Mercurial > libavcodec.hg
annotate wmavoice.c @ 12197:fbf4d5b1b664 libavcodec
Remove FF_MM_SSE2/3 flags for CPUs where this is generally not faster than
regular MMX code. Examples of this are the Core1 CPU. Instead, set a new flag,
FF_MM_SSE2/3SLOW, which can be checked for particular SSE2/3 functions that
have been checked specifically on such CPUs and are actually faster than
their MMX counterparts.
In addition, use this flag to enable particular VP8 and LPC SSE2 functions
that are faster than their MMX counterparts.
Based on a patch by Loren Merritt <lorenm AT u washington edu>.
author | rbultje |
---|---|
date | Mon, 19 Jul 2010 22:38:23 +0000 |
parents | 30867f2c9009 |
children | 7323559a53fd |
rev | line source |
---|---|
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 | |
12114 | 663 * @param fcb_type Frame (codebook) type |
11653 | 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 | |
12114 | 747 * @param zero_exc_pf destination for zero synthesis filter (16-byte aligned) |
11653 | 748 * @param fcb_type Frame type (silence, hardcoded, AW-pulses or FCB-pulses) |
749 * @param pitch Pitch of the input signal | |
750 */ | |
751 static void postfilter(WMAVoiceContext *s, const float *synth, | |
752 float *samples, int size, | |
753 const float *lpcs, float *zero_exc_pf, | |
754 int fcb_type, int pitch) | |
755 { | |
756 float synth_filter_in_buf[MAX_FRAMESIZE / 2], | |
757 *synth_pf = &s->synth_filter_out_buf[MAX_LSPS_ALIGN16], | |
758 *synth_filter_in = zero_exc_pf; | |
759 | |
760 assert(size <= MAX_FRAMESIZE / 2); | |
761 | |
762 /* generate excitation from input signal */ | |
763 ff_celp_lp_zero_synthesis_filterf(zero_exc_pf, lpcs, synth, size, s->lsps); | |
764 | |
765 if (fcb_type >= FCB_TYPE_AW_PULSES && | |
766 !kalman_smoothen(s, pitch, zero_exc_pf, synth_filter_in_buf, size)) | |
767 synth_filter_in = synth_filter_in_buf; | |
768 | |
769 /* re-synthesize speech after smoothening, and keep history */ | |
770 ff_celp_lp_synthesis_filterf(synth_pf, lpcs, | |
771 synth_filter_in, size, s->lsps); | |
772 memcpy(&synth_pf[-s->lsps], &synth_pf[size - s->lsps], | |
773 sizeof(synth_pf[0]) * s->lsps); | |
774 | |
775 wiener_denoise(s, fcb_type, synth_pf, size, lpcs); | |
776 | |
777 adaptive_gain_control(samples, synth_pf, synth, size, 0.99, | |
778 &s->postfilter_agc); | |
779 | |
780 if (s->dc_level > 8) { | |
781 /* remove ultra-low frequency DC noise / highpass filter; | |
782 * coefficients are identical to those used in SIPR decoding, | |
783 * and very closely resemble those used in AMR-NB decoding. */ | |
784 ff_acelp_apply_order_2_transfer_function(samples, samples, | |
785 (const float[2]) { -1.99997, 1.0 }, | |
786 (const float[2]) { -1.9330735188, 0.93589198496 }, | |
787 0.93980580475, s->dcf_mem, size); | |
788 } | |
789 } | |
790 /** | |
791 * @} | |
792 */ | |
793 | |
794 /** | |
11123 | 795 * Dequantize LSPs |
796 * @param lsps output pointer to the array that will hold the LSPs | |
797 * @param num number of LSPs to be dequantized | |
798 * @param values quantized values, contains n_stages values | |
799 * @param sizes range (i.e. max value) of each quantized value | |
800 * @param n_stages number of dequantization runs | |
801 * @param table dequantization table to be used | |
802 * @param mul_q LSF multiplier | |
803 * @param base_q base (lowest) LSF values | |
804 */ | |
805 static void dequant_lsps(double *lsps, int num, | |
806 const uint16_t *values, | |
807 const uint16_t *sizes, | |
808 int n_stages, const uint8_t *table, | |
809 const double *mul_q, | |
810 const double *base_q) | |
811 { | |
812 int n, m; | |
813 | |
814 memset(lsps, 0, num * sizeof(*lsps)); | |
815 for (n = 0; n < n_stages; n++) { | |
816 const uint8_t *t_off = &table[values[n] * num]; | |
817 double base = base_q[n], mul = mul_q[n]; | |
818 | |
819 for (m = 0; m < num; m++) | |
820 lsps[m] += base + mul * t_off[m]; | |
821 | |
822 table += sizes[n] * num; | |
