Mercurial > libavcodec.hg
annotate wmavoice.c @ 12104:2a6873ee2fc9 libavcodec
ARM: hide a .size directive on non-ELF targets
author | mru |
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date | Wed, 07 Jul 2010 20:09:37 +0000 |
parents | fdafbcef52f5 |
children | b00c10643f24 |
rev | line source |
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11123 | 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 /** | |
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23 * @file |
11123 | 24 * @brief Windows Media Audio Voice compatible decoder |
25 * @author Ronald S. Bultje <rsbultje@gmail.com> | |
26 */ | |
27 | |
28 #include <math.h> | |
29 #include "avcodec.h" | |
30 #include "get_bits.h" | |
31 #include "put_bits.h" | |
32 #include "wmavoice_data.h" | |
33 #include "celp_math.h" | |
34 #include "celp_filters.h" | |
35 #include "acelp_vectors.h" | |
36 #include "acelp_filters.h" | |
37 #include "lsp.h" | |
38 #include "libavutil/lzo.h" | |
11653 | 39 #include "avfft.h" |
40 #include "fft.h" | |
11123 | 41 |
42 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame | |
43 #define MAX_LSPS 16 ///< maximum filter order | |
11653 | 44 #define MAX_LSPS_ALIGN16 16 ///< same as #MAX_LSPS; needs to be multiple |
45 ///< of 16 for ASM input buffer alignment | |
11123 | 46 #define MAX_FRAMES 3 ///< maximum number of frames per superframe |
47 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame | |
48 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history | |
49 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) | |
50 ///< maximum number of samples per superframe | |
51 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that | |
52 ///< was split over two packets | |
53 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration | |
54 | |
55 /** | |
56 * Frame type VLC coding. | |
57 */ | |
58 static VLC frame_type_vlc; | |
59 | |
60 /** | |
61 * Adaptive codebook types. | |
62 */ | |
63 enum { | |
64 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) | |
65 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which | |
66 ///< we interpolate to get a per-sample pitch. | |
67 ///< Signal is generated using an asymmetric sinc | |
68 ///< window function | |
69 ///< @note see #wmavoice_ipol1_coeffs | |
70 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using | |
71 ///< a Hamming sinc window function | |
72 ///< @note see #wmavoice_ipol2_coeffs | |
73 }; | |
74 | |
75 /** | |
76 * Fixed codebook types. | |
77 */ | |
78 enum { | |
79 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence | |
80 ///< generated from a hardcoded (fixed) codebook | |
81 ///< with per-frame (low) gain values | |
82 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block | |
83 ///< gain values | |
84 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, | |
85 ///< used in particular for low-bitrate streams | |
86 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in | |
87 ///< combinations of either single pulses or | |
88 ///< pulse pairs | |
89 }; | |
90 | |
91 /** | |
92 * Description of frame types. | |
93 */ | |
94 static const struct frame_type_desc { | |
95 uint8_t n_blocks; ///< amount of blocks per frame (each block | |
96 ///< (contains 160/#n_blocks samples) | |
97 uint8_t log_n_blocks; ///< log2(#n_blocks) | |
98 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) | |
99 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) | |
100 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs | |
101 ///< (rather than just one single pulse) | |
102 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES | |
103 uint16_t frame_size; ///< the amount of bits that make up the block | |
104 ///< data (per frame) | |
105 } frame_descs[17] = { | |
106 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, | |
107 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, | |
108 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, | |
109 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, | |
110 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, | |
111 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, | |
112 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, | |
113 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, | |
114 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, | |
115 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, | |
116 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, | |
117 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, | |
118 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, | |
119 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, | |
120 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, | |
121 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, | |
122 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } | |
123 }; | |
124 | |
125 /** | |
126 * WMA Voice decoding context. | |
127 */ | |
128 typedef struct { | |
129 /** | |
130 * @defgroup struct_global Global values | |
131 * Global values, specified in the stream header / extradata or used | |
132 * all over. | |
133 * @{ | |
134 */ | |
135 GetBitContext gb; ///< packet bitreader. During decoder init, | |
136 ///< it contains the extradata from the | |
137 ///< demuxer. During decoding, it contains | |
138 ///< packet data. | |
139 int8_t vbm_tree[25]; ///< converts VLC codes to frame type | |
140 | |
141 int spillover_bitsize; ///< number of bits used to specify | |
142 ///< #spillover_nbits in the packet header | |
143 ///< = ceil(log2(ctx->block_align << 3)) | |
144 int history_nsamples; ///< number of samples in history for signal | |
145 ///< prediction (through ACB) | |
146 | |
11653 | 147 /* postfilter specific values */ |
11123 | 148 int do_apf; ///< whether to apply the averaged |
149 ///< projection filter (APF) | |
11653 | 150 int denoise_strength; ///< strength of denoising in Wiener filter |
151 ///< [0-11] | |
152 int denoise_tilt_corr; ///< Whether to apply tilt correction to the | |
153 ///< Wiener filter coefficients (postfilter) | |
154 int dc_level; ///< Predicted amount of DC noise, based | |
155 ///< on which a DC removal filter is used | |
11123 | 156 |
157 int lsps; ///< number of LSPs per frame [10 or 16] | |
158 int lsp_q_mode; ///< defines quantizer defaults [0, 1] | |
159 int lsp_def_mode; ///< defines different sets of LSP defaults | |
160 ///< [0, 1] | |
161 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded | |
162 ///< per-frame (independent coding) | |
163 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded | |
164 ///< per superframe (residual coding) | |
165 | |
166 int min_pitch_val; ///< base value for pitch parsing code | |
167 int max_pitch_val; ///< max value + 1 for pitch parsing | |
168 int pitch_nbits; ///< number of bits used to specify the | |
169 ///< pitch value in the frame header | |
170 int block_pitch_nbits; ///< number of bits used to specify the | |
171 ///< first block's pitch value | |
172 int block_pitch_range; ///< range of the block pitch | |
173 int block_delta_pitch_nbits; ///< number of bits used to specify the | |
174 ///< delta pitch between this and the last | |
175 ///< block's pitch value, used in all but | |
176 ///< first block | |
177 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is | |
178 ///< from -this to +this-1) | |
179 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale | |
180 ///< conversion | |
181 | |
182 /** | |
183 * @} | |
184 * @defgroup struct_packet Packet values | |
185 * Packet values, specified in the packet header or related to a packet. | |
186 * A packet is considered to be a single unit of data provided to this | |
187 * decoder by the demuxer. | |
188 * @{ | |
189 */ | |
190 int spillover_nbits; ///< number of bits of the previous packet's | |
191 ///< last superframe preceeding this | |
192 ///< packet's first full superframe (useful | |
193 ///< for re-synchronization also) | |
194 int has_residual_lsps; ///< if set, superframes contain one set of | |
195 ///< LSPs that cover all frames, encoded as | |
196 ///< independent and residual LSPs; if not | |
197 ///< set, each frame contains its own, fully | |
198 ///< independent, LSPs | |
199 int skip_bits_next; ///< number of bits to skip at the next call | |
200 ///< to #wmavoice_decode_packet() (since | |
201 ///< they're part of the previous superframe) | |
202 | |
203 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE]; | |
204 ///< cache for superframe data split over | |
205 ///< multiple packets | |
206 int sframe_cache_size; ///< set to >0 if we have data from an | |
207 ///< (incomplete) superframe from a previous | |
208 ///< packet that spilled over in the current | |
209 ///< packet; specifies the amount of bits in | |
210 ///< #sframe_cache | |
211 PutBitContext pb; ///< bitstream writer for #sframe_cache | |
212 | |
213 /** | |
214 * @} | |
215 * @defgroup struct_frame 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 float gain_pred_err[6]; ///< cache for gain prediction | |
251 float excitation_history[MAX_SIGNAL_HISTORY]; | |
252 ///< cache of the signal of previous | |
253 ///< superframes, used as a history for | |
254 ///< signal generation | |
255 float synth_history[MAX_LSPS]; ///< see #excitation_history | |
256 /** | |
257 * @} | |
11653 | 258 * @defgroup post_filter Postfilter values |
259 * Varibales used for postfilter implementation, mostly history for | |
260 * smoothing and so on, and context variables for FFT/iFFT. | |
261 * @{ | |
262 */ | |
263 RDFTContext rdft, irdft; ///< contexts for FFT-calculation in the | |
264 ///< postfilter (for denoise filter) | |
265 DCTContext dct, dst; ///< contexts for phase shift (in Hilbert | |
266 ///< transform, part of postfilter) | |
267 float sin[511], cos[511]; ///< 8-bit cosine/sine windows over [-pi,pi] | |
268 ///< range | |
269 float postfilter_agc; ///< gain control memory, used in | |
270 ///< #adaptive_gain_control() | |
271 float dcf_mem[2]; ///< DC filter history | |
