | { | ||
1183 |
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682 | int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); |
1184 | float v; | ||
1185 | |||
1186 |
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682 | if (s->aw_n_pulses[block_idx] > 0) { |
1187 | int n, v_mask, i_mask, sh, n_pulses; | ||
1188 | |||
1189 |
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657 | if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each |
1190 | 652 | n_pulses = 3; | |
1191 | 652 | v_mask = 8; | |
1192 | 652 | i_mask = 7; | |
1193 | 652 | sh = 4; | |
1194 | } else { // 4 pulses, 1:sign + 2:index each | ||
1195 | 5 | n_pulses = 4; | |
1196 | 5 | v_mask = 4; | |
1197 | 5 | i_mask = 3; | |
1198 | 5 | sh = 3; | |
1199 | } | ||
1200 | |||
1201 |
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2633 | for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { |
1202 |
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1976 | fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; |
1203 | 1976 | fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + | |
1204 | 1976 | s->aw_first_pulse_off[block_idx]; | |
1205 |
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2217 | while (fcb->x[fcb->n] < 0) |
1206 | 241 | fcb->x[fcb->n] += fcb->pitch_lag; | |
1207 |
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1976 | if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) |
1208 | 1959 | fcb->n++; | |
1209 | } | ||
1210 | } else { | ||
1211 | 25 | int num2 = (val & 0x1FF) >> 1, delta, idx; | |
1212 | |||
1213 |
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25 | if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } |
1214 |
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21 | else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } |
1215 |
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15 | else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } |
1216 | 5 | else { delta = 7; idx = num2 + 1 - 3 * 75; } | |
1217 |
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25 | v = (val & 0x200) ? -1.0 : 1.0; |
1218 | |||
1219 | 25 | fcb->no_repeat_mask |= 3 << fcb->n; | |
1220 | 25 | fcb->x[fcb->n] = idx - delta; | |
1221 | 25 | fcb->y[fcb->n] = v; | |
1222 | 25 | fcb->x[fcb->n + 1] = idx; | |
1223 |
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25 | fcb->y[fcb->n + 1] = (val & 1) ? -v : v; |
1224 | 25 | fcb->n += 2; | |
1225 | } | ||
1226 | 682 | } | |
1227 | |||
1228 | /** | ||
1229 | * @} | ||
1230 | * | ||
1231 | * Generate a random number from frame_cntr and block_idx, which will live | ||
1232 | * in the range [0, 1000 - block_size] (so it can be used as an index in a | ||
1233 | * table of size 1000 of which you want to read block_size entries). | ||
1234 | * | ||
1235 | * @param frame_cntr current frame number | ||
1236 | * @param block_num current block index | ||
1237 | * @param block_size amount of entries we want to read from a table | ||
1238 | * that has 1000 entries | ||
1239 | * @return a (non-)random number in the [0, 1000 - block_size] range. | ||
1240 | */ | ||
1241 | 499 | static int pRNG(int frame_cntr, int block_num, int block_size) | |
1242 | { | ||
1243 | /* array to simplify the calculation of z: | ||
1244 | * y = (x % 9) * 5 + 6; | ||
1245 | * z = (49995 * x) / y; | ||
1246 | * Since y only has 9 values, we can remove the division by using a | ||
1247 | * LUT and using FASTDIV-style divisions. For each of the 9 values | ||
1248 | * of y, we can rewrite z as: | ||
1249 | * z = x * (49995 / y) + x * ((49995 % y) / y) | ||
1250 | * In this table, each col represents one possible value of y, the | ||
1251 | * first number is 49995 / y, and the second is the FASTDIV variant | ||
1252 | * of 49995 % y / y. */ | ||
1253 | static const unsigned int div_tbl[9][2] = { | ||
1254 | { 8332, 3 * 715827883U }, // y = 6 | ||
1255 | { 4545, 0 * 390451573U }, // y = 11 | ||
1256 | { 3124, 11 * 268435456U }, // y = 16 | ||
1257 | { 2380, 15 * 204522253U }, // y = 21 | ||
1258 | { 1922, 23 * 165191050U }, // y = 26 | ||
1259 | { 1612, 23 * 138547333U }, // y = 31 | ||
1260 | { 1388, 27 * 119304648U }, // y = 36 | ||
1261 | { 1219, 16 * 104755300U }, // y = 41 | ||
1262 | { 1086, 39 * 93368855U } // y = 46 | ||
1263 | }; | ||
1264 | 499 | unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; | |
1265 |
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499 | if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, |
1266 | // so this is effectively a modulo (%) | ||
1267 | 499 | y = x - 9 * MULH(477218589, x); // x % 9 | |
1268 | 499 | z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); | |
1269 | // z = x * 49995 / (y * 5 + 6) | ||
1270 | 499 | return z % (1000 - block_size); | |
1271 | } | ||
1272 | |||
1273 | /** | ||
1274 | * Parse hardcoded signal for a single block. | ||
1275 | * @note see #synth_block(). | ||
1276 | */ | ||
1277 | 1043 | static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, | |
1278 | int block_idx, int size, | ||
1279 | const struct frame_type_desc *frame_desc, | ||
1280 | float *excitation) | ||
1281 | { | ||
1282 | float gain; | ||
1283 | int n, r_idx; | ||
1284 | |||
1285 |
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1043 | av_assert0(size <= MAX_FRAMESIZE); |
1286 | |||