823 } | |
824 } | |
825 | |
826 /** | |
827 * @defgroup lsp_dequant LSP dequantization routines | |
828 * LSP dequantization routines, for 10/16LSPs and independent/residual coding. | |
829 * @note we assume enough bits are available, caller should check. | |
830 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; | |
831 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. | |
832 * @{ | |
833 */ | |
834 /** | |
835 * Parse 10 independently-coded LSPs. | |
836 */ | |
837 static void dequant_lsp10i(GetBitContext *gb, double *lsps) | |
838 { | |
839 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; | |
840 static const double mul_lsf[4] = { | |
841 5.2187144800e-3, 1.4626986422e-3, | |
842 9.6179549166e-4, 1.1325736225e-3 | |
843 }; | |
844 static const double base_lsf[4] = { | |
845 M_PI * -2.15522e-1, M_PI * -6.1646e-2, | |
846 M_PI * -3.3486e-2, M_PI * -5.7408e-2 | |
847 }; | |
848 uint16_t v[4]; | |
849 | |
850 v[0] = get_bits(gb, 8); | |
851 v[1] = get_bits(gb, 6); | |
852 v[2] = get_bits(gb, 5); | |
853 v[3] = get_bits(gb, 5); | |
854 | |
855 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, | |
856 mul_lsf, base_lsf); | |
857 } | |
858 | |
859 /** | |
860 * Parse 10 independently-coded LSPs, and then derive the tables to | |
861 * generate LSPs for the other frames from them (residual coding). | |
862 */ | |
863 static void dequant_lsp10r(GetBitContext *gb, | |
864 double *i_lsps, const double *old, | |
865 double *a1, double *a2, int q_mode) | |
866 { | |
867 static const uint16_t vec_sizes[3] = { 128, 64, 64 }; | |
868 static const double mul_lsf[3] = { | |
869 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 | |
870 }; | |
871 static const double base_lsf[3] = { | |
872 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 | |
873 }; | |
874 const float (*ipol_tab)[2][10] = q_mode ? | |
875 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; | |
876 uint16_t interpol, v[3]; | |
877 int n; | |
878 | |
879 dequant_lsp10i(gb, i_lsps); | |
880 | |
881 interpol = get_bits(gb, 5); | |
882 v[0] = get_bits(gb, 7); | |
883 v[1] = get_bits(gb, 6); | |
884 v[2] = get_bits(gb, 6); | |
885 | |
886 for (n = 0; n < 10; n++) { | |
887 double delta = old[n] - i_lsps[n]; | |
888 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
889 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
890 } | |
891 | |
892 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, | |
893 mul_lsf, base_lsf); | |
894 } | |
895 | |
896 /** | |
897 * Parse 16 independently-coded LSPs. | |
898 */ | |
899 static void dequant_lsp16i(GetBitContext *gb, double *lsps) | |
900 { | |
901 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; | |
902 static const double mul_lsf[5] = { | |
903 3.3439586280e-3, 6.9908173703e-4, | |
904 3.3216608306e-3, 1.0334960326e-3, | |
905 3.1899104283e-3 | |
906 }; | |
907 static const double base_lsf[5] = { | |
908 M_PI * -1.27576e-1, M_PI * -2.4292e-2, | |
909 M_PI * -1.28094e-1, M_PI * -3.2128e-2, | |
910 M_PI * -1.29816e-1 | |
911 }; | |
912 uint16_t v[5]; | |
913 | |
914 v[0] = get_bits(gb, 8); | |
915 v[1] = get_bits(gb, 6); | |
916 v[2] = get_bits(gb, 7); | |
917 v[3] = get_bits(gb, 6); | |
918 v[4] = get_bits(gb, 7); | |
919 | |
920 dequant_lsps( lsps, 5, v, vec_sizes, 2, | |
921 wmavoice_dq_lsp16i1, mul_lsf, base_lsf); | |
922 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, | |
923 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); | |
924 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, | |
925 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); | |
926 } | |
927 | |
928 /** | |
929 * Parse 16 independently-coded LSPs, and then derive the tables to | |
930 * generate LSPs for the other frames from them (residual coding). | |
931 */ | |
932 static void dequant_lsp16r(GetBitContext *gb, | |
933 double *i_lsps, const double *old, | |
934 double *a1, double *a2, int q_mode) | |
935 { | |