272 float zero_exc_pf[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE]; | |
273 ///< zero filter output (i.e. excitation) | |
274 ///< by postfilter | |
275 float denoise_filter_cache[MAX_FRAMESIZE]; | |
276 int denoise_filter_cache_size; ///< samples in #denoise_filter_cache | |
277 DECLARE_ALIGNED(16, float, tilted_lpcs_pf)[0x80]; | |
278 ///< aligned buffer for LPC tilting | |
279 DECLARE_ALIGNED(16, float, denoise_coeffs_pf)[0x80]; | |
280 ///< aligned buffer for denoise coefficients | |
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281 DECLARE_ALIGNED(16, float, synth_filter_out_buf)[0x80 + MAX_LSPS_ALIGN16]; |
11653 | 282 ///< aligned buffer for postfilter speech |
283 ///< synthesis | |
284 /** | |
285 * @} | |
11123 | 286 */ |
287 } WMAVoiceContext; | |
288 | |
289 /** | |
12024 | 290 * Set up the variable bit mode (VBM) tree from container extradata. |
11123 | 291 * @param gb bit I/O context. |
292 * The bit context (s->gb) should be loaded with byte 23-46 of the | |
293 * container extradata (i.e. the ones containing the VBM tree). | |
294 * @param vbm_tree pointer to array to which the decoded VBM tree will be | |
295 * written. | |
296 * @return 0 on success, <0 on error. | |
297 */ | |
298 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) | |
299 { | |
300 static const uint8_t bits[] = { | |
301 2, 2, 2, 4, 4, 4, | |
302 6, 6, 6, 8, 8, 8, | |
303 10, 10, 10, 12, 12, 12, | |
304 14, 14, 14, 14 | |
305 }; | |
306 static const uint16_t codes[] = { | |
307 0x0000, 0x0001, 0x0002, // 00/01/10 | |
308 0x000c, 0x000d, 0x000e, // 11+00/01/10 | |
309 0x003c, 0x003d, 0x003e, // 1111+00/01/10 | |
310 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 | |
311 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 | |
312 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 | |
313 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx | |
314 }; | |
315 int cntr[8], n, res; | |
316 | |
317 memset(vbm_tree, 0xff, sizeof(vbm_tree)); | |
318 memset(cntr, 0, sizeof(cntr)); | |
319 for (n = 0; n < 17; n++) { | |
320 res = get_bits(gb, 3); | |
321 if (cntr[res] > 3) // should be >= 3 + (res == 7)) | |
322 return -1; | |
323 vbm_tree[res * 3 + cntr[res]++] = n; | |
324 } | |
325 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), | |
326 bits, 1, 1, codes, 2, 2, 132); | |
327 return 0; | |
328 } | |
329 | |
330 /** | |
331 * Set up decoder with parameters from demuxer (extradata etc.). | |
332 */ | |
333 static av_cold int wmavoice_decode_init(AVCodecContext *ctx) | |
334 { | |
335 int n, flags, pitch_range, lsp16_flag; | |
336 WMAVoiceContext *s = ctx->priv_data; | |
337 | |
338 /** | |
339 * Extradata layout: | |
340 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), | |
341 * - byte 19-22: flags field (annoyingly in LE; see below for known | |
342 * values), | |
343 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, | |
344 * rest is 0). | |
345 */ | |
346 if (ctx->extradata_size != 46) { | |
347 av_log(ctx, AV_LOG_ERROR, | |
348 "Invalid extradata size %d (should be 46)\n", | |
349 ctx->extradata_size); | |
350 return -1; | |
351 } | |
352 flags = AV_RL32(ctx->extradata + 18); | |
353 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); | |
354 s->do_apf = flags & 0x1; | |
11653 | 355 if (s->do_apf) { |
356 ff_rdft_init(&s->rdft, 7, DFT_R2C); | |
357 ff_rdft_init(&s->irdft, 7, IDFT_C2R); | |
358 ff_dct_init(&s->dct, 6, DCT_I); | |
359 ff_dct_init(&s->dst, 6, DST_I); | |
360 | |
361 ff_sine_window_init(s->cos, 256); | |
362 memcpy(&s->sin[255], s->cos, 256 * sizeof(s->cos[0])); | |
363 for (n = 0; n < 255; n++) { | |
364 s->sin[n] = -s->sin[510 - n]; | |
365 s->cos[510 - n] = s->cos[n]; | |
366 } | |
367 } | |
368 s->denoise_strength = (flags >> 2) & 0xF; | |
369 if (s->denoise_strength >= 12) { | |
370 av_log(ctx, AV_LOG_ERROR, | |
371 "Invalid denoise filter strength %d (max=11)\n", | |
372 s->denoise_strength); | |
373 return -1; | |
374 } | |
375 s->denoise_tilt_corr = !!(flags & 0x40); | |
376 s->dc_level = (flags >> 7) & 0xF; | |
11123 | 377 s->lsp_q_mode = !!(flags & 0x2000); |
378 s->lsp_def_mode = !!(flags & 0x4000); | |
379 lsp16_flag = flags & 0x1000; | |
380 if (lsp16_flag) { | |
381 s->lsps = 16; | |
382 s->frame_lsp_bitsize = 34; | |
383 s->sframe_lsp_bitsize = 60; | |
384 } else { | |
385 s->lsps = 10; | |
386 s->frame_lsp_bitsize = 24; | |
387 s->sframe_lsp_bitsize = 48; | |
388 } | |
389 for (n = 0; n < s->lsps; n++) | |
390 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
391 | |
392 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); | |
393 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { | |
394 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); | |
395 return -1; | |
396 } | |
397 | |
398 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; | |
399 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; | |
400 pitch_range = s->max_pitch_val - s->min_pitch_val; | |
401 s->pitch_nbits = av_ceil_log2(pitch_range); | |
402 s->last_pitch_val = 40; | |
403 s->last_acb_type = ACB_TYPE_NONE; | |
404 s->history_nsamples = s->max_pitch_val + 8; | |
405 | |
406 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { | |
407 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, | |
408 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; | |
409 | |
410 av_log(ctx, AV_LOG_ERROR, | |
411 "Unsupported samplerate %d (min=%d, max=%d)\n", | |
412 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz | |
413 | |
414 return -1; | |
415 } | |
416 | |
417 s->block_conv_table[0] = s->min_pitch_val; | |
418 s->block_conv_table[1] = (pitch_range * 25) >> 6; | |
419 s->block_conv_table[2] = (pitch_range * 44) >> 6; | |
420 s->block_conv_table[3] = s->max_pitch_val - 1; | |
421 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; | |
422 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); | |
423 s->block_pitch_range = s->block_conv_table[2] + | |
424 s->block_conv_table[3] + 1 + | |
425 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); | |
426 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); | |
427 | |
428 ctx->sample_fmt = SAMPLE_FMT_FLT; | |
429 | |
430 return 0; | |
431 } | |
432 | |
433 /** | |
11653 | 434 * @defgroup postfilter Postfilter functions |
435 * Postfilter functions (gain control, wiener denoise filter, DC filter, | |
436 * kalman smoothening, plus surrounding code to wrap it) | |
437 * @{ | |
438 */ | |
439 /** | |
440 * Adaptive gain control (as used in postfilter). | |
441 * | |
442 * Identical to #ff_adaptive_gain_control() in acelp_vectors.c, except | |
443 * that the energy here is calculated using sum(abs(...)), whereas the | |
444 * other codecs (e.g. AMR-NB, SIPRO) use sqrt(dotproduct(...)). | |
445 * | |
446 * @param out output buffer for filtered samples | |
447 * @param in input buffer containing the samples as they are after the | |
448 * postfilter steps so far | |
449 * @param speech_synth input buffer containing speech synth before postfilter | |
450 * @param size input buffer size | |
451 * @param alpha exponential filter factor | |
452 * @param gain_mem pointer to filter memory (single float) | |
453 */ | |
454 static void adaptive_gain_control(float *out, const float *in, | |
455 const float *speech_synth, | |
456 int size, float alpha, float *gain_mem) | |
457 { | |
458 int i; | |
459 float speech_energy = 0.0, postfilter_energy = 0.0, gain_scale_factor; | |
460 float mem = *gain_mem; | |
461 | |
462 for (i = 0; i < size; i++) { | |
463 speech_energy += fabsf(speech_synth[i]); | |
464 postfilter_energy += fabsf(in[i]); | |
465 } | |
466 gain_scale_factor = (1.0 - alpha) * speech_energy / postfilter_energy; | |
467 | |
468 for (i = 0; i < size; i++) { | |
469 mem = alpha * mem + gain_scale_factor; | |
470 out[i] = in[i] * mem; | |
471 } | |
472 | |
473 *gain_mem = mem; | |
474 } | |
475 | |
476 /** | |
477 * Kalman smoothing function. | |
478 * | |
479 * This function looks back pitch +/- 3 samples back into history to find | |
480 * the best fitting curve (that one giving the optimal gain of the two | |
481 * signals, i.e. the highest dot product between the two), and then | |
482 * uses that signal history to smoothen the output of the speech synthesis | |
483 * filter. | |
484 * | |
485 * @param s WMA Voice decoding context | |
486 * @param pitch pitch of the speech signal | |
487 * @param in input speech signal | |
488 * @param out output pointer for smoothened signal | |
489 * @param size input/output buffer size | |
490 * | |
491 * @returns -1 if no smoothening took place, e.g. because no optimal | |
492 * fit could be found, or 0 on success. | |
493 */ | |
494 static int kalman_smoothen(WMAVoiceContext *s, int pitch, | |
495 const float *in, float *out, int size) | |
496 { | |
497 int n; | |
498 float optimal_gain = 0, dot; | |
499 const float *ptr = &in[-FFMAX(s->min_pitch_val, pitch - 3)], | |
500 *end = &in[-FFMIN(s->max_pitch_val, pitch + 3)], | |
501 *best_hist_ptr; | |
502 | |
503 /* find best fitting point in history */ | |
504 do { | |
505 dot = ff_dot_productf(in, ptr, size); | |
506 if (dot > optimal_gain) { | |
507 optimal_gain = dot; | |
508 best_hist_ptr = ptr; | |
509 } | |
510 } while (--ptr >= end); | |
511 | |
512 if (optimal_gain <= 0) | |
513 return -1; | |
514 dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size); | |
515 if (dot <= 0) // would be 1.0 | |
516 return -1; | |
517 | |
518 if (optimal_gain <= dot) { | |
519 dot = dot / (dot + 0.6 * optimal_gain); // 0.625-1.000 | |
520 } else | |
521 dot = 0.625; | |
522 | |
523 /* actual smoothing */ | |
524 for (n = 0; n < size; n++) | |
525 out[n] = best_hist_ptr[n] + dot * (in[n] - best_hist_ptr[n]); | |
526 | |
527 return 0; | |
528 } | |
529 | |
530 /** | |
531 * Get the tilt factor of a formant filter from its transfer function | |
532 * @see #tilt_factor() in amrnbdec.c, which does essentially the same, | |
533 * but somehow (??) it does a speech synthesis filter in the | |
534 * middle, which is missing here | |
535 * | |