1287 | /* Set the offset from which we start reading wmavoice_std_codebook */ | ||
1288 |
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1043 | if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { |
1289 | 499 | r_idx = pRNG(s->frame_cntr, block_idx, size); | |
1290 | 499 | gain = s->silence_gain; | |
1291 | } else /* FCB_TYPE_HARDCODED */ { | ||
1292 | 544 | r_idx = get_bits(gb, 8); | |
1293 | 544 | gain = wmavoice_gain_universal[get_bits(gb, 6)]; | |
1294 | } | ||
1295 | |||
1296 | /* Clear gain prediction parameters */ | ||
1297 | 1043 | memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); | |
1298 | |||
1299 | /* Apply gain to hardcoded codebook and use that as excitation signal */ | ||
1300 |
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124403 | for (n = 0; n < size; n++) |
1301 | 123360 | excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; | |
1302 | 1043 | } | |
1303 | |||
1304 | /** | ||
1305 | * Parse FCB/ACB signal for a single block. | ||
1306 | * @note see #synth_block(). | ||
1307 | */ | ||
1308 | 9740 | static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, | |
1309 | int block_idx, int size, | ||
1310 | int block_pitch_sh2, | ||
1311 | const struct frame_type_desc *frame_desc, | ||
1312 | float *excitation) | ||
1313 | { | ||
1314 | static const float gain_coeff[6] = { | ||
1315 | 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 | ||
1316 | }; | ||
1317 | float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; | ||
1318 | int n, idx, gain_weight; | ||
1319 | AMRFixed fcb; | ||
1320 | |||
1321 |
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9740 | av_assert0(size <= MAX_FRAMESIZE / 2); |
1322 | 9740 | memset(pulses, 0, sizeof(*pulses) * size); | |
1323 | |||
1324 | 9740 | fcb.pitch_lag = block_pitch_sh2 >> 2; | |
1325 | 9740 | fcb.pitch_fac = 1.0; | |
1326 | 9740 | fcb.no_repeat_mask = 0; | |
1327 | 9740 | fcb.n = 0; | |
1328 | |||
1329 | /* For the other frame types, this is where we apply the innovation | ||
1330 | * (fixed) codebook pulses of the speech signal. */ | ||
1331 |
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9740 | if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { |
1332 | 682 | aw_pulse_set1(s, gb, block_idx, &fcb); | |
1333 |
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682 | if (aw_pulse_set2(s, gb, block_idx, &fcb)) { |
1334 | /* Conceal the block with silence and return. | ||
1335 | * Skip the correct amount of bits to read the next | ||
1336 | * block from the correct offset. */ | ||
1337 | ✗ | int r_idx = pRNG(s->frame_cntr, block_idx, size); | |
1338 | |||
1339 | ✗ | for (n = 0; n < size; n++) | |
1340 | ✗ | excitation[n] = | |
1341 | ✗ | wmavoice_std_codebook[r_idx + n] * s->silence_gain; | |
1342 | ✗ | skip_bits(gb, 7 + 1); | |
1343 | ✗ | return; | |
1344 | } | ||
1345 | } else /* FCB_TYPE_EXC_PULSES */ { | ||
1346 | 9058 | int offset_nbits = 5 - frame_desc->log_n_blocks; | |
1347 | |||
1348 | 9058 | fcb.no_repeat_mask = -1; | |
1349 | /* similar to ff_decode_10_pulses_35bits(), but with single pulses | ||
1350 | * (instead of double) for a subset of pulses */ | ||
1351 |
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54348 | for (n = 0; n < 5; n++) { |
1352 | float sign; | ||
1353 | int pos1, pos2; | ||
1354 | |||
1355 |
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45290 | sign = get_bits1(gb) ? 1.0 : -1.0; |
1356 | 45290 | pos1 = get_bits(gb, offset_nbits); | |
1357 | 45290 | fcb.x[fcb.n] = n + 5 * pos1; | |
1358 | 45290 | fcb.y[fcb.n++] = sign; | |
1359 |
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45290 | if (n < frame_desc->dbl_pulses) { |
1360 | 36270 | pos2 = get_bits(gb, offset_nbits); | |
1361 | 36270 | fcb.x[fcb.n] = n + 5 * pos2; | |
1362 |
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36270 | fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; |
1363 | } | ||
1364 | } | ||
1365 | } | ||
1366 | 9740 | ff_set_fixed_vector(pulses, &fcb, 1.0, size); | |
1367 | |||
1368 | /* Calculate gain for adaptive & fixed codebook signal. | ||
1369 | * see ff_amr_set_fixed_gain(). */ | ||
1370 | 9740 | idx = get_bits(gb, 7); | |
1371 | 9740 | fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err, | |
1372 | 9740 | gain_coeff, 6) - | |
1373 | 9740 | 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); | |
1374 | 9740 | acb_gain = wmavoice_gain_codebook_acb[idx]; | |
1375 | 9740 | pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], | |
1376 | -2.9957322736 /* log(0.05) */, | ||
1377 | 1.6094379124 /* log(5.0) */); | ||
1378 | |||
1379 | 9740 | gain_weight = 8 >> frame_desc->log_n_blocks; | |
1380 | 9740 | memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, | |
1381 | 9740 | sizeof(*s->gain_pred_err) * (6 - gain_weight)); | |
1382 |
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30020 | for (n = 0; n < gain_weight; n++) |
1383 | 20280 | s->gain_pred_err[n] = pred_err; | |
1384 | |||
1385 | /* Calculation of adaptive codebook */ | ||
1386 |
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9740 | if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { |
1387 | int len; | ||
1388 |
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19152 | for (n = 0; n < size; n += len) { |
1389 | int next_idx_sh16; | ||
1390 | 17876 | int abs_idx = block_idx * size + n; | |