936 static const uint16_t vec_sizes[3] = { 128, 128, 128 }; | |
937 static const double mul_lsf[3] = { | |
938 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 | |
939 }; | |
940 static const double base_lsf[3] = { | |
941 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 | |
942 }; | |
943 const float (*ipol_tab)[2][16] = q_mode ? | |
944 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; | |
945 uint16_t interpol, v[3]; | |
946 int n; | |
947 | |
948 dequant_lsp16i(gb, i_lsps); | |
949 | |
950 interpol = get_bits(gb, 5); | |
951 v[0] = get_bits(gb, 7); | |
952 v[1] = get_bits(gb, 7); | |
953 v[2] = get_bits(gb, 7); | |
954 | |
955 for (n = 0; n < 16; n++) { | |
956 double delta = old[n] - i_lsps[n]; | |
957 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
958 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
959 } | |
960 | |
961 dequant_lsps( a2, 10, v, vec_sizes, 1, | |
962 wmavoice_dq_lsp16r1, mul_lsf, base_lsf); | |
963 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, | |
964 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); | |
965 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, | |
966 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); | |
967 } | |
968 | |
969 /** | |
970 * @} | |
971 * @defgroup aw Pitch-adaptive window coding functions | |
972 * The next few functions are for pitch-adaptive window coding. | |
973 * @{ | |
974 */ | |
975 /** | |
976 * Parse the offset of the first pitch-adaptive window pulses, and | |
977 * the distribution of pulses between the two blocks in this frame. | |
978 * @param s WMA Voice decoding context private data | |
979 * @param gb bit I/O context | |
980 * @param pitch pitch for each block in this frame | |
981 */ | |
982 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, | |
983 const int *pitch) | |
984 { | |
985 static const int16_t start_offset[94] = { | |
986 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, | |
987 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, | |
988 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, | |
989 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, | |
990 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, | |
991 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, | |
992 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, | |
993 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 | |
994 }; | |
995 int bits, offset; | |
996 | |
997 /* position of pulse */ | |
998 s->aw_idx_is_ext = 0; | |
999 if ((bits = get_bits(gb, 6)) >= 54) { | |
1000 s->aw_idx_is_ext = 1; | |
1001 bits += (bits - 54) * 3 + get_bits(gb, 2); | |
1002 } | |
1003 | |
1004 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count | |
1005 * the distribution of the pulses in each block contained in this frame. */ | |
1006 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; | |
1007 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; | |
1008 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; | |
1009 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; | |
1010 offset += s->aw_n_pulses[0] * pitch[0]; | |
1011 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; | |
1012 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; | |
1013 | |
1014 /* if continuing from a position before the block, reset position to | |
1015 * start of block (when corrected for the range over which it can be | |
1016 * spread in aw_pulse_set1()). */ | |
1017 if (start_offset[bits] < MAX_FRAMESIZE / 2) { | |
1018 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) | |
1019 s->aw_first_pulse_off[1] -= pitch[1]; | |
1020 if (start_offset[bits] < 0) | |
1021 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) | |
1022 s->aw_first_pulse_off[0] -= pitch[0]; | |
1023 } | |
1024 } | |
1025 | |
1026 /** | |
1027 * Apply second set of pitch-adaptive window pulses. | |
1028 * @param s WMA Voice decoding context private data | |
1029 * @param gb bit I/O context | |