536 * @param lpcs LPC coefficients | |
537 * @param n_lpcs Size of LPC buffer | |
538 * @returns the tilt factor | |
539 */ | |
540 static float tilt_factor(const float *lpcs, int n_lpcs) | |
541 { | |
542 float rh0, rh1; | |
543 | |
544 rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs); | |
545 rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1); | |
546 | |
547 return rh1 / rh0; | |
548 } | |
549 | |
550 /** | |
551 * Derive denoise filter coefficients (in real domain) from the LPCs. | |
552 */ | |
553 static void calc_input_response(WMAVoiceContext *s, float *lpcs, | |
554 int fcb_type, float *coeffs, int remainder) | |
555 { | |
556 float last_coeff, min = 15.0, max = -15.0; | |
557 float irange, angle_mul, gain_mul, range, sq; | |
558 int n, idx; | |
559 | |
560 /* Create frequency power spectrum of speech input (i.e. RDFT of LPCs) */ | |
561 ff_rdft_calc(&s->rdft, lpcs); | |
562 #define log_range(var, assign) do { \ | |
563 float tmp = log10f(assign); var = tmp; \ | |
564 max = FFMAX(max, tmp); min = FFMIN(min, tmp); \ | |
565 } while (0) | |
566 log_range(last_coeff, lpcs[1] * lpcs[1]); | |
567 for (n = 1; n < 64; n++) | |
568 log_range(lpcs[n], lpcs[n * 2] * lpcs[n * 2] + | |
569 lpcs[n * 2 + 1] * lpcs[n * 2 + 1]); | |
570 log_range(lpcs[0], lpcs[0] * lpcs[0]); | |
571 #undef log_range | |
572 range = max - min; | |
573 lpcs[64] = last_coeff; | |
574 | |
575 /* Now, use this spectrum to pick out these frequencies with higher | |
576 * (relative) power/energy (which we then take to be "not noise"), | |
577 * and set up a table (still in lpc[]) of (relative) gains per frequency. | |
578 * These frequencies will be maintained, while others ("noise") will be | |
579 * decreased in the filter output. */ | |
580 irange = 64.0 / range; // so irange*(max-value) is in the range [0, 63] | |
581 gain_mul = range * (fcb_type == FCB_TYPE_HARDCODED ? (5.0 / 13.0) : | |
582 (5.0 / 14.7)); | |
583 angle_mul = gain_mul * (8.0 * M_LN10 / M_PI); | |
584 for (n = 0; n <= 64; n++) { | |
585 float pow; | |
586 | |
587 idx = FFMAX(0, lrint((max - lpcs[n]) * irange) - 1); | |
588 pow = wmavoice_denoise_power_table[s->denoise_strength][idx]; | |
589 lpcs[n] = angle_mul * pow; | |
590 | |
591 /* 70.57 =~ 1/log10(1.0331663) */ | |
592 idx = (pow * gain_mul - 0.0295) * 70.570526123; | |
593 if (idx > 127) { // fallback if index falls outside table range | |
594 coeffs[n] = wmavoice_energy_table[127] * | |
595 powf(1.0331663, idx - 127); | |
596 } else | |
597 coeffs[n] = wmavoice_energy_table[FFMAX(0, idx)]; | |
598 } | |
599 | |
600 /* calculate the Hilbert transform of the gains, which we do (since this | |
601 * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()). | |
602 * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the | |
603 * "moment" of the LPCs in this filter. */ | |
604 ff_dct_calc(&s->dct, lpcs); | |
605 ff_dct_calc(&s->dst, lpcs); | |
606 | |
607 /* Split out the coefficient indexes into phase/magnitude pairs */ | |
608 idx = 255 + av_clip(lpcs[64], -255, 255); | |
609 coeffs[0] = coeffs[0] * s->cos[idx]; | |
610 idx = 255 + av_clip(lpcs[64] - 2 * lpcs[63], -255, 255); | |
611 last_coeff = coeffs[64] * s->cos[idx]; | |
612 for (n = 63;; n--) { | |
613 idx = 255 + av_clip(-lpcs[64] - 2 * lpcs[n - 1], -255, 255); | |
614 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; | |
615 coeffs[n * 2] = coeffs[n] * s->cos[idx]; | |
616 | |
617 if (!--n) break; | |
618 | |
619 idx = 255 + av_clip( lpcs[64] - 2 * lpcs[n - 1], -255, 255); | |
620 coeffs[n * 2 + 1] = coeffs[n] * s->sin[idx]; | |
621 coeffs[n * 2] = coeffs[n] * s->cos[idx]; | |
622 } | |
623 coeffs[1] = last_coeff; | |
624 | |
625 /* move into real domain */ | |
626 ff_rdft_calc(&s->irdft, coeffs); | |
627 | |
628 /* tilt correction and normalize scale */ | |
629 memset(&coeffs[remainder], 0, sizeof(coeffs[0]) * (128 - remainder)); | |
630 if (s->denoise_tilt_corr) { | |
631 float tilt_mem = 0; | |
632 | |
633 coeffs[remainder - 1] = 0; | |
634 ff_tilt_compensation(&tilt_mem, | |
635 -1.8 * tilt_factor(coeffs, remainder - 1), | |
636 coeffs, remainder); | |
637 } | |
638 sq = (1.0 / 64.0) * sqrtf(1 / ff_dot_productf(coeffs, coeffs, remainder)); | |
639 for (n = 0; n < remainder; n++) | |
640 coeffs[n] *= sq; | |
641 } | |
642 | |
643 /** | |
644 * This function applies a Wiener filter on the (noisy) speech signal as | |
645 * a means to denoise it. | |
646 * | |
647 * - take RDFT of LPCs to get the power spectrum of the noise + speech; | |
648 * - using this power spectrum, calculate (for each frequency) the Wiener | |
649 * filter gain, which depends on the frequency power and desired level | |
650 * of noise subtraction (when set too high, this leads to artifacts) | |
651 * We can do this symmetrically over the X-axis (so 0-4kHz is the inverse | |
652 * of 4-8kHz); | |
653 * - by doing a phase shift, calculate the Hilbert transform of this array | |
654 * of per-frequency filter-gains to get the filtering coefficients; | |
655 * - smoothen/normalize/de-tilt these filter coefficients as desired; | |
656 * - take RDFT of noisy sound, apply the coefficients and take its IRDFT | |
657 * to get the denoised speech signal; | |
658 * - the leftover (i.e. output of the IRDFT on denoised speech data beyond | |
659 * the frame boundary) are saved and applied to subsequent frames by an | |
660 * overlap-add method (otherwise you get clicking-artifacts). | |
661 * | |
662 * @param s WMA Voice decoding context | |
663 * @param s fcb_type Frame (codebook) type | |
664 * @param synth_pf input: the noisy speech signal, output: denoised speech | |
665 * data; should be 16-byte aligned (for ASM purposes) | |
666 * @param size size of the speech data | |
667 * @param lpcs LPCs used to synthesize this frame's speech data | |
668 */ | |
669 static void wiener_denoise(WMAVoiceContext *s, int fcb_type, | |
670 float *synth_pf, int size, | |
671 const float *lpcs) | |
672 { | |
673 int remainder, lim, n; | |
674 | |
675 if (fcb_type != FCB_TYPE_SILENCE) { | |
676 float *tilted_lpcs = s->tilted_lpcs_pf, | |
677 *coeffs = s->denoise_coeffs_pf, tilt_mem = 0; | |
678 | |
679 tilted_lpcs[0] = 1.0; | |
680 memcpy(&tilted_lpcs[1], lpcs, sizeof(lpcs[0]) * s->lsps); | |
681 memset(&tilted_lpcs[s->lsps + 1], 0, | |
682 sizeof(tilted_lpcs[0]) * (128 - s->lsps - 1)); | |
683 ff_tilt_compensation(&tilt_mem, 0.7 * tilt_factor(lpcs, s->lsps), | |
684 tilted_lpcs, s->lsps + 2); | |
685 | |
686 /* The IRDFT output (127 samples for 7-bit filter) beyond the frame | |
687 * size is applied to the next frame. All input beyond this is zero, | |
688 * and thus all output beyond this will go towards zero, hence we can | |
689 * limit to min(size-1, 127-size) as a performance consideration. */ | |
690 remainder = FFMIN(127 - size, size - 1); | |
691 calc_input_response(s, tilted_lpcs, fcb_type, coeffs, remainder); | |
692 | |
693 /* apply coefficients (in frequency spectrum domain), i.e. complex | |
694 * number multiplication */ | |
695 memset(&synth_pf[size], 0, sizeof(synth_pf[0]) * (128 - size)); | |
696 ff_rdft_calc(&s->rdft, synth_pf); | |
697 ff_rdft_calc(&s->rdft, coeffs); | |
698 synth_pf[0] *= coeffs[0]; | |
699 synth_pf[1] *= coeffs[1]; | |
11675 | 700 for (n = 1; n < 64; n++) { |
11653 | 701 float v1 = synth_pf[n * 2], v2 = synth_pf[n * 2 + 1]; |
702 synth_pf[n * 2] = v1 * coeffs[n * 2] - v2 * coeffs[n * 2 + 1]; | |
703 synth_pf[n * 2 + 1] = v2 * coeffs[n * 2] + v1 * coeffs[n * 2 + 1]; | |
704 } | |
705 ff_rdft_calc(&s->irdft, synth_pf); | |
706 } | |
707 | |
708 /* merge filter output with the history of previous runs */ | |
709 if (s->denoise_filter_cache_size) { | |
710 lim = FFMIN(s->denoise_filter_cache_size, size); | |
711 for (n = 0; n < lim; n++) | |
712 synth_pf[n] += s->denoise_filter_cache[n]; | |
713 s->denoise_filter_cache_size -= lim; | |
714 memmove(s->denoise_filter_cache, &s->denoise_filter_cache[size], | |
715 sizeof(s->denoise_filter_cache[0]) * s->denoise_filter_cache_size); | |
716 } | |
717 | |
718 /* move remainder of filter output into a cache for future runs */ | |
719 if (fcb_type != FCB_TYPE_SILENCE) { | |
720 lim = FFMIN(remainder, s->denoise_filter_cache_size); | |
721 for (n = 0; n < lim; n++) | |
722 s->denoise_filter_cache[n] += synth_pf[size + n]; | |
723 if (lim < remainder) { | |
724 memcpy(&s->denoise_filter_cache[lim], &synth_pf[size + lim], | |
725 sizeof(s->denoise_filter_cache[0]) * (remainder - lim)); | |
726 s->denoise_filter_cache_size = remainder; | |
727 } | |
728 } | |
729 } | |
730 | |
731 /** | |
732 * Averaging projection filter, the postfilter used in WMAVoice. | |
733 * | |
734 * This uses the following steps: | |
735 * - A zero-synthesis filter (generate excitation from synth signal) | |
736 * - Kalman smoothing on excitation, based on pitch | |
737 * - Re-synthesized smoothened output | |
738 * - Iterative Wiener denoise filter | |
739 * - Adaptive gain filter | |
740 * - DC filter | |
741 * | |
742 * @param s WMAVoice decoding context | |
743 * @param synth Speech synthesis output (before postfilter) | |
744 * @param samples Output buffer for filtered samples | |
745 * @param size Buffer size of synth & samples | |
746 * @param lpcs Generated LPCs used for speech synthesis | |
747 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) | |
748 * @param pitch Pitch of the input signal | |
749 */ | |
750 static void postfilter(WMAVoiceContext *s, const float *synth, | |
751 float *samples, int size, | |
752 const float *lpcs, float *zero_exc_pf, | |
753 int fcb_type, int pitch) | |
754 { | |
755 float synth_filter_in_buf[MAX_FRAMESIZE / 2], | |
756 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], | |
757 *synth_filter_in = zero_exc_pf; | |
758 | |
759 assert(size <= MAX_FRAMESIZE / 2); | |
760 | |
761 /* generate excitation from input signal */ | |
762 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); | |
763 | |
764 if (fcb_type >= FCB_TYPE_AW_PULSES && | |
765 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) | |