1391 | 17876 | int pitch_sh16 = (s->last_pitch_val << 16) + | |
1392 | 17876 | s->pitch_diff_sh16 * abs_idx; | |
1393 | 17876 | int pitch = (pitch_sh16 + 0x6FFF) >> 16; | |
1394 | 17876 | int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; | |
1395 | 17876 | idx = idx_sh16 >> 16; | |
1396 |
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17876 | if (s->pitch_diff_sh16) { |
1397 |
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17442 | if (s->pitch_diff_sh16 > 0) { |
1398 | 10526 | next_idx_sh16 = (idx_sh16) &~ 0xFFFF; | |
1399 | } else | ||
1400 | 6916 | next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; | |
1401 | 17442 | len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, | |
1402 | 1, size - n); | ||
1403 | } else | ||
1404 | 434 | len = size; | |
1405 | |||
1406 | 17876 | ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], | |
1407 | wmavoice_ipol1_coeffs, 17, | ||
1408 | idx, 9, len); | ||
1409 | } | ||
1410 | } else /* ACB_TYPE_HAMMING */ { | ||
1411 | 8464 | int block_pitch = block_pitch_sh2 >> 2; | |
1412 | 8464 | idx = block_pitch_sh2 & 3; | |
1413 |
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8464 | if (idx) { |
1414 | 3652 | ff_acelp_interpolatef(excitation, &excitation[-block_pitch], | |
1415 | wmavoice_ipol2_coeffs, 4, | ||
1416 | idx, 8, size); | ||
1417 | } else | ||
1418 | 4812 | av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, | |
1419 | sizeof(float) * size); | ||
1420 | } | ||
1421 | |||
1422 | /* Interpolate ACB/FCB and use as excitation signal */ | ||
1423 | 9740 | ff_weighted_vector_sumf(excitation, excitation, pulses, | |
1424 | acb_gain, fcb_gain, size); | ||
1425 | } | ||
1426 | |||
1427 | /** | ||
1428 | * Parse data in a single block. | ||
1429 | * | ||
1430 | * @param s WMA Voice decoding context private data | ||
1431 | * @param gb bit I/O context | ||
1432 | * @param block_idx index of the to-be-read block | ||
1433 | * @param size amount of samples to be read in this block | ||
1434 | * @param block_pitch_sh2 pitch for this block << 2 | ||
1435 | * @param lsps LSPs for (the end of) this frame | ||
1436 | * @param prev_lsps LSPs for the last frame | ||
1437 | * @param frame_desc frame type descriptor | ||
1438 | * @param excitation target memory for the ACB+FCB interpolated signal | ||
1439 | * @param synth target memory for the speech synthesis filter output | ||
1440 | * @return 0 on success, <0 on error. | ||
1441 | */ | ||
1442 | 10783 | static void synth_block(WMAVoiceContext *s, GetBitContext *gb, | |
1443 | int block_idx, int size, | ||
1444 | int block_pitch_sh2, | ||
1445 | const double *lsps, const double *prev_lsps, | ||
1446 | const struct frame_type_desc *frame_desc, | ||
1447 | float *excitation, float *synth) | ||
1448 | { | ||
1449 | double i_lsps[MAX_LSPS]; | ||
1450 | float lpcs[MAX_LSPS]; | ||
1451 | float fac; | ||
1452 | int n; | ||
1453 | |||
1454 |
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10783 | if (frame_desc->acb_type == ACB_TYPE_NONE) |
1455 | 1043 | synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); | |
1456 | else | ||
1457 | 9740 | synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, | |
1458 | frame_desc, excitation); | ||
1459 | |||
1460 | /* convert interpolated LSPs to LPCs */ | ||
1461 | 10783 | fac = (block_idx + 0.5) / frame_desc->n_blocks; | |
1462 |
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151559 | for (n = 0; n < s->lsps; n++) // LSF -> LSP |
1463 | 140776 | i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); | |
1464 | 10783 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1465 | |||
1466 | /* Speech synthesis */ | ||
1467 | 10783 | ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); | |
1468 | 10783 | } | |
1469 | |||
1470 | /** | ||
1471 | * Synthesize output samples for a single frame. | ||
1472 | * | ||
1473 | * @param ctx WMA Voice decoder context | ||
1474 | * @param gb bit I/O context (s->gb or one for cross-packet superframes) | ||
1475 | * @param frame_idx Frame number within superframe [0-2] | ||
1476 | * @param samples pointer to output sample buffer, has space for at least 160 | ||
1477 | * samples | ||
1478 | * @param lsps LSP array | ||
1479 | * @param prev_lsps array of previous frame's LSPs | ||
1480 | * @param excitation target buffer for excitation signal | ||
1481 | * @param synth target buffer for synthesized speech data | ||
1482 | * @return 0 on success, <0 on error. | ||
1483 | */ | ||
1484 | 3306 | static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, | |
1485 | float *samples, | ||
1486 | const double *lsps, const double *prev_lsps, | ||
1487 | float *excitation, float *synth) | ||
1488 | { | ||
1489 | 3306 | WMAVoiceContext *s = ctx->priv_data; | |
1490 | 3306 | int n, n_blocks_x2, log_n_blocks_x2, av_uninit(cur_pitch_val); | |
1491 | 3306 | int pitch[MAX_BLOCKS], av_uninit(last_block_pitch); | |
1492 | |||
1493 | /* Parse frame type ("frame header"), see frame_descs */ | ||
1494 | 3306 | int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], block_nsamples; | |
1495 | |||
1496 |
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3306 | if (bd_idx < 0) { |
1497 | ✗ | av_log(ctx, AV_LOG_ERROR, | |
1498 | "Invalid frame type VLC code, skipping\n"); | ||