1030 * @param block_idx block index in frame [0, 1] | |
1031 * @param fcb structure containing fixed codebook vector info | |
1032 */ | |
1033 static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, | |
1034 int block_idx, AMRFixed *fcb) | |
1035 { | |
1036 uint16_t use_mask[7]; // only 5 are used, rest is padding | |
1037 /* in this function, idx is the index in the 80-bit (+ padding) use_mask | |
1038 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits | |
1039 * of idx are the position of the bit within a particular item in the | |
1040 * array (0 being the most significant bit, and 15 being the least | |
1041 * significant bit), and the remainder (>> 4) is the index in the | |
1042 * use_mask[]-array. This is faster and uses less memory than using a | |
1043 * 80-byte/80-int array. */ | |
1044 int pulse_off = s->aw_first_pulse_off[block_idx], | |
1045 pulse_start, n, idx, range, aidx, start_off = 0; | |
1046 | |
1047 /* set offset of first pulse to within this block */ | |
1048 if (s->aw_n_pulses[block_idx] > 0) | |
1049 while (pulse_off + s->aw_pulse_range < 1) | |
1050 pulse_off += fcb->pitch_lag; | |
1051 | |
1052 /* find range per pulse */ | |
1053 if (s->aw_n_pulses[0] > 0) { | |
1054 if (block_idx == 0) { | |
1055 range = 32; | |
1056 } else /* block_idx = 1 */ { | |
1057 range = 8; | |
1058 if (s->aw_n_pulses[block_idx] > 0) | |
1059 pulse_off = s->aw_next_pulse_off_cache; | |
1060 } | |
1061 } else | |
1062 range = 16; | |
1063 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; | |
1064 | |
1065 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, | |
1066 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus | |
1067 * we exclude that range from being pulsed again in this function. */ | |
1068 memset( use_mask, -1, 5 * sizeof(use_mask[0])); | |
1069 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); | |
1070 if (s->aw_n_pulses[block_idx] > 0) | |
1071 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { | |
1072 int excl_range = s->aw_pulse_range; // always 16 or 24 | |
1073 uint16_t *use_mask_ptr = &use_mask[idx >> 4]; | |
1074 int first_sh = 16 - (idx & 15); | |
1075 *use_mask_ptr++ &= 0xFFFF << first_sh; | |
1076 excl_range -= first_sh; | |
1077 if (excl_range >= 16) { | |
1078 *use_mask_ptr++ = 0; | |
1079 *use_mask_ptr &= 0xFFFF >> (excl_range - 16); | |
1080 } else | |
1081 *use_mask_ptr &= 0xFFFF >> excl_range; | |
1082 } | |
1083 | |
1084 /* find the 'aidx'th offset that is not excluded */ | |
1085 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); | |
1086 for (n = 0; n <= aidx; pulse_start++) { | |
1087 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; | |
1088 if (idx >= MAX_FRAMESIZE / 2) { // find from zero | |
1089 if (use_mask[0]) idx = 0x0F; | |
1090 else if (use_mask[1]) idx = 0x1F; | |
1091 else if (use_mask[2]) idx = 0x2F; | |
1092 else if (use_mask[3]) idx = 0x3F; | |
1093 else if (use_mask[4]) idx = 0x4F; | |
1094 else return; | |
1095 idx -= av_log2_16bit(use_mask[idx >> 4]); | |
1096 } | |
1097 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { | |
1098 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); | |
1099 n++; | |
1100 start_off = idx; | |
1101 } | |
1102 } | |
1103 | |
1104 fcb->x[fcb->n] = start_off; | |
1105 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; | |
1106 fcb->n++; | |
1107 | |
1108 /* set offset for next block, relative to start of that block */ | |
1109 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; | |
1110 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; | |
1111 } | |
1112 | |
1113 /** | |
1114 * Apply first set of pitch-adaptive window pulses. | |
1115 * @param s WMA Voice decoding context private data | |
1116 * @param gb bit I/O context | |
1117 * @param block_idx block index in frame [0, 1] | |
1118 * @param fcb storage location for fixed codebook pulse info | |
1119 */ | |
1120 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, | |