766 synth_filter_in = synth_filter_in_buf; | |
767 | |
768 /* re-synthesize speech after smoothening, and keep history */ | |
769 ff_celp_lp_synthesis_filterf(synth_pf, lpcs, | |
770 synth_filter_in, size, s->lsps); | |
771 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], | |
772 sizeof(synth_pf[0]) * s->lsps); | |
773 | |
774 wiener_denoise(s, fcb_type, synth_pf, size, lpcs); | |
775 | |
776 adaptive_gain_control(samples, synth_pf, synth, size, 0.99, | |
777 &s->postfilter_agc); | |
778 | |
779 if (s->dc_level > 8) { | |
780 /* remove ultra-low frequency DC noise / highpass filter; | |
781 * coefficients are identical to those used in SIPR decoding, | |
782 * and very closely resemble those used in AMR-NB decoding. */ | |
783 ff_acelp_apply_order_2_transfer_function(samples, samples, | |
784 (const float[2]) { -1.99997, 1.0 }, | |
785 (const float[2]) { -1.9330735188, 0.93589198496 }, | |
786 0.93980580475, s->dcf_mem, size); | |
787 } | |
788 } | |
789 /** | |
790 * @} | |
791 */ | |
792 | |
793 /** | |
11123 | 794 * Dequantize LSPs |
795 * @param lsps output pointer to the array that will hold the LSPs | |
796 * @param num number of LSPs to be dequantized | |
797 * @param values quantized values, contains n_stages values | |
798 * @param sizes range (i.e. max value) of each quantized value | |
799 * @param n_stages number of dequantization runs | |
800 * @param table dequantization table to be used | |
801 * @param mul_q LSF multiplier | |
802 * @param base_q base (lowest) LSF values | |
803 */ | |
804 static void dequant_lsps(double *lsps, int num, | |
805 const uint16_t *values, | |
806 const uint16_t *sizes, | |
807 int n_stages, const uint8_t *table, | |
808 const double *mul_q, | |
809 const double *base_q) | |
810 { | |
811 int n, m; | |
812 | |
813 memset(lsps, 0, num * sizeof(*lsps)); | |
814 for (n = 0; n < n_stages; n++) { | |
815 const uint8_t *t_off = &table[values[n] * num]; | |
816 double base = base_q[n], mul = mul_q[n]; | |
817 | |
818 for (m = 0; m < num; m++) | |
819 lsps[m] += base + mul * t_off[m]; | |
820 | |
821 table += sizes[n] * num; | |
822 } | |
823 } | |
824 | |
825 /** | |
826 * @defgroup lsp_dequant LSP dequantization routines | |
827 * LSP dequantization routines, for 10/16LSPs and independent/residual coding. | |
828 * @note we assume enough bits are available, caller should check. | |
829 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; | |
830 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. | |
831 * @{ | |
832 */ | |
833 /** | |
834 * Parse 10 independently-coded LSPs. | |
835 */ | |
836 static void dequant_lsp10i(GetBitContext *gb, double *lsps) | |
837 { | |
838 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; | |
839 static const double mul_lsf[4] = { | |
840 5.2187144800e-3, 1.4626986422e-3, | |
841 9.6179549166e-4, 1.1325736225e-3 | |
842 }; | |
843 static const double base_lsf[4] = { | |
844 M_PI * -2.15522e-1, M_PI * -6.1646e-2, | |
845 M_PI * -3.3486e-2, M_PI * -5.7408e-2 | |
846 }; | |
847 uint16_t v[4]; | |
848 | |
849 v[0] = get_bits(gb, 8); | |
850 v[1] = get_bits(gb, 6); | |
851 v[2] = get_bits(gb, 5); | |
852 v[3] = get_bits(gb, 5); | |
853 | |
854 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, | |
855 mul_lsf, base_lsf); | |
856 } | |
857 | |
858 /** | |
859 * Parse 10 independently-coded LSPs, and then derive the tables to | |
860 * generate LSPs for the other frames from them (residual coding). | |
861 */ | |
862 static void dequant_lsp10r(GetBitContext *gb, | |
863 double *i_lsps, const double *old, | |
864 double *a1, double *a2, int q_mode) | |
865 { | |
866 static const uint16_t vec_sizes[3] = { 128, 64, 64 }; | |
867 static const double mul_lsf[3] = { | |
868 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 | |
869 }; | |
870 static const double base_lsf[3] = { | |
871 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 | |
872 }; | |
873 const float (*ipol_tab)[2][10] = q_mode ? | |
874 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; | |
875 uint16_t interpol, v[3]; | |
876 int n; | |
877 | |
878 dequant_lsp10i(gb, i_lsps); | |
879 | |
880 interpol = get_bits(gb, 5); | |
881 v[0] = get_bits(gb, 7); | |
882 v[1] = get_bits(gb, 6); | |
883 v[2] = get_bits(gb, 6); | |
884 | |
885 for (n = 0; n < 10; n++) { | |
886 double delta = old[n] - i_lsps[n]; | |
887 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
888 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
889 } | |
890 | |
891 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, | |
892 mul_lsf, base_lsf); | |
893 } | |
894 | |
895 /** | |
896 * Parse 16 independently-coded LSPs. | |
897 */ | |
898 static void dequant_lsp16i(GetBitContext *gb, double *lsps) | |
899 { | |
900 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; | |
901 static const double mul_lsf[5] = { | |
902 3.3439586280e-3, 6.9908173703e-4, | |
903 3.3216608306e-3, 1.0334960326e-3, | |
904 3.1899104283e-3 | |
905 }; | |
906 static const double base_lsf[5] = { | |
907 M_PI * -1.27576e-1, M_PI * -2.4292e-2, | |
908 M_PI * -1.28094e-1, M_PI * -3.2128e-2, | |
909 M_PI * -1.29816e-1 | |
910 }; | |
911 uint16_t v[5]; | |
912 | |
913 v[0] = get_bits(gb, 8); | |
914 v[1] = get_bits(gb, 6); | |
915 v[2] = get_bits(gb, 7); | |
916 v[3] = get_bits(gb, 6); | |
917 v[4] = get_bits(gb, 7); | |
918 | |
919 dequant_lsps( lsps, 5, v, vec_sizes, 2, | |
920 wmavoice_dq_lsp16i1, mul_lsf, base_lsf); | |
921 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, | |
922 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); | |
923 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, | |
924 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); | |
925 } | |
926 | |
927 /** | |
928 * Parse 16 independently-coded LSPs, and then derive the tables to | |
929 * generate LSPs for the other frames from them (residual coding). | |
930 */ | |
931 static void dequant_lsp16r(GetBitContext *gb, | |
932 double *i_lsps, const double *old, | |
933 double *a1, double *a2, int q_mode) | |
934 { | |
935 static const uint16_t vec_sizes[3] = { 128, 128, 128 }; | |
936 static const double mul_lsf[3] = { | |
937 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 | |
938 }; | |
939 static const double base_lsf[3] = { | |
940 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 | |
941 }; | |
942 const float (*ipol_tab)[2][16] = q_mode ? | |
943 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; | |
944 uint16_t interpol, v[3]; | |
945 int n; | |
946 | |
947 dequant_lsp16i(gb, i_lsps); | |
948 | |
949 interpol = get_bits(gb, 5); | |
950 v[0] = get_bits(gb, 7); | |
951 v[1] = get_bits(gb, 7); | |
952 v[2] = get_bits(gb, 7); | |
953 | |
954 for (n = 0; n < 16; n++) { | |
955 double delta = old[n] - i_lsps[n]; | |
956 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
957 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
958 } | |
959 | |
960 dequant_lsps( a2, 10, v, vec_sizes, 1, | |
961 wmavoice_dq_lsp16r1, mul_lsf, base_lsf); | |
962 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, | |
963 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); | |
964 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, | |
965 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); | |
966 } | |
967 | |
968 /** | |
969 * @} | |
970 * @defgroup aw Pitch-adaptive window coding functions | |
971 * The next few functions are for pitch-adaptive window coding. | |
972 * @{ | |
973 */ | |
974 /** | |
975 * Parse the offset of the first pitch-adaptive window pulses, and | |
976 * the distribution of pulses between the two blocks in this frame. | |
977 * @param s WMA Voice decoding context private data | |
978 * @param gb bit I/O context | |
979 * @param pitch pitch for each block in this frame | |
980 */ | |
981 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, | |
982 const int *pitch) | |
983 { | |
984 static const int16_t start_offset[94] = { | |
985 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, | |
986 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, | |
987 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, | |
988 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, | |
989 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, | |
990 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, | |
991 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, | |
992 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 | |
993 }; | |
994 int bits, offset; | |
995 | |
996 /* position of pulse */ | |
997 s->aw_idx_is_ext = 0; | |
998 if ((bits = get_bits(gb, 6)) >= 54) { | |
999 s->aw_idx_is_ext = 1; | |
1000 bits += (bits - 54) * 3 + get_bits(gb, 2); | |
1001 } | |
1002 | |
1003 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count | |
1004 * the distribution of the pulses in each block contained in this frame. */ | |
1005 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; | |
1006 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; | |
1007 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; | |
1008 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; | |
1009 offset += s->aw_n_pulses[0] * pitch[0]; | |
1010 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; | |
1011 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; | |
1012 | |
1013 /* if continuing from a position before the block, reset position to | |
1014 * start of block (when corrected for the range over which it can be | |
1015 * spread in aw_pulse_set1()). */ | |
1016 if (start_offset[bits] < MAX_FRAMESIZE / 2) { | |
1017 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) | |
1018 s->aw_first_pulse_off[1] -= pitch[1]; | |
1019 if (start_offset[bits] < 0) | |
1020 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) | |
1021 s->aw_first_pulse_off[0] -= pitch[0]; | |
1022 } | |
1023 } | |
1024 | |
1025 /** | |
1026 * Apply second set of pitch-adaptive window pulses. | |
1027 * @param s WMA Voice decoding context private data | |
1028 * @param gb bit I/O context | |