1499 | ✗ | return AVERROR_INVALIDDATA; | |
1500 | } | ||
1501 | |||
1502 | 3306 | block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; | |
1503 | |||
1504 | /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ | ||
1505 |
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3306 | if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { |
1506 | /* Pitch is provided per frame, which is interpreted as the pitch of | ||
1507 | * the last sample of the last block of this frame. We can interpolate | ||
1508 | * the pitch of other blocks (and even pitch-per-sample) by gradually | ||
1509 | * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ | ||
1510 | 560 | n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; | |
1511 | 560 | log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; | |
1512 | 560 | cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); | |
1513 |
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560 | cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); |
1514 |
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560 | if (s->last_acb_type == ACB_TYPE_NONE || |
1515 | 524 | 20 * abs(cur_pitch_val - s->last_pitch_val) > | |
1516 |
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524 | (cur_pitch_val + s->last_pitch_val)) |
1517 | 138 | s->last_pitch_val = cur_pitch_val; | |
1518 | |||
1519 | /* pitch per block */ | ||
1520 |
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|
1836 | for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
1521 | 1276 | int fac = n * 2 + 1; | |
1522 | |||
1523 | 1276 | pitch[n] = (MUL16(fac, cur_pitch_val) + | |
1524 | 1276 | MUL16((n_blocks_x2 - fac), s->last_pitch_val) + | |
1525 | 1276 | frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; | |
1526 | } | ||
1527 | |||
1528 | /* "pitch-diff-per-sample" for calculation of pitch per sample */ | ||
1529 | 560 | s->pitch_diff_sh16 = | |
1530 | 560 | (cur_pitch_val - s->last_pitch_val) * (1 << 16) / MAX_FRAMESIZE; | |
1531 | } | ||
1532 | |||
1533 | /* Global gain (if silence) and pitch-adaptive window coordinates */ | ||
1534 |
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|
3306 | switch (frame_descs[bd_idx].fcb_type) { |
1535 | 499 | case FCB_TYPE_SILENCE: | |
1536 | 499 | s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; | |
1537 | 499 | break; | |
1538 | 341 | case FCB_TYPE_AW_PULSES: | |
1539 | 341 | aw_parse_coords(s, gb, pitch); | |
1540 | 341 | break; | |
1541 | } | ||
1542 | |||
1543 |
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|
14089 | for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { |
1544 | int bl_pitch_sh2; | ||
1545 | |||
1546 | /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ | ||
1547 |
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|
10783 | switch (frame_descs[bd_idx].acb_type) { |
1548 | 8464 | case ACB_TYPE_HAMMING: { | |
1549 | /* Pitch is given per block. Per-block pitches are encoded as an | ||
1550 | * absolute value for the first block, and then delta values | ||
1551 | * relative to this value) for all subsequent blocks. The scale of | ||
1552 | * this pitch value is semi-logarithmic compared to its use in the | ||
1553 | * decoder, so we convert it to normal scale also. */ | ||
1554 | int block_pitch, | ||
1555 | 8464 | t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, | |
1556 | 8464 | t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, | |
1557 | 8464 | t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; | |
1558 | |||
1559 |
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|
8464 | if (n == 0) { |
1560 | 1975 | block_pitch = get_bits(gb, s->block_pitch_nbits); | |
1561 | } else | ||
1562 | 6489 | block_pitch = last_block_pitch - s->block_delta_pitch_hrange + | |
1563 | 6489 | get_bits(gb, s->block_delta_pitch_nbits); | |
1564 | /* Convert last_ so that any next delta is within _range */ | ||
1565 | 8464 | last_block_pitch = av_clip(block_pitch, | |
1566 | s->block_delta_pitch_hrange, | ||
1567 | 8464 | s->block_pitch_range - | |
1568 | 8464 | s->block_delta_pitch_hrange); | |
1569 | |||
1570 | /* Convert semi-log-style scale back to normal scale */ | ||
1571 |
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|
8464 | if (block_pitch < t1) { |
1572 | 1491 | bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; | |
1573 | } else { | ||
1574 | 6973 | block_pitch -= t1; | |
1575 |
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|
6973 | if (block_pitch < t2) { |
1576 | 5712 | bl_pitch_sh2 = | |
1577 | 5712 | (s->block_conv_table[1] << 2) + (block_pitch << 1); | |
1578 | } else { | ||
1579 | 1261 | block_pitch -= t2; | |
1580 |
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|
1261 | if (block_pitch < t3) { |
1581 | 1261 | bl_pitch_sh2 = | |
1582 | 1261 | (s->block_conv_table[2] + block_pitch) << 2; | |
1583 | } else | ||
1584 | ✗ | bl_pitch_sh2 = s->block_conv_table[3] << 2; | |
1585 | } | ||
1586 | } | ||
1587 | 8464 | pitch[n] = bl_pitch_sh2 >> 2; | |
1588 | 8464 | break; | |
1589 | } | ||
1590 | |||
1591 | 1276 | case ACB_TYPE_ASYMMETRIC: { | |
1592 | 1276 | bl_pitch_sh2 = pitch[n] << 2; | |
1593 | 1276 | break; | |
1594 | } | ||
1595 | |||
1596 | 1043 | default: // ACB_TYPE_NONE has no pitch | |
1597 | 1043 | bl_pitch_sh2 = 0; | |
1598 | 1043 | break; | |