1121 int block_idx, AMRFixed *fcb) | |
1122 { | |
1123 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); | |
1124 float v; | |
1125 | |
1126 if (s->aw_n_pulses[block_idx] > 0) { | |
1127 int n, v_mask, i_mask, sh, n_pulses; | |
1128 | |
1129 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each | |
1130 n_pulses = 3; | |
1131 v_mask = 8; | |
1132 i_mask = 7; | |
1133 sh = 4; | |
1134 } else { // 4 pulses, 1:sign + 2:index each | |
1135 n_pulses = 4; | |
1136 v_mask = 4; | |
1137 i_mask = 3; | |
1138 sh = 3; | |
1139 } | |
1140 | |
1141 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { | |
1142 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; | |
1143 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + | |
1144 s->aw_first_pulse_off[block_idx]; | |
1145 while (fcb->x[fcb->n] < 0) | |
1146 fcb->x[fcb->n] += fcb->pitch_lag; | |
1147 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) | |
1148 fcb->n++; | |
1149 } | |
1150 } else { | |
1151 int num2 = (val & 0x1FF) >> 1, delta, idx; | |
1152 | |
1153 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } | |
1154 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } | |
1155 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } | |
1156 else { delta = 7; idx = num2 + 1 - 3 * 75; } | |
1157 v = (val & 0x200) ? -1.0 : 1.0; | |
1158 | |
1159 fcb->no_repeat_mask |= 3 << fcb->n; | |
1160 fcb->x[fcb->n] = idx - delta; | |
1161 fcb->y[fcb->n] = v; | |
1162 fcb->x[fcb->n + 1] = idx; | |
1163 fcb->y[fcb->n + 1] = (val & 1) ? -v : v; | |
1164 fcb->n += 2; | |
1165 } | |
1166 } | |
1167 | |
1168 /** | |
1169 * @} | |
1170 * | |
1171 * Generate a random number from frame_cntr and block_idx, which will lief | |
1172 * in the range [0, 1000 - block_size] (so it can be used as an index in a | |
1173 * table of size 1000 of which you want to read block_size entries). | |
1174 * | |
1175 * @param frame_cntr current frame number | |
1176 * @param block_num current block index | |
1177 * @param block_size amount of entries we want to read from a table | |
1178 * that has 1000 entries | |
11556 | 1179 * @return a (non-)random number in the [0, 1000 - block_size] range. |
11123 | 1180 */ |
1181 static int pRNG(int frame_cntr, int block_num, int block_size) | |
1182 { | |
1183 /* array to simplify the calculation of z: | |
1184 * y = (x % 9) * 5 + 6; | |
1185 * z = (49995 * x) / y; | |
1186 * Since y only has 9 values, we can remove the division by using a | |
1187 * LUT and using FASTDIV-style divisions. For each of the 9 values | |
1188 * of y, we can rewrite z as: | |
1189 * z = x * (49995 / y) + x * ((49995 % y) / y) | |
1190 * In this table, each col represents one possible value of y, the | |
1191 * first number is 49995 / y, and the second is the FASTDIV variant | |
1192 * of 49995 % y / y. */ | |
1193 static const unsigned int div_tbl[9][2] = { | |
1194 { 8332, 3 * 715827883U }, // y = 6 | |
1195 { 4545, 0 * 390451573U }, // y = 11 | |
1196 { 3124, 11 * 268435456U }, // y = 16 | |
1197 { 2380, 15 * 204522253U }, // y = 21 | |
1198 { 1922, 23 * 165191050U }, // y = 26 | |
1199 { 1612, 23 * 138547333U }, // y = 31 | |
1200 { 1388, 27 * 119304648U }, // y = 36 | |
1201 { 1219, 16 * 104755300U }, // y = 41 | |
1202 { 1086, 39 * 93368855U } // y = 46 | |
1203 }; | |
1204 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; | |
1205 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, | |
1206 // so this is effectively a modulo (%) | |
1207 y = x - 9 * MULH(477218589, x); // x % 9 | |
1208 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); | |
1209 // z = x * 49995 / (y * 5 + 6) | |
1210 return z % (1000 - block_size); | |
1211 } | |
1212 | |
1213 /** | |
1214 * Parse hardcoded signal for a single block. | |
1215 * @note see #synth_block(). | |
1216 */ | |
1217 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, | |
1218 int block_idx, int size, | |
1219 const struct frame_type_desc *frame_desc, | |