1029 * @param block_idx block index in frame [0, 1] | |
1030 * @param fcb structure containing fixed codebook vector info | |
1031 */ | |
1032 static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, | |
1033 int block_idx, AMRFixed *fcb) | |
1034 { | |
1035 uint16_t use_mask[7]; // only 5 are used, rest is padding | |
1036 /* in this function, idx is the index in the 80-bit (+ padding) use_mask | |
1037 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits | |
1038 * of idx are the position of the bit within a particular item in the | |
1039 * array (0 being the most significant bit, and 15 being the least | |
1040 * significant bit), and the remainder (>> 4) is the index in the | |
1041 * use_mask[]-array. This is faster and uses less memory than using a | |
1042 * 80-byte/80-int array. */ | |
1043 int pulse_off = s->aw_first_pulse_off[block_idx], | |
1044 pulse_start, n, idx, range, aidx, start_off = 0; | |
1045 | |
1046 /* set offset of first pulse to within this block */ | |
1047 if (s->aw_n_pulses[block_idx] > 0) | |
1048 while (pulse_off + s->aw_pulse_range < 1) | |
1049 pulse_off += fcb->pitch_lag; | |
1050 | |
1051 /* find range per pulse */ | |
1052 if (s->aw_n_pulses[0] > 0) { | |
1053 if (block_idx == 0) { | |
1054 range = 32; | |
1055 } else /* block_idx = 1 */ { | |
1056 range = 8; | |
1057 if (s->aw_n_pulses[block_idx] > 0) | |
1058 pulse_off = s->aw_next_pulse_off_cache; | |
1059 } | |
1060 } else | |
1061 range = 16; | |
1062 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; | |
1063 | |
1064 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, | |
1065 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus | |
1066 * we exclude that range from being pulsed again in this function. */ | |
1067 memset( use_mask, -1, 5 * sizeof(use_mask[0])); | |
1068 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); | |
1069 if (s->aw_n_pulses[block_idx] > 0) | |
1070 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { | |
1071 int excl_range = s->aw_pulse_range; // always 16 or 24 | |
1072 uint16_t *use_mask_ptr = &use_mask[idx >> 4]; | |
1073 int first_sh = 16 - (idx & 15); | |
1074 *use_mask_ptr++ &= 0xFFFF << first_sh; | |
1075 excl_range -= first_sh; | |
1076 if (excl_range >= 16) { | |
1077 *use_mask_ptr++ = 0; | |
1078 *use_mask_ptr &= 0xFFFF >> (excl_range - 16); | |
1079 } else | |
1080 *use_mask_ptr &= 0xFFFF >> excl_range; | |
1081 } | |
1082 | |
1083 /* find the 'aidx'th offset that is not excluded */ | |
1084 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); | |
1085 for (n = 0; n <= aidx; pulse_start++) { | |
1086 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; | |
1087 if (idx >= MAX_FRAMESIZE / 2) { // find from zero | |
1088 if (use_mask[0]) idx = 0x0F; | |
1089 else if (use_mask[1]) idx = 0x1F; | |
1090 else if (use_mask[2]) idx = 0x2F; | |
1091 else if (use_mask[3]) idx = 0x3F; | |
1092 else if (use_mask[4]) idx = 0x4F; | |
1093 else return; | |
1094 idx -= av_log2_16bit(use_mask[idx >> 4]); | |
1095 } | |
1096 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { | |
1097 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); | |
1098 n++; | |
1099 start_off = idx; | |
1100 } | |
1101 } | |
1102 | |
1103 fcb->x[fcb->n] = start_off; | |
1104 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; | |
1105 fcb->n++; | |
1106 | |
1107 /* set offset for next block, relative to start of that block */ | |
1108 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; | |
1109 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; | |
1110 } | |
1111 | |
1112 /** | |
1113 * Apply first set of pitch-adaptive window pulses. | |
1114 * @param s WMA Voice decoding context private data | |
1115 * @param gb bit I/O context | |
1116 * @param block_idx block index in frame [0, 1] | |
1117 * @param fcb storage location for fixed codebook pulse info | |
1118 */ | |
1119 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, | |
1120 int block_idx, AMRFixed *fcb) | |
1121 { | |
1122 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); | |
1123 float v; | |
1124 | |
1125 if (s->aw_n_pulses[block_idx] > 0) { | |
1126 int n, v_mask, i_mask, sh, n_pulses; | |
1127 | |
1128 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each | |
1129 n_pulses = 3; | |
1130 v_mask = 8; | |
1131 i_mask = 7; | |
1132 sh = 4; | |
1133 } else { // 4 pulses, 1:sign + 2:index each | |
1134 n_pulses = 4; | |
1135 v_mask = 4; | |
1136 i_mask = 3; | |
1137 sh = 3; | |
1138 } | |
1139 | |
1140 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { | |
1141 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; | |
1142 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + | |
1143 s->aw_first_pulse_off[block_idx]; | |
1144 while (fcb->x[fcb->n] < 0) | |
1145 fcb->x[fcb->n] += fcb->pitch_lag; | |
1146 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) | |
1147 fcb->n++; | |
1148 } | |
1149 } else { | |
1150 int num2 = (val & 0x1FF) >> 1, delta, idx; | |
1151 | |
1152 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } | |
1153 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } | |
1154 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } | |
1155 else { delta = 7; idx = num2 + 1 - 3 * 75; } | |
1156 v = (val & 0x200) ? -1.0 : 1.0; | |
1157 | |
1158 fcb->no_repeat_mask |= 3 << fcb->n; | |
1159 fcb->x[fcb->n] = idx - delta; | |
1160 fcb->y[fcb->n] = v; | |
1161 fcb->x[fcb->n + 1] = idx; | |
1162 fcb->y[fcb->n + 1] = (val & 1) ? -v : v; | |
1163 fcb->n += 2; | |
1164 } | |
1165 } | |
1166 | |
1167 /** | |
1168 * @} | |
1169 * | |
1170 * Generate a random number from frame_cntr and block_idx, which will lief | |
1171 * in the range [0, 1000 - block_size] (so it can be used as an index in a | |
1172 * table of size 1000 of which you want to read block_size entries). | |
1173 * | |
1174 * @param frame_cntr current frame number | |
1175 * @param block_num current block index | |
1176 * @param block_size amount of entries we want to read from a table | |
1177 * that has 1000 entries | |
11556 | 1178 * @return a (non-)random number in the [0, 1000 - block_size] range. |
11123 | 1179 */ |
1180 static int pRNG(int frame_cntr, int block_num, int block_size) | |
1181 { | |
1182 /* array to simplify the calculation of z: | |
1183 * y = (x % 9) * 5 + 6; | |
1184 * z = (49995 * x) / y; | |
1185 * Since y only has 9 values, we can remove the division by using a | |
1186 * LUT and using FASTDIV-style divisions. For each of the 9 values | |
1187 * of y, we can rewrite z as: | |
1188 * z = x * (49995 / y) + x * ((49995 % y) / y) | |
1189 * In this table, each col represents one possible value of y, the | |
1190 * first number is 49995 / y, and the second is the FASTDIV variant | |
1191 * of 49995 % y / y. */ | |
1192 static const unsigned int div_tbl[9][2] = { | |
1193 { 8332, 3 * 715827883U }, // y = 6 | |
1194 { 4545, 0 * 390451573U }, // y = 11 | |
1195 { 3124, 11 * 268435456U }, // y = 16 | |
1196 { 2380, 15 * 204522253U }, // y = 21 | |
1197 { 1922, 23 * 165191050U }, // y = 26 | |
1198 { 1612, 23 * 138547333U }, // y = 31 | |
1199 { 1388, 27 * 119304648U }, // y = 36 | |
1200 { 1219, 16 * 104755300U }, // y = 41 | |
1201 { 1086, 39 * 93368855U } // y = 46 | |
1202 }; | |
1203 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; | |
1204 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, | |
1205 // so this is effectively a modulo (%) | |
1206 y = x - 9 * MULH(477218589, x); // x % 9 | |
1207 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); | |
1208 // z = x * 49995 / (y * 5 + 6) | |
1209 return z % (1000 - block_size); | |
1210 } | |
1211 | |
1212 /** | |
1213 * Parse hardcoded signal for a single block. | |
1214 * @note see #synth_block(). | |
1215 */ | |
1216 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, | |
1217 int block_idx, int size, | |
1218 const struct frame_type_desc *frame_desc, | |
1219 float *excitation) | |
1220 { | |
1221 float gain; | |
1222 int n, r_idx; | |
1223 | |
1224 assert(size <= MAX_FRAMESIZE); | |
1225 | |
1226 /* Set the offset from which we start reading wmavoice_std_codebook */ | |
1227 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1228 r_idx = pRNG(s->frame_cntr, block_idx, size); | |
1229 gain = s->silence_gain; | |
1230 } else /* FCB_TYPE_HARDCODED */ { | |
1231 r_idx = get_bits(gb, 8); | |
1232 gain = wmavoice_gain_universal[get_bits(gb, 6)]; | |
1233 } | |
1234 | |
1235 /* Clear gain prediction parameters */ | |
1236 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); | |
1237 | |
1238 /* Apply gain to hardcoded codebook and use that as excitation signal */ | |
1239 for (n = 0; n < size; n++) | |
1240 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; | |
1241 } | |
1242 | |
1243 /** | |
1244 * Parse FCB/ACB signal for a single block. | |
1245 * @note see #synth_block(). | |
1246 */ | |
1247 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, | |
1248 int block_idx, int size, | |
1249 int block_pitch_sh2, | |
1250 const struct frame_type_desc *frame_desc, | |
1251 float *excitation) | |
1252 { | |
1253 static const float gain_coeff[6] = { | |
1254 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 | |
1255 }; | |
1256 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; | |
1257 int n, idx, gain_weight; | |
1258 AMRFixed fcb; | |
1259 | |
1260 assert(size <= MAX_FRAMESIZE / 2); | |
1261 memset(pulses, 0, sizeof(*pulses) * size); | |
1262 | |
1263 fcb.pitch_lag = block_pitch_sh2 >> 2; | |
1264 fcb.pitch_fac = 1.0; | |
1265 fcb.no_repeat_mask = 0; | |
1266 fcb.n = 0; | |
1267 | |
1268 /* For the other frame types, this is where we apply the innovation | |
1269 * (fixed) codebook pulses of the speech signal. */ | |
1270 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1271 aw_pulse_set1(s, gb, block_idx, &fcb); | |
1272 aw_pulse_set2(s, gb, block_idx, &fcb); | |
1273 } else /* FCB_TYPE_EXC_PULSES */ { | |
1274 int offset_nbits = 5 - frame_desc->log_n_blocks; | |
1275 | |
1276 fcb.no_repeat_mask = -1; | |