1599 | } | ||
1600 | |||
1601 | 10783 | synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, | |
1602 | lsps, prev_lsps, &frame_descs[bd_idx], | ||
1603 | 10783 | &excitation[n * block_nsamples], | |
1604 | 10783 | &synth[n * block_nsamples]); | |
1605 | } | ||
1606 | |||
1607 | /* Averaging projection filter, if applicable. Else, just copy samples | ||
1608 | * from synthesis buffer */ | ||
1609 |
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|
3306 | if (s->do_apf) { |
1610 | double i_lsps[MAX_LSPS]; | ||
1611 | float lpcs[MAX_LSPS]; | ||
1612 | |||
1613 |
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|
46266 | for (n = 0; n < s->lsps; n++) // LSF -> LSP |
1614 | 42960 | i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); | |
1615 | 3306 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1616 | 3306 | postfilter(s, synth, samples, 80, lpcs, | |
1617 | 3306 | &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], | |
1618 | 3306 | frame_descs[bd_idx].fcb_type, pitch[0]); | |
1619 | |||
1620 |
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|
46266 | for (n = 0; n < s->lsps; n++) // LSF -> LSP |
1621 | 42960 | i_lsps[n] = cos(lsps[n]); | |
1622 | 3306 | ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1623 | 3306 | postfilter(s, &synth[80], &samples[80], 80, lpcs, | |
1624 | 3306 | &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], | |
1625 | 3306 | frame_descs[bd_idx].fcb_type, pitch[0]); | |
1626 | } else | ||
1627 | ✗ | memcpy(samples, synth, 160 * sizeof(synth[0])); | |
1628 | |||
1629 | /* Cache values for next frame */ | ||
1630 | 3306 | s->frame_cntr++; | |
1631 |
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|
3306 | if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) |
1632 | 3306 | s->last_acb_type = frame_descs[bd_idx].acb_type; | |
1633 |
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3306 | switch (frame_descs[bd_idx].acb_type) { |
1634 | 771 | case ACB_TYPE_NONE: | |
1635 | 771 | s->last_pitch_val = 0; | |
1636 | 771 | break; | |
1637 | 560 | case ACB_TYPE_ASYMMETRIC: | |
1638 | 560 | s->last_pitch_val = cur_pitch_val; | |
1639 | 560 | break; | |
1640 | 1975 | case ACB_TYPE_HAMMING: | |
1641 | 1975 | s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; | |
1642 | 1975 | break; | |
1643 | } | ||
1644 | |||
1645 | 3306 | return 0; | |
1646 | } | ||
1647 | |||
1648 | /** | ||
1649 | * Ensure minimum value for first item, maximum value for last value, | ||
1650 | * proper spacing between each value and proper ordering. | ||
1651 | * | ||
1652 | * @param lsps array of LSPs | ||
1653 | * @param num size of LSP array | ||
1654 | * | ||
1655 | * @note basically a double version of #ff_acelp_reorder_lsf(), might be | ||
1656 | * useful to put in a generic location later on. Parts are also | ||
1657 | * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), | ||
1658 | * which is in float. | ||
1659 | */ | ||
1660 | 3306 | static void stabilize_lsps(double *lsps, int num) | |
1661 | { | ||
1662 | int n, m, l; | ||
1663 | |||
1664 | /* set minimum value for first, maximum value for last and minimum | ||
1665 | * spacing between LSF values. | ||
1666 | * Very similar to ff_set_min_dist_lsf(), but in double. */ | ||
1667 |
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3306 | lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); |
1668 |
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42960 | for (n = 1; n < num; n++) |
1669 |
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|
39654 | lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); |
1670 |
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|
3306 | lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); |
1671 | |||
1672 | /* reorder (looks like one-time / non-recursed bubblesort). | ||
1673 | * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ | ||
1674 |
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42960 | for (n = 1; n < num; n++) { |
1675 |
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|
39654 | if (lsps[n] < lsps[n - 1]) { |
1676 | ✗ | for (m = 1; m < num; m++) { | |
1677 | ✗ | double tmp = lsps[m]; | |
1678 | ✗ | for (l = m - 1; l >= 0; l--) { | |
1679 | ✗ | if (lsps[l] <= tmp) break; | |
1680 | ✗ | lsps[l + 1] = lsps[l]; | |
1681 | } | ||
1682 | ✗ | lsps[l + 1] = tmp; | |
1683 | } | ||
1684 | ✗ | break; | |
1685 | } | ||
1686 | } | ||
1687 | 3306 | } | |
1688 | |||
1689 | /** | ||
1690 | * Synthesize output samples for a single superframe. If we have any data | ||
1691 | * cached in s->sframe_cache, that will be used instead of whatever is loaded | ||
1692 | * in s->gb. | ||
1693 | * | ||
1694 | * WMA Voice superframes contain 3 frames, each containing 160 audio samples, | ||
1695 | * to give a total of 480 samples per frame. See #synth_frame() for frame | ||
1696 | * parsing. In addition to 3 frames, superframes can also contain the LSPs | ||
1697 | * (if these are globally specified for all frames (residually); they can | ||
1698 | * also be specified individually per-frame. See the s->has_residual_lsps | ||
1699 | * option), and can specify the number of samples encoded in this superframe | ||
1700 | * (if less than 480), usually used to prevent blanks at track boundaries. | ||
1701 | * | ||