1220 float *excitation) | |
1221 { | |
1222 float gain; | |
1223 int n, r_idx; | |
1224 | |
1225 assert(size <= MAX_FRAMESIZE); | |
1226 | |
1227 /* Set the offset from which we start reading wmavoice_std_codebook */ | |
1228 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1229 r_idx = pRNG(s->frame_cntr, block_idx, size); | |
1230 gain = s->silence_gain; | |
1231 } else /* FCB_TYPE_HARDCODED */ { | |
1232 r_idx = get_bits(gb, 8); | |
1233 gain = wmavoice_gain_universal[get_bits(gb, 6)]; | |
1234 } | |
1235 | |
1236 /* Clear gain prediction parameters */ | |
1237 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); | |
1238 | |
1239 /* Apply gain to hardcoded codebook and use that as excitation signal */ | |
1240 for (n = 0; n < size; n++) | |
1241 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; | |
1242 } | |
1243 | |
1244 /** | |
1245 * Parse FCB/ACB signal for a single block. | |
1246 * @note see #synth_block(). | |
1247 */ | |
1248 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, | |
1249 int block_idx, int size, | |
1250 int block_pitch_sh2, | |
1251 const struct frame_type_desc *frame_desc, | |
1252 float *excitation) | |
1253 { | |
1254 static const float gain_coeff[6] = { | |
1255 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 | |
1256 }; | |
1257 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; | |
1258 int n, idx, gain_weight; | |
1259 AMRFixed fcb; | |
1260 | |
1261 assert(size <= MAX_FRAMESIZE / 2); | |
1262 memset(pulses, 0, sizeof(*pulses) * size); | |
1263 | |
1264 fcb.pitch_lag = block_pitch_sh2 >> 2; | |
1265 fcb.pitch_fac = 1.0; | |
1266 fcb.no_repeat_mask = 0; | |
1267 fcb.n = 0; | |
1268 | |
1269 /* For the other frame types, this is where we apply the innovation | |
1270 * (fixed) codebook pulses of the speech signal. */ | |
1271 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1272 aw_pulse_set1(s, gb, block_idx, &fcb); | |
1273 aw_pulse_set2(s, gb, block_idx, &fcb); | |
1274 } else /* FCB_TYPE_EXC_PULSES */ { | |
1275 int offset_nbits = 5 - frame_desc->log_n_blocks; | |
1276 | |
1277 fcb.no_repeat_mask = -1; | |
1278 /* similar to ff_decode_10_pulses_35bits(), but with single pulses | |
1279 * (instead of double) for a subset of pulses */ | |
1280 for (n = 0; n < 5; n++) { | |
1281 float sign; | |
1282 int pos1, pos2; | |
1283 | |
1284 sign = get_bits1(gb) ? 1.0 : -1.0; | |
1285 pos1 = get_bits(gb, offset_nbits); | |
1286 fcb.x[fcb.n] = n + 5 * pos1; | |
1287 fcb.y[fcb.n++] = sign; | |
1288 if (n < frame_desc->dbl_pulses) { | |
1289 pos2 = get_bits(gb, offset_nbits); | |
1290 fcb.x[fcb.n] = n + 5 * pos2; | |
1291 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; | |
1292 } | |
1293 } | |
1294 } | |
1295 ff_set_fixed_vector(pulses, &fcb, 1.0, size); | |
1296 | |
1297 /* Calculate gain for adaptive & fixed codebook signal. | |
1298 * see ff_amr_set_fixed_gain(). */ | |
1299 idx = get_bits(gb, 7); | |
1300 fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) - | |
1301 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); | |
1302 acb_gain = wmavoice_gain_codebook_acb[idx]; | |
1303 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], | |
1304 -2.9957322736 /* log(0.05) */, | |
1305 1.6094379124 /* log(5.0) */); | |
1306 | |
1307 gain_weight = 8 >> frame_desc->log_n_blocks; | |
1308 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, | |
1309 sizeof(*s->gain_pred_err) * (6 - gain_weight)); | |
1310 for (n = 0; n < gain_weight; n++) | |
1311 s->gain_pred_err[n] = pred_err; | |
1312 | |
1313 /* Calculation of adaptive codebook */ | |
1314 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1315 int len; | |
1316 for (n = 0; n < size; n += len) { | |
1317 int next_idx_sh16; | |
1318 int abs_idx = block_idx * size + n; | |
1319 int pitch_sh16 = (s->last_pitch_val << 16) + | |
1320 s->pitch_diff_sh16 * abs_idx; | |
1321 int pitch = (pitch_sh16 + 0x6FFF) >> 16; | |