1277 /* similar to ff_decode_10_pulses_35bits(), but with single pulses | |
1278 * (instead of double) for a subset of pulses */ | |
1279 for (n = 0; n < 5; n++) { | |
1280 float sign; | |
1281 int pos1, pos2; | |
1282 | |
1283 sign = get_bits1(gb) ? 1.0 : -1.0; | |
1284 pos1 = get_bits(gb, offset_nbits); | |
1285 fcb.x[fcb.n] = n + 5 * pos1; | |
1286 fcb.y[fcb.n++] = sign; | |
1287 if (n < frame_desc->dbl_pulses) { | |
1288 pos2 = get_bits(gb, offset_nbits); | |
1289 fcb.x[fcb.n] = n + 5 * pos2; | |
1290 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; | |
1291 } | |
1292 } | |
1293 } | |
1294 ff_set_fixed_vector(pulses, &fcb, 1.0, size); | |
1295 | |
1296 /* Calculate gain for adaptive & fixed codebook signal. | |
1297 * see ff_amr_set_fixed_gain(). */ | |
1298 idx = get_bits(gb, 7); | |
1299 fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) - | |
1300 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); | |
1301 acb_gain = wmavoice_gain_codebook_acb[idx]; | |
1302 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], | |
1303 -2.9957322736 /* log(0.05) */, | |
1304 1.6094379124 /* log(5.0) */); | |
1305 | |
1306 gain_weight = 8 >> frame_desc->log_n_blocks; | |
1307 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, | |
1308 sizeof(*s->gain_pred_err) * (6 - gain_weight)); | |
1309 for (n = 0; n < gain_weight; n++) | |
1310 s->gain_pred_err[n] = pred_err; | |
1311 | |
1312 /* Calculation of adaptive codebook */ | |
1313 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1314 int len; | |
1315 for (n = 0; n < size; n += len) { | |
1316 int next_idx_sh16; | |
1317 int abs_idx = block_idx * size + n; | |
1318 int pitch_sh16 = (s->last_pitch_val << 16) + | |
1319 s->pitch_diff_sh16 * abs_idx; | |
1320 int pitch = (pitch_sh16 + 0x6FFF) >> 16; | |
1321 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; | |
1322 idx = idx_sh16 >> 16; | |
1323 if (s->pitch_diff_sh16) { | |
1324 if (s->pitch_diff_sh16 > 0) { | |
1325 next_idx_sh16 = (idx_sh16) &~ 0xFFFF; | |
1326 } else | |
1327 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; | |
1328 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, | |
1329 1, size - n); | |
1330 } else | |
1331 len = size; | |
1332 | |
1333 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], | |
1334 wmavoice_ipol1_coeffs, 17, | |
1335 idx, 9, len); | |
1336 } | |
1337 } else /* ACB_TYPE_HAMMING */ { | |
1338 int block_pitch = block_pitch_sh2 >> 2; | |
1339 idx = block_pitch_sh2 & 3; | |
1340 if (idx) { | |
1341 ff_acelp_interpolatef(excitation, &excitation[-block_pitch], | |
1342 wmavoice_ipol2_coeffs, 4, | |
1343 idx, 8, size); | |
1344 } else | |
12021 | 1345 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, |
11123 | 1346 sizeof(float) * size); |
1347 } | |
1348 | |
1349 /* Interpolate ACB/FCB and use as excitation signal */ | |
1350 ff_weighted_vector_sumf(excitation, excitation, pulses, | |
1351 acb_gain, fcb_gain, size); | |
1352 } | |
1353 | |
1354 /** | |
1355 * Parse data in a single block. | |
1356 * @note we assume enough bits are available, caller should check. | |
1357 * | |
1358 * @param s WMA Voice decoding context private data | |
1359 * @param gb bit I/O context | |
1360 * @param block_idx index of the to-be-read block | |
1361 * @param size amount of samples to be read in this block | |
1362 * @param block_pitch_sh2 pitch for this block << 2 | |
1363 * @param lsps LSPs for (the end of) this frame | |
1364 * @param prev_lsps LSPs for the last frame | |
1365 * @param frame_desc frame type descriptor | |
1366 * @param excitation target memory for the ACB+FCB interpolated signal | |
1367 * @param synth target memory for the speech synthesis filter output | |
1368 * @return 0 on success, <0 on error. | |
1369 */ | |
1370 static void synth_block(WMAVoiceContext *s, GetBitContext *gb, | |
1371 int block_idx, int size, | |
1372 int block_pitch_sh2, | |
1373 const double *lsps, const double *prev_lsps, | |
1374 const struct frame_type_desc *frame_desc, | |
1375 float *excitation, float *synth) | |
1376 { | |
1377 double i_lsps[MAX_LSPS]; | |
1378 float lpcs[MAX_LSPS]; | |
1379 float fac; | |
1380 int n; | |
1381 | |
1382 if (frame_desc->acb_type == ACB_TYPE_NONE) | |
1383 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); | |
1384 else | |
1385 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, | |
1386 frame_desc, excitation); | |
1387 | |
1388 /* convert interpolated LSPs to LPCs */ | |
1389 fac = (block_idx + 0.5) / frame_desc->n_blocks; | |
1390 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1391 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); | |
1392 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1393 | |
1394 /* Speech synthesis */ | |
1395 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); | |
1396 } | |
1397 | |
1398 /** | |
1399 * Synthesize output samples for a single frame. | |
1400 * @note we assume enough bits are available, caller should check. | |
1401 * | |
1402 * @param ctx WMA Voice decoder context | |
1403 * @param gb bit I/O context (s->gb or one for cross-packet superframes) | |
11653 | 1404 * @param frame_idx Frame number within superframe [0-2] |
11123 | 1405 * @param samples pointer to output sample buffer, has space for at least 160 |
1406 * samples | |
1407 * @param lsps LSP array | |
1408 * @param prev_lsps array of previous frame's LSPs | |
1409 * @param excitation target buffer for excitation signal | |
1410 * @param synth target buffer for synthesized speech data | |
1411 * @return 0 on success, <0 on error. | |
1412 */ | |
11653 | 1413 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
11123 | 1414 float *samples, |
1415 const double *lsps, const double *prev_lsps, | |
1416 float *excitation, float *synth) | |
1417 { | |
1418 WMAVoiceContext *s = ctx->priv_data; | |
1419 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val; | |
1420 int pitch[MAX_BLOCKS], last_block_pitch; | |
1421 | |
1422 /* Parse frame type ("frame header"), see frame_descs */ | |
1423 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], | |
1424 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; | |
1425 | |
1426 if (bd_idx < 0) { | |
1427 av_log(ctx, AV_LOG_ERROR, | |
1428 "Invalid frame type VLC code, skipping\n"); | |
1429 return -1; | |
1430 } | |
1431 | |
1432 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ | |
1433 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { | |
1434 /* Pitch is provided per frame, which is interpreted as the pitch of | |
1435 * the last sample of the last block of this frame. We can interpolate | |
1436 * the pitch of other blocks (and even pitch-per-sample) by gradually | |
1437 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ | |
1438 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; | |
1439 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; | |
1440 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); | |
1441 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); | |
1442 if (s->last_acb_type == ACB_TYPE_NONE || | |
1443 20 * abs(cur_pitch_val - s->last_pitch_val) > | |
1444 (cur_pitch_val + s->last_pitch_val)) | |
1445 s->last_pitch_val = cur_pitch_val; | |
1446 | |
1447 /* pitch per block */ | |
1448 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1449 int fac = n * 2 + 1; | |
1450 | |
1451 pitch[n] = (MUL16(fac, cur_pitch_val) + | |
1452 MUL16((n_blocks_x2 - fac), s->last_pitch_val) + | |
1453 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; | |
1454 } | |
1455 | |
1456 /* "pitch-diff-per-sample" for calculation of pitch per sample */ | |
1457 s->pitch_diff_sh16 = | |
1458 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; | |
1459 } | |
1460 | |
1461 /* Global gain (if silence) and pitch-adaptive window coordinates */ | |
1462 switch (frame_descs[bd_idx].fcb_type) { | |
1463 case FCB_TYPE_SILENCE: | |
1464 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; | |
1465 break; | |
1466 case FCB_TYPE_AW_PULSES: | |
1467 aw_parse_coords(s, gb, pitch); | |
1468 break; | |
1469 } | |
1470 | |
1471 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1472 int bl_pitch_sh2; | |
1473 | |
1474 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ | |
1475 switch (frame_descs[bd_idx].acb_type) { | |
1476 case ACB_TYPE_HAMMING: { | |
1477 /* Pitch is given per block. Per-block pitches are encoded as an | |
1478 * absolute value for the first block, and then delta values | |
1479 * relative to this value) for all subsequent blocks. The scale of | |
1480 * this pitch value is semi-logaritmic compared to its use in the | |
1481 * decoder, so we convert it to normal scale also. */ | |
1482 int block_pitch, | |
1483 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, | |
1484 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, | |
1485 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; | |
1486 | |
1487 if (n == 0) { | |
1488 block_pitch = get_bits(gb, s->block_pitch_nbits); | |
1489 } else | |
1490 block_pitch = last_block_pitch - s->block_delta_pitch_hrange + | |
1491 get_bits(gb, s->block_delta_pitch_nbits); | |
1492 /* Convert last_ so that any next delta is within _range */ | |
1493 last_block_pitch = av_clip(block_pitch, | |
1494 s->block_delta_pitch_hrange, | |
1495 s->block_pitch_range - | |
1496 s->block_delta_pitch_hrange); | |
1497 | |
1498 /* Convert semi-log-style scale back to normal scale */ | |
1499 if (block_pitch < t1) { | |
1500 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; | |
1501 } else { | |
1502 block_pitch -= t1; | |
1503 if (block_pitch < t2) { | |
1504 bl_pitch_sh2 = | |
1505 (s->block_conv_table[1] << 2) + (block_pitch << 1); | |
1506 } else { | |
1507 block_pitch -= t2; | |
1508 if (block_pitch < t3) { | |
1509 bl_pitch_sh2 = | |
1510 (s->block_conv_table[2] + block_pitch) << 2; | |
1511 } else | |
1512 bl_pitch_sh2 = s->block_conv_table[3] << 2; | |
1513 } | |
1514 } | |
1515 pitch[n] = bl_pitch_sh2 >> 2; | |
1516 break; | |
1517 } | |
1518 | |
1519 case ACB_TYPE_ASYMMETRIC: { | |