1702 | * @param ctx WMA Voice decoder context | ||
1703 | * @return 0 on success, <0 on error or 1 if there was not enough data to | ||
1704 | * fully parse the superframe | ||
1705 | */ | ||
1706 | 1102 | static int synth_superframe(AVCodecContext *ctx, AVFrame *frame, | |
1707 | int *got_frame_ptr) | ||
1708 | { | ||
1709 | 1102 | WMAVoiceContext *s = ctx->priv_data; | |
1710 | 1102 | GetBitContext *gb = &s->gb, s_gb; | |
1711 | 1102 | int n, res, n_samples = MAX_SFRAMESIZE; | |
1712 | double lsps[MAX_FRAMES][MAX_LSPS]; | ||
1713 | 2204 | const double *mean_lsf = s->lsps == 16 ? | |
1714 |
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1102 | wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; |
1715 | float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; | ||
1716 | float synth[MAX_LSPS + MAX_SFRAMESIZE]; | ||
1717 | float *samples; | ||
1718 | |||
1719 | 1102 | memcpy(synth, s->synth_history, | |
1720 | 1102 | s->lsps * sizeof(*synth)); | |
1721 | 1102 | memcpy(excitation, s->excitation_history, | |
1722 | 1102 | s->history_nsamples * sizeof(*excitation)); | |
1723 | |||
1724 |
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|
1102 | if (s->sframe_cache_size > 0) { |
1725 | 185 | gb = &s_gb; | |
1726 | 185 | init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); | |
1727 | 185 | s->sframe_cache_size = 0; | |
1728 | } | ||
1729 | |||
1730 | /* First bit is speech/music bit, it differentiates between WMAVoice | ||
1731 | * speech samples (the actual codec) and WMAVoice music samples, which | ||
1732 | * are really WMAPro-in-WMAVoice-superframes. I've never seen those in | ||
1733 | * the wild yet. */ | ||
1734 |
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|
1102 | if (!get_bits1(gb)) { |
1735 | ✗ | avpriv_request_sample(ctx, "WMAPro-in-WMAVoice"); | |
1736 | ✗ | return AVERROR_PATCHWELCOME; | |
1737 | } | ||
1738 | |||
1739 | /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ | ||
1740 |
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1102 | if (get_bits1(gb)) { |
1741 |
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|
3 | if ((n_samples = get_bits(gb, 12)) > MAX_SFRAMESIZE) { |
1742 | ✗ | av_log(ctx, AV_LOG_ERROR, | |
1743 | "Superframe encodes > %d samples (%d), not allowed\n", | ||
1744 | MAX_SFRAMESIZE, n_samples); | ||
1745 | ✗ | return AVERROR_INVALIDDATA; | |
1746 | } | ||
1747 | } | ||
1748 | |||
1749 | /* Parse LSPs, if global for the superframe (can also be per-frame). */ | ||
1750 |
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1102 | if (s->has_residual_lsps) { |
1751 | double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; | ||
1752 | |||
1753 |
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15422 | for (n = 0; n < s->lsps; n++) |
1754 | 14320 | prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; | |
1755 | |||
1756 |
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1102 | if (s->lsps == 10) { |
1757 | 552 | dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1758 | } else /* s->lsps == 16 */ | ||
1759 | 550 | dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1760 | |||
1761 |
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15422 | for (n = 0; n < s->lsps; n++) { |
1762 | 14320 | lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); | |
1763 | 14320 | lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); | |
1764 | 14320 | lsps[2][n] += mean_lsf[n]; | |
1765 | } | ||
1766 |
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4408 | for (n = 0; n < 3; n++) |
1767 | 3306 | stabilize_lsps(lsps[n], s->lsps); | |
1768 | } | ||
1769 | |||
1770 | /* synth_superframe can run multiple times per packet | ||
1771 | * free potential previous frame */ | ||
1772 | 1102 | av_frame_unref(frame); | |
1773 | |||
1774 | /* get output buffer */ | ||
1775 | 1102 | frame->nb_samples = MAX_SFRAMESIZE; | |
1776 |
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|
1102 | if ((res = ff_get_buffer(ctx, frame, 0)) < 0) |
1777 | ✗ | return res; | |
1778 | 1102 | frame->nb_samples = n_samples; | |
1779 | 1102 | samples = (float *)frame->data[0]; | |
1780 | |||
1781 | /* Parse frames, optionally preceded by per-frame (independent) LSPs. */ | ||
1782 |
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4408 | for (n = 0; n < 3; n++) { |
1783 |
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|
3306 | if (!s->has_residual_lsps) { |
1784 | int m; | ||
1785 | |||
1786 | ✗ | if (s->lsps == 10) { | |
1787 | ✗ | dequant_lsp10i(gb, lsps[n]); | |
1788 | } else /* s->lsps == 16 */ | ||
1789 | ✗ | dequant_lsp16i(gb, lsps[n]); | |
1790 | |||
1791 | ✗ | for (m = 0; m < s->lsps; m++) | |
1792 | ✗ | lsps[n][m] += mean_lsf[m]; | |
1793 | ✗ | stabilize_lsps(lsps[n], s->lsps); | |
1794 | } | ||
1795 | |||
1796 |
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|
4408 | if ((res = synth_frame(ctx, gb, n, |
1797 | 3306 | &samples[n * MAX_FRAMESIZE], | |
1798 | 3306 | lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], | |
1799 | 3306 | &excitation[s->history_nsamples + n * MAX_FRAMESIZE], | |
1800 |
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|
3306 | &synth[s->lsps + n * MAX_FRAMESIZE]))) { |
1801 | ✗ | *got_frame_ptr = 0; | |
1802 | ✗ | return res; | |
1803 | } | ||
1804 | } | ||
1805 | |||