1322 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; | |
1323 idx = idx_sh16 >> 16; | |
1324 if (s->pitch_diff_sh16) { | |
1325 if (s->pitch_diff_sh16 > 0) { | |
1326 next_idx_sh16 = (idx_sh16) &~ 0xFFFF; | |
1327 } else | |
1328 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; | |
1329 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, | |
1330 1, size - n); | |
1331 } else | |
1332 len = size; | |
1333 | |
1334 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], | |
1335 wmavoice_ipol1_coeffs, 17, | |
1336 idx, 9, len); | |
1337 } | |
1338 } else /* ACB_TYPE_HAMMING */ { | |
1339 int block_pitch = block_pitch_sh2 >> 2; | |
1340 idx = block_pitch_sh2 & 3; | |
1341 if (idx) { | |
1342 ff_acelp_interpolatef(excitation, &excitation[-block_pitch], | |
1343 wmavoice_ipol2_coeffs, 4, | |
1344 idx, 8, size); | |
1345 } else | |
12021 | 1346 av_memcpy_backptr((uint8_t *) excitation, sizeof(float) * block_pitch, |
11123 | 1347 sizeof(float) * size); |
1348 } | |
1349 | |
1350 /* Interpolate ACB/FCB and use as excitation signal */ | |
1351 ff_weighted_vector_sumf(excitation, excitation, pulses, | |
1352 acb_gain, fcb_gain, size); | |
1353 } | |
1354 | |
1355 /** | |
1356 * Parse data in a single block. | |
1357 * @note we assume enough bits are available, caller should check. | |
1358 * | |
1359 * @param s WMA Voice decoding context private data | |
1360 * @param gb bit I/O context | |
1361 * @param block_idx index of the to-be-read block | |
1362 * @param size amount of samples to be read in this block | |
1363 * @param block_pitch_sh2 pitch for this block << 2 | |
1364 * @param lsps LSPs for (the end of) this frame | |
1365 * @param prev_lsps LSPs for the last frame | |
1366 * @param frame_desc frame type descriptor | |
1367 * @param excitation target memory for the ACB+FCB interpolated signal | |
1368 * @param synth target memory for the speech synthesis filter output | |
1369 * @return 0 on success, <0 on error. | |
1370 */ | |
1371 static void synth_block(WMAVoiceContext *s, GetBitContext *gb, | |
1372 int block_idx, int size, | |
1373 int block_pitch_sh2, | |
1374 const double *lsps, const double *prev_lsps, | |
1375 const struct frame_type_desc *frame_desc, | |
1376 float *excitation, float *synth) | |
1377 { | |
1378 double i_lsps[MAX_LSPS]; | |
1379 float lpcs[MAX_LSPS]; | |
1380 float fac; | |
1381 int n; | |
1382 | |
1383 if (frame_desc->acb_type == ACB_TYPE_NONE) | |
1384 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); | |
1385 else | |
1386 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, | |
1387 frame_desc, excitation); | |
1388 | |
1389 /* convert interpolated LSPs to LPCs */ | |
1390 fac = (block_idx + 0.5) / frame_desc->n_blocks; | |
1391 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1392 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); | |
1393 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1394 | |
1395 /* Speech synthesis */ | |
1396 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); | |
1397 } | |
1398 | |
1399 /** | |
1400 * Synthesize output samples for a single frame. | |
1401 * @note we assume enough bits are available, caller should check. | |
1402 * | |
1403 * @param ctx WMA Voice decoder context | |
1404 * @param gb bit I/O context (s->gb or one for cross-packet superframes) | |
11653 | 1405 * @param frame_idx Frame number within superframe [0-2] |
11123 | 1406 * @param samples pointer to output sample buffer, has space for at least 160 |
1407 * samples | |
1408 * @param lsps LSP array | |
1409 * @param prev_lsps array of previous frame's LSPs | |
1410 * @param excitation target buffer for excitation signal | |
1411 * @param synth target buffer for synthesized speech data | |
1412 * @return 0 on success, <0 on error. | |
1413 */ | |
11653 | 1414 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, int frame_idx, |
11123 | 1415 float *samples, |
1416 const double *lsps, const double *prev_lsps, | |