1520 bl_pitch_sh2 = pitch[n] << 2; | |
1521 break; | |
1522 } | |
1523 | |
1524 default: // ACB_TYPE_NONE has no pitch | |
1525 bl_pitch_sh2 = 0; | |
1526 break; | |
1527 } | |
1528 | |
1529 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, | |
1530 lsps, prev_lsps, &frame_descs[bd_idx], | |
1531 &excitation[n * block_nsamples], | |
1532 &synth[n * block_nsamples]); | |
1533 } | |
1534 | |
1535 /* Averaging projection filter, if applicable. Else, just copy samples | |
1536 * from synthesis buffer */ | |
1537 if (s->do_apf) { | |
11653 | 1538 double i_lsps[MAX_LSPS]; |
1539 float lpcs[MAX_LSPS]; | |
1540 | |
1541 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1542 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); | |
1543 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1544 postfilter(s, synth, samples, 80, lpcs, | |
1545 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], | |
1546 frame_descs[bd_idx].fcb_type, pitch[0]); | |
1547 | |
1548 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1549 i_lsps[n] = cos(lsps[n]); | |
1550 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1551 postfilter(s, &synth[80], &samples[80], 80, lpcs, | |
1552 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], | |
1553 frame_descs[bd_idx].fcb_type, pitch[0]); | |
1554 } else | |
11652
8b6f3d3b55cb
Move clipping of audio samples (for those codecs outputting float) from decoder
rbultje
parents:
11644
diff
changeset
|
1555 memcpy(samples, synth, 160 * sizeof(synth[0])); |
11123 | 1556 |
1557 /* Cache values for next frame */ | |
1558 s->frame_cntr++; | |
1559 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) | |
1560 s->last_acb_type = frame_descs[bd_idx].acb_type; | |
1561 switch (frame_descs[bd_idx].acb_type) { | |
1562 case ACB_TYPE_NONE: | |
1563 s->last_pitch_val = 0; | |
1564 break; | |
1565 case ACB_TYPE_ASYMMETRIC: | |
1566 s->last_pitch_val = cur_pitch_val; | |
1567 break; | |
1568 case ACB_TYPE_HAMMING: | |
1569 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; | |
1570 break; | |
1571 } | |
1572 | |
1573 return 0; | |
1574 } | |
1575 | |
1576 /** | |
1577 * Ensure minimum value for first item, maximum value for last value, | |
1578 * proper spacing between each value and proper ordering. | |
1579 * | |
1580 * @param lsps array of LSPs | |
1581 * @param num size of LSP array | |
1582 * | |
1583 * @note basically a double version of #ff_acelp_reorder_lsf(), might be | |
1584 * useful to put in a generic location later on. Parts are also | |
1585 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), | |
1586 * which is in float. | |
1587 */ | |
1588 static void stabilize_lsps(double *lsps, int num) | |
1589 { | |
1590 int n, m, l; | |
1591 | |
1592 /* set minimum value for first, maximum value for last and minimum | |
1593 * spacing between LSF values. | |
1594 * Very similar to ff_set_min_dist_lsf(), but in double. */ | |
1595 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); | |
1596 for (n = 1; n < num; n++) | |
1597 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); | |
1598 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); | |
1599 | |
1600 /* reorder (looks like one-time / non-recursed bubblesort). | |
1601 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ | |
1602 for (n = 1; n < num; n++) { | |
1603 if (lsps[n] < lsps[n - 1]) { | |
1604 for (m = 1; m < num; m++) { | |
1605 double tmp = lsps[m]; | |
1606 for (l = m - 1; l >= 0; l--) { | |
1607 if (lsps[l] <= tmp) break; | |
1608 lsps[l + 1] = lsps[l]; | |
1609 } | |
1610 lsps[l + 1] = tmp; | |
1611 } | |
1612 break; | |
1613 } | |
1614 } | |
1615 } | |
1616 | |
1617 /** | |
1618 * Test if there's enough bits to read 1 superframe. | |
1619 * | |
1620 * @param orig_gb bit I/O context used for reading. This function | |
1621 * does not modify the state of the bitreader; it | |
1622 * only uses it to copy the current stream position | |
1623 * @param s WMA Voice decoding context private data | |
11556 | 1624 * @return -1 if unsupported, 1 on not enough bits or 0 if OK. |
11123 | 1625 */ |
1626 static int check_bits_for_superframe(GetBitContext *orig_gb, | |
1627 WMAVoiceContext *s) | |
1628 { | |
1629 GetBitContext s_gb, *gb = &s_gb; | |
1630 int n, need_bits, bd_idx; | |
1631 const struct frame_type_desc *frame_desc; | |
1632 | |
1633 /* initialize a copy */ | |
1634 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); | |
1635 skip_bits_long(gb, get_bits_count(orig_gb)); | |
1636 assert(get_bits_left(gb) == get_bits_left(orig_gb)); | |
1637 | |
1638 /* superframe header */ | |
1639 if (get_bits_left(gb) < 14) | |
1640 return 1; | |
1641 if (!get_bits1(gb)) | |
1642 return -1; // WMAPro-in-WMAVoice superframe | |
1643 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe | |
1644 if (s->has_residual_lsps) { // residual LSPs (for all frames) | |
1645 if (get_bits_left(gb) < s->sframe_lsp_bitsize) | |
1646 return 1; | |
1647 skip_bits_long(gb, s->sframe_lsp_bitsize); | |
1648 } | |
1649 | |
1650 /* frames */ | |
1651 for (n = 0; n < MAX_FRAMES; n++) { | |
1652 int aw_idx_is_ext = 0; | |
1653 | |
1654 if (!s->has_residual_lsps) { // independent LSPs (per-frame) | |
1655 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; | |
1656 skip_bits_long(gb, s->frame_lsp_bitsize); | |
1657 } | |
1658 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; | |
1659 if (bd_idx < 0) | |
1660 return -1; // invalid frame type VLC code | |
1661 frame_desc = &frame_descs[bd_idx]; | |
1662 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1663 if (get_bits_left(gb) < s->pitch_nbits) | |
1664 return 1; | |
1665 skip_bits_long(gb, s->pitch_nbits); | |
1666 } | |
1667 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1668 skip_bits(gb, 8); | |
1669 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1670 int tmp = get_bits(gb, 6); | |
1671 if (tmp >= 0x36) { | |
1672 skip_bits(gb, 2); | |
1673 aw_idx_is_ext = 1; | |
1674 } | |
1675 } | |
1676 | |
1677 /* blocks */ | |
1678 if (frame_desc->acb_type == ACB_TYPE_HAMMING) { | |
1679 need_bits = s->block_pitch_nbits + | |
1680 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; | |
1681 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1682 need_bits = 2 * !aw_idx_is_ext; | |
1683 } else | |
1684 need_bits = 0; | |
1685 need_bits += frame_desc->frame_size; | |
1686 if (get_bits_left(gb) < need_bits) | |
1687 return 1; | |
1688 skip_bits_long(gb, need_bits); | |
1689 } | |
1690 | |
1691 return 0; | |
1692 } | |
1693 | |
1694 /** | |
1695 * Synthesize output samples for a single superframe. If we have any data | |
1696 * cached in s->sframe_cache, that will be used instead of whatever is loaded | |
1697 * in s->gb. | |
1698 * | |
1699 * WMA Voice superframes contain 3 frames, each containing 160 audio samples, | |
1700 * to give a total of 480 samples per frame. See #synth_frame() for frame | |
1701 * parsing. In addition to 3 frames, superframes can also contain the LSPs | |
1702 * (if these are globally specified for all frames (residually); they can | |
1703 * also be specified individually per-frame. See the s->has_residual_lsps | |
1704 * option), and can specify the number of samples encoded in this superframe | |
1705 * (if less than 480), usually used to prevent blanks at track boundaries. | |
1706 * | |
1707 * @param ctx WMA Voice decoder context | |
1708 * @param samples pointer to output buffer for voice samples | |
1709 * @param data_size pointer containing the size of #samples on input, and the | |
1710 * amount of #samples filled on output | |
1711 * @return 0 on success, <0 on error or 1 if there was not enough data to | |
1712 * fully parse the superframe | |
1713 */ | |
1714 static int synth_superframe(AVCodecContext *ctx, | |
1715 float *samples, int *data_size) | |
1716 { | |
1717 WMAVoiceContext *s = ctx->priv_data; | |
1718 GetBitContext *gb = &s->gb, s_gb; | |
1719 int n, res, n_samples = 480; | |
1720 double lsps[MAX_FRAMES][MAX_LSPS]; | |
1721 const double *mean_lsf = s->lsps == 16 ? | |
1722 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; | |
1723 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; | |
1724 float synth[MAX_LSPS + MAX_SFRAMESIZE]; | |
1725 | |
1726 memcpy(synth, s->synth_history, | |
1727 s->lsps * sizeof(*synth)); | |
1728 memcpy(excitation, s->excitation_history, | |
1729 s->history_nsamples * sizeof(*excitation)); | |
1730 | |
1731 if (s->sframe_cache_size > 0) { | |
1732 gb = &s_gb; | |
1733 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); | |
1734 s->sframe_cache_size = 0; | |
1735 } | |
1736 | |
1737 if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; | |
1738 | |
1739 /* First bit is speech/music bit, it differentiates between WMAVoice | |
1740 * speech samples (the actual codec) and WMAVoice music samples, which | |
1741 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in | |
1742 * the wild yet. */ | |
1743 if (!get_bits1(gb)) { | |
1744 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1); | |
1745 return -1; | |
1746 } | |
1747 | |
1748 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ | |
1749 if (get_bits1(gb)) { | |
1750 if ((n_samples = get_bits(gb, 12)) > 480) { | |
1751 av_log(ctx, AV_LOG_ERROR, | |
1752 "Superframe encodes >480 samples (%d), not allowed\n", | |
1753 n_samples); | |
1754 return -1; | |
1755 } | |
1756 } | |
1757 /* Parse LSPs, if global for the superframe (can also be per-frame). */ | |
1758 if (s->has_residual_lsps) { | |
1759 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; | |
1760 | |
1761 for (n = 0; n < s->lsps; n++) | |
1762 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; | |
1763 | |
1764 if (s->lsps == 10) { | |
1765 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1766 } else /* s->lsps == 16 */ | |
1767 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1768 | |
1769 for (n = 0; n < s->lsps; n++) { | |
1770 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); | |
1771 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); | |