1806 | /* Statistics? FIXME - we don't check for length, a slight overrun | ||
1807 | * will be caught by internal buffer padding, and anything else | ||
1808 | * will be skipped, not read. */ | ||
1809 |
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|
1102 | if (get_bits1(gb)) { |
1810 | ✗ | res = get_bits(gb, 4); | |
1811 | ✗ | skip_bits(gb, 10 * (res + 1)); | |
1812 | } | ||
1813 | |||
1814 |
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|
1102 | if (get_bits_left(gb) < 0) { |
1815 | ✗ | wmavoice_flush(ctx); | |
1816 | ✗ | return AVERROR_INVALIDDATA; | |
1817 | } | ||
1818 | |||
1819 | 1102 | *got_frame_ptr = 1; | |
1820 | |||
1821 | /* Update history */ | ||
1822 | 1102 | memcpy(s->prev_lsps, lsps[2], | |
1823 | 1102 | s->lsps * sizeof(*s->prev_lsps)); | |
1824 | 1102 | memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], | |
1825 | 1102 | s->lsps * sizeof(*synth)); | |
1826 | 1102 | memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], | |
1827 | 1102 | s->history_nsamples * sizeof(*excitation)); | |
1828 |
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1102 | if (s->do_apf) |
1829 | 1102 | memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], | |
1830 | 1102 | s->history_nsamples * sizeof(*s->zero_exc_pf)); | |
1831 | |||
1832 | 1102 | return 0; | |
1833 | } | ||
1834 | |||
1835 | /** | ||
1836 | * Parse the packet header at the start of each packet (input data to this | ||
1837 | * decoder). | ||
1838 | * | ||
1839 | * @param s WMA Voice decoding context private data | ||
1840 | * @return <0 on error, nb_superframes on success. | ||
1841 | */ | ||
1842 | 186 | static int parse_packet_header(WMAVoiceContext *s) | |
1843 | { | ||
1844 | 186 | GetBitContext *gb = &s->gb; | |
1845 | 186 | unsigned int res, n_superframes = 0; | |
1846 | |||
1847 | 186 | skip_bits(gb, 4); // packet sequence number | |
1848 | 186 | s->has_residual_lsps = get_bits1(gb); | |
1849 | do { | ||
1850 |
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|
186 | if (get_bits_left(gb) < 6 + s->spillover_bitsize) |
1851 | ✗ | return AVERROR_INVALIDDATA; | |
1852 | |||
1853 | 186 | res = get_bits(gb, 6); // number of superframes per packet | |
1854 | // (minus first one if there is spillover) | ||
1855 | 186 | n_superframes += res; | |
1856 |
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|
186 | } while (res == 0x3F); |
1857 | 186 | s->spillover_nbits = get_bits(gb, s->spillover_bitsize); | |
1858 | |||
1859 |
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|
186 | return get_bits_left(gb) >= 0 ? n_superframes : AVERROR_INVALIDDATA; |
1860 | } | ||
1861 | |||
1862 | /** | ||
1863 | * Copy (unaligned) bits from gb/data/size to pb. | ||
1864 | * | ||
1865 | * @param pb target buffer to copy bits into | ||
1866 | * @param data source buffer to copy bits from | ||
1867 | * @param size size of the source data, in bytes | ||
1868 | * @param gb bit I/O context specifying the current position in the source. | ||
1869 | * data. This function might use this to align the bit position to | ||
1870 | * a whole-byte boundary before calling #ff_copy_bits() on aligned | ||
1871 | * source data | ||
1872 | * @param nbits the amount of bits to copy from source to target | ||
1873 | * | ||
1874 | * @note after calling this function, the current position in the input bit | ||
1875 | * I/O context is undefined. | ||
1876 | */ | ||
1877 | 370 | static void copy_bits(PutBitContext *pb, | |
1878 | const uint8_t *data, int size, | ||
1879 | GetBitContext *gb, int nbits) | ||
1880 | { | ||
1881 | int rmn_bytes, rmn_bits; | ||
1882 | |||
1883 | 370 | rmn_bits = rmn_bytes = get_bits_left(gb); | |
1884 |
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|
370 | if (rmn_bits < nbits) |
1885 | ✗ | return; | |
1886 |
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370 | if (nbits > put_bits_left(pb)) |
1887 | ✗ | return; | |
1888 | 370 | rmn_bits &= 7; rmn_bytes >>= 3; | |
1889 |
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370 | if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) |
1890 | 290 | put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); | |
1891 | 370 | ff_copy_bits(pb, data + size - rmn_bytes, | |
1892 | 370 | FFMIN(nbits - rmn_bits, rmn_bytes << 3)); | |
1893 | } | ||
1894 | |||
1895 | /** | ||
1896 | * Packet decoding: a packet is anything that the (ASF) demuxer contains, | ||
1897 | * and we expect that the demuxer / application provides it to us as such | ||
1898 | * (else you'll probably get garbage as output). Every packet has a size of | ||
1899 | * ctx->block_align bytes, starts with a packet header (see | ||
1900 | * #parse_packet_header()), and then a series of superframes. Superframe | ||
1901 | * boundaries may exceed packets, i.e. superframes can split data over | ||
1902 | * multiple (two) packets. | ||
1903 | * | ||
1904 | * For more information about frames, see #synth_superframe(). | ||
1905 | */ | ||
1906 | 1291 | static int wmavoice_decode_packet(AVCodecContext *ctx, AVFrame *frame, | |
1907 | int *got_frame_ptr, AVPacket *avpkt) | ||
1908 | { | ||
1909 | 1291 | WMAVoiceContext *s = ctx->priv_data; | |
1910 | 1291 | GetBitContext *gb = &s->gb; | |
1911 | int size, res, pos; | ||
1912 | |||
1913 | /* Packets are sometimes a multiple of ctx->block_align, with a packet | ||