1772 lsps[2][n] += mean_lsf[n]; | |
1773 } | |
1774 for (n = 0; n < 3; n++) | |
1775 stabilize_lsps(lsps[n], s->lsps); | |
1776 } | |
1777 | |
1778 /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ | |
1779 for (n = 0; n < 3; n++) { | |
1780 if (!s->has_residual_lsps) { | |
1781 int m; | |
1782 | |
1783 if (s->lsps == 10) { | |
1784 dequant_lsp10i(gb, lsps[n]); | |
1785 } else /* s->lsps == 16 */ | |
1786 dequant_lsp16i(gb, lsps[n]); | |
1787 | |
1788 for (m = 0; m < s->lsps; m++) | |
1789 lsps[n][m] += mean_lsf[m]; | |
1790 stabilize_lsps(lsps[n], s->lsps); | |
1791 } | |
1792 | |
11653 | 1793 if ((res = synth_frame(ctx, gb, n, |
11123 | 1794 &samples[n * MAX_FRAMESIZE], |
1795 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], | |
1796 &excitation[s->history_nsamples + n * MAX_FRAMESIZE], | |
1797 &synth[s->lsps + n * MAX_FRAMESIZE]))) | |
1798 return res; | |
1799 } | |
1800 | |
1801 /* Statistics? FIXME - we don't check for length, a slight overrun | |
1802 * will be caught by internal buffer padding, and anything else | |
1803 * will be skipped, not read. */ | |
1804 if (get_bits1(gb)) { | |
1805 res = get_bits(gb, 4); | |
1806 skip_bits(gb, 10 * (res + 1)); | |
1807 } | |
1808 | |
1809 /* Specify nr. of output samples */ | |
1810 *data_size = n_samples * sizeof(float); | |
1811 | |
1812 /* Update history */ | |
1813 memcpy(s->prev_lsps, lsps[2], | |
1814 s->lsps * sizeof(*s->prev_lsps)); | |
1815 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], | |
1816 s->lsps * sizeof(*synth)); | |
1817 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], | |
1818 s->history_nsamples * sizeof(*excitation)); | |
11653 | 1819 if (s->do_apf) |
1820 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], | |
1821 s->history_nsamples * sizeof(*s->zero_exc_pf)); | |
11123 | 1822 |
1823 return 0; | |
1824 } | |
1825 | |
1826 /** | |
1827 * Parse the packet header at the start of each packet (input data to this | |
1828 * decoder). | |
1829 * | |
1830 * @param s WMA Voice decoding context private data | |
11556 | 1831 * @return 1 if not enough bits were available, or 0 on success. |
11123 | 1832 */ |
1833 static int parse_packet_header(WMAVoiceContext *s) | |
1834 { | |
1835 GetBitContext *gb = &s->gb; | |
1836 unsigned int res; | |
1837 | |
1838 if (get_bits_left(gb) < 11) | |
1839 return 1; | |
1840 skip_bits(gb, 4); // packet sequence number | |
1841 s->has_residual_lsps = get_bits1(gb); | |
1842 do { | |
1843 res = get_bits(gb, 6); // number of superframes per packet | |
1844 // (minus first one if there is spillover) | |
1845 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) | |
1846 return 1; | |
1847 } while (res == 0x3F); | |
1848 s->spillover_nbits = get_bits(gb, s->spillover_bitsize); | |
1849 | |
1850 return 0; | |
1851 } | |
1852 | |
1853 /** | |
1854 * Copy (unaligned) bits from gb/data/size to pb. | |
1855 * | |
1856 * @param pb target buffer to copy bits into | |
1857 * @param data source buffer to copy bits from | |
1858 * @param size size of the source data, in bytes | |
1859 * @param gb bit I/O context specifying the current position in the source. | |
1860 * data. This function might use this to align the bit position to | |
1861 * a whole-byte boundary before calling #ff_copy_bits() on aligned | |
1862 * source data | |
1863 * @param nbits the amount of bits to copy from source to target | |
1864 * | |
1865 * @note after calling this function, the current position in the input bit | |
1866 * I/O context is undefined. | |
1867 */ | |
1868 static void copy_bits(PutBitContext *pb, | |
1869 const uint8_t *data, int size, | |
1870 GetBitContext *gb, int nbits) | |
1871 { | |
1872 int rmn_bytes, rmn_bits; | |
1873 | |
1874 rmn_bits = rmn_bytes = get_bits_left(gb); | |
1875 if (rmn_bits < nbits) | |
1876 return; | |
1877 rmn_bits &= 7; rmn_bytes >>= 3; | |
1878 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) | |
1879 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); | |
1880 ff_copy_bits(pb, data + size - rmn_bytes, | |
1881 FFMIN(nbits - rmn_bits, rmn_bytes << 3)); | |
1882 } | |
1883 | |
1884 /** | |
1885 * Packet decoding: a packet is anything that the (ASF) demuxer contains, | |
1886 * and we expect that the demuxer / application provides it to us as such | |
1887 * (else you'll probably get garbage as output). Every packet has a size of | |
1888 * ctx->block_align bytes, starts with a packet header (see | |
1889 * #parse_packet_header()), and then a series of superframes. Superframe | |
1890 * boundaries may exceed packets, i.e. superframes can split data over | |
1891 * multiple (two) packets. | |
1892 * | |
1893 * For more information about frames, see #synth_superframe(). | |
1894 */ | |
1895 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, | |
1896 int *data_size, AVPacket *avpkt) | |
1897 { | |
1898 WMAVoiceContext *s = ctx->priv_data; | |
1899 GetBitContext *gb = &s->gb; | |
1900 int size, res, pos; | |
1901 | |
1902 if (*data_size < 480 * sizeof(float)) { | |
1903 av_log(ctx, AV_LOG_ERROR, | |
1904 "Output buffer too small (%d given - %lu needed)\n", | |
1905 *data_size, 480 * sizeof(float)); | |
1906 return -1; | |
1907 } | |
1908 *data_size = 0; | |
1909 | |
1910 /* Packets are sometimes a multiple of ctx->block_align, with a packet | |
1911 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer | |
1912 * feeds us ASF packets, which may concatenate multiple "codec" packets | |
1913 * in a single "muxer" packet, so we artificially emulate that by | |
1914 * capping the packet size at ctx->block_align. */ | |
1915 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); | |
1916 if (!size) | |
1917 return 0; | |
1918 init_get_bits(&s->gb, avpkt->data, size << 3); | |
1919 | |
1920 /* size == ctx->block_align is used to indicate whether we are dealing with | |
1921 * a new packet or a packet of which we already read the packet header | |
1922 * previously. */ | |
1923 if (size == ctx->block_align) { // new packet header | |
1924 if ((res = parse_packet_header(s)) < 0) | |
1925 return res; | |
1926 | |
1927 /* If the packet header specifies a s->spillover_nbits, then we want | |
1928 * to push out all data of the previous packet (+ spillover) before | |
1929 * continuing to parse new superframes in the current packet. */ | |
1930 if (s->spillover_nbits > 0) { | |
1931 if (s->sframe_cache_size > 0) { | |
1932 int cnt = get_bits_count(gb); | |
1933 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); | |
1934 flush_put_bits(&s->pb); | |
1935 s->sframe_cache_size += s->spillover_nbits; | |
1936 if ((res = synth_superframe(ctx, data, data_size)) == 0 && | |
1937 *data_size > 0) { | |
1938 cnt += s->spillover_nbits; | |
1939 s->skip_bits_next = cnt & 7; | |
1940 return cnt >> 3; | |
1941 } else | |
1942 skip_bits_long (gb, s->spillover_nbits - cnt + | |
1943 get_bits_count(gb)); // resync | |
1944 } else | |
1945 skip_bits_long(gb, s->spillover_nbits); // resync | |
1946 } | |
1947 } else if (s->skip_bits_next) | |
1948 skip_bits(gb, s->skip_bits_next); | |
1949 | |
1950 /* Try parsing superframes in current packet */ | |
1951 s->sframe_cache_size = 0; | |
1952 s->skip_bits_next = 0; | |
1953 pos = get_bits_left(gb); | |
1954 if ((res = synth_superframe(ctx, data, data_size)) < 0) { | |
1955 return res; | |
1956 } else if (*data_size > 0) { | |
1957 int cnt = get_bits_count(gb); | |
1958 s->skip_bits_next = cnt & 7; | |
1959 return cnt >> 3; | |
1960 } else if ((s->sframe_cache_size = pos) > 0) { | |
1961 /* rewind bit reader to start of last (incomplete) superframe... */ | |
1962 init_get_bits(gb, avpkt->data, size << 3); | |
1963 skip_bits_long(gb, (size << 3) - pos); | |
1964 assert(get_bits_left(gb) == pos); | |
1965 | |
1966 /* ...and cache it for spillover in next packet */ | |
1967 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); | |
1968 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); | |
1969 // FIXME bad - just copy bytes as whole and add use the | |
1970 // skip_bits_next field | |
1971 } | |
1972 | |
1973 return size; | |
1974 } | |
1975 | |
11653 | 1976 static av_cold int wmavoice_decode_end(AVCodecContext *ctx) |
1977 { | |
1978 WMAVoiceContext *s = ctx->priv_data; | |
1979 | |
1980 if (s->do_apf) { | |
1981 ff_rdft_end(&s->rdft); | |
1982 ff_rdft_end(&s->irdft); | |
1983 ff_dct_end(&s->dct); | |
1984 ff_dct_end(&s->dst); | |
1985 } | |
1986 | |
1987 return 0; | |
1988 } | |
1989 | |
11123 | 1990 static av_cold void wmavoice_flush(AVCodecContext *ctx) |
1991 { | |
1992 WMAVoiceContext *s = ctx->priv_data; | |
1993 int n; | |
1994 | |
11653 | 1995 s->postfilter_agc = 0; |
11123 | 1996 s->sframe_cache_size = 0; |
1997 s->skip_bits_next = 0; | |
1998 for (n = 0; n < s->lsps; n++) | |
1999 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
2000 memset(s->excitation_history, 0, | |
2001 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); | |
2002 memset(s->synth_history, 0, | |
2003 sizeof(*s->synth_history) * MAX_LSPS); | |
2004 memset(s->gain_pred_err, 0, | |
2005 sizeof(s->gain_pred_err)); | |
11653 | 2006 |
2007 if (s->do_apf) { | |
2008 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, | |
2009 sizeof(*s->synth_filter_out_buf) * s->lsps); | |
2010 memset(s->dcf_mem, 0, | |
2011 sizeof(*s->dcf_mem) * 2); | |
2012 memset(s->zero_exc_pf, 0, | |
2013 sizeof(*s->zero_exc_pf) * s->history_nsamples); | |
2014 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); | |
2015 } | |
11123 | 2016 } |
2017 | |
2018 AVCodec wmavoice_decoder = { | |
2019 "wmavoice", | |
11560
8a4984c5cacc
Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents:
11556
diff
changeset
|
2020 AVMEDIA_TYPE_AUDIO, |
11123 | 2021 CODEC_ID_WMAVOICE, |
2022 sizeof(WMAVoiceContext), | |
2023 wmavoice_decode_init, | |
2024 NULL, | |
11653 | 2025 wmavoice_decode_end, |
11123 | 2026 wmavoice_decode_packet, |
2027 CODEC_CAP_SUBFRAMES, | |
2028 .flush = wmavoice_flush, | |
2029 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), | |
2030 }; |