1914 | * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer | ||
1915 | * feeds us ASF packets, which may concatenate multiple "codec" packets | ||
1916 | * in a single "muxer" packet, so we artificially emulate that by | ||
1917 | * capping the packet size at ctx->block_align. */ | ||
1918 |
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1471 | for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); |
1919 | 1291 | init_get_bits8(&s->gb, avpkt->data, size); | |
1920 | |||
1921 | /* size == ctx->block_align is used to indicate whether we are dealing with | ||
1922 | * a new packet or a packet of which we already read the packet header | ||
1923 | * previously. */ | ||
1924 |
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1291 | if (!(size % ctx->block_align)) { // new packet header |
1925 |
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191 | if (!size) { |
1926 | 5 | s->spillover_nbits = 0; | |
1927 | 5 | s->nb_superframes = 0; | |
1928 | } else { | ||
1929 |
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186 | if ((res = parse_packet_header(s)) < 0) |
1930 | ✗ | return res; | |
1931 | 186 | s->nb_superframes = res; | |
1932 | } | ||
1933 | |||
1934 | /* If the packet header specifies a s->spillover_nbits, then we want | ||
1935 | * to push out all data of the previous packet (+ spillover) before | ||
1936 | * continuing to parse new superframes in the current packet. */ | ||
1937 |
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191 | if (s->sframe_cache_size > 0) { |
1938 | 185 | int cnt = get_bits_count(gb); | |
1939 |
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185 | if (cnt + s->spillover_nbits > avpkt->size * 8) { |
1940 | ✗ | s->spillover_nbits = avpkt->size * 8 - cnt; | |
1941 | } | ||
1942 | 185 | copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); | |
1943 | 185 | flush_put_bits(&s->pb); | |
1944 | 185 | s->sframe_cache_size += s->spillover_nbits; | |
1945 |
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185 | if ((res = synth_superframe(ctx, frame, got_frame_ptr)) == 0 && |
1946 |
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185 | *got_frame_ptr) { |
1947 | 185 | cnt += s->spillover_nbits; | |
1948 | 185 | s->skip_bits_next = cnt & 7; | |
1949 | 185 | res = cnt >> 3; | |
1950 | 185 | return res; | |
1951 | } else | ||
1952 | ✗ | skip_bits_long (gb, s->spillover_nbits - cnt + | |
1953 | ✗ | get_bits_count(gb)); // resync | |
1954 |
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6 | } else if (s->spillover_nbits) { |
1955 | ✗ | skip_bits_long(gb, s->spillover_nbits); // resync | |
1956 | } | ||
1957 |
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1100 | } else if (s->skip_bits_next) |
1958 | 971 | skip_bits(gb, s->skip_bits_next); | |
1959 | |||
1960 | /* Try parsing superframes in current packet */ | ||
1961 | 1106 | s->sframe_cache_size = 0; | |
1962 | 1106 | s->skip_bits_next = 0; | |
1963 | 1106 | pos = get_bits_left(gb); | |
1964 |
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1106 | if (s->nb_superframes-- == 0) { |
1965 | 4 | *got_frame_ptr = 0; | |
1966 | 4 | return size; | |
1967 |
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1102 | } else if (s->nb_superframes > 0) { |
1968 |
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917 | if ((res = synth_superframe(ctx, frame, got_frame_ptr)) < 0) { |
1969 | ✗ | return res; | |
1970 |
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917 | } else if (*got_frame_ptr) { |
1971 | 917 | int cnt = get_bits_count(gb); | |
1972 | 917 | s->skip_bits_next = cnt & 7; | |
1973 | 917 | res = cnt >> 3; | |
1974 | 917 | return res; | |
1975 | } | ||
1976 |
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185 | } else if ((s->sframe_cache_size = pos) > 0) { |
1977 | /* ... cache it for spillover in next packet */ | ||
1978 | 185 | init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); | |
1979 | 185 | copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); | |
1980 | // FIXME bad - just copy bytes as whole and add use the | ||
1981 | // skip_bits_next field | ||
1982 | } | ||
1983 | |||
1984 | 185 | return size; | |
1985 | } | ||
1986 | |||
1987 | 8 | static av_cold int wmavoice_decode_end(AVCodecContext *ctx) | |
1988 | { | ||
1989 | 8 | WMAVoiceContext *s = ctx->priv_data; | |
1990 | |||
1991 |
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8 | if (s->do_apf) { |
1992 | 8 | ff_rdft_end(&s->rdft); | |
1993 | 8 | ff_rdft_end(&s->irdft); | |
1994 | 8 | ff_dct_end(&s->dct); | |
1995 | 8 | ff_dct_end(&s->dst); | |
1996 | } | ||
1997 | |||
1998 | 8 | return 0; | |
1999 | } | ||
2000 | |||
2001 | const FFCodec ff_wmavoice_decoder = { | ||
2002 | .p.name = "wmavoice", | ||
2003 | .p.long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), | ||
2004 | .p.type = AVMEDIA_TYPE_AUDIO, | ||
2005 | .p.id = AV_CODEC_ID_WMAVOICE, | ||
2006 | .priv_data_size = sizeof(WMAVoiceContext), | ||
2007 | .init = wmavoice_decode_init, | ||
2008 | .close = wmavoice_decode_end, | ||
2009 | FF_CODEC_DECODE_CB(wmavoice_decode_packet), | ||
2010 | .p.capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, | ||
2011 | .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, | ||
2012 | .flush = wmavoice_flush, | ||
2013 | }; | ||
2014 |