1417 float *excitation, float *synth) | |
1418 { | |
1419 WMAVoiceContext *s = ctx->priv_data; | |
1420 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val; | |
1421 int pitch[MAX_BLOCKS], last_block_pitch; | |
1422 | |
1423 /* Parse frame type ("frame header"), see frame_descs */ | |
1424 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], | |
1425 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; | |
1426 | |
1427 if (bd_idx < 0) { | |
1428 av_log(ctx, AV_LOG_ERROR, | |
1429 "Invalid frame type VLC code, skipping\n"); | |
1430 return -1; | |
1431 } | |
1432 | |
1433 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ | |
1434 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { | |
1435 /* Pitch is provided per frame, which is interpreted as the pitch of | |
1436 * the last sample of the last block of this frame. We can interpolate | |
1437 * the pitch of other blocks (and even pitch-per-sample) by gradually | |
1438 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ | |
1439 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; | |
1440 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; | |
1441 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); | |
1442 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); | |
1443 if (s->last_acb_type == ACB_TYPE_NONE || | |
1444 20 * abs(cur_pitch_val - s->last_pitch_val) > | |
1445 (cur_pitch_val + s->last_pitch_val)) | |
1446 s->last_pitch_val = cur_pitch_val; | |
1447 | |
1448 /* pitch per block */ | |
1449 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1450 int fac = n * 2 + 1; | |
1451 | |
1452 pitch[n] = (MUL16(fac, cur_pitch_val) + | |
1453 MUL16((n_blocks_x2 - fac), s->last_pitch_val) + | |
1454 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; | |
1455 } | |
1456 | |
1457 /* "pitch-diff-per-sample" for calculation of pitch per sample */ | |
1458 s->pitch_diff_sh16 = | |
1459 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; | |
1460 } | |
1461 | |
1462 /* Global gain (if silence) and pitch-adaptive window coordinates */ | |
1463 switch (frame_descs[bd_idx].fcb_type) { | |
1464 case FCB_TYPE_SILENCE: | |
1465 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; | |
1466 break; | |
1467 case FCB_TYPE_AW_PULSES: | |
1468 aw_parse_coords(s, gb, pitch); | |
1469 break; | |
1470 } | |
1471 | |
1472 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1473 int bl_pitch_sh2; | |
1474 | |
1475 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ | |
1476 switch (frame_descs[bd_idx].acb_type) { | |
1477 case ACB_TYPE_HAMMING: { | |
1478 /* Pitch is given per block. Per-block pitches are encoded as an | |
1479 * absolute value for the first block, and then delta values | |
1480 * relative to this value) for all subsequent blocks. The scale of | |
1481 * this pitch value is semi-logaritmic compared to its use in the | |
1482 * decoder, so we convert it to normal scale also. */ | |
1483 int block_pitch, | |
1484 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, | |
1485 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, | |
1486 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; | |
1487 | |
1488 if (n == 0) { | |
1489 block_pitch = get_bits(gb, s->block_pitch_nbits); | |
1490 } else | |
1491 block_pitch = last_block_pitch - s->block_delta_pitch_hrange + | |
1492 get_bits(gb, s->block_delta_pitch_nbits); | |
1493 /* Convert last_ so that any next delta is within _range */ | |
1494 last_block_pitch = av_clip(block_pitch, | |
1495 s->block_delta_pitch_hrange, | |
1496 s->block_pitch_range - | |
1497 s->block_delta_pitch_hrange); | |
1498 | |
1499 /* Convert semi-log-style scale back to normal scale */ | |
1500 if (block_pitch < t1) { | |
1501 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; | |
1502 } else { | |
1503 block_pitch -= t1; | |
1504 if (block_pitch < t2) { | |
1505 bl_pitch_sh2 = | |
1506 (s->block_conv_table[1] << 2) + (block_pitch << 1); | |
1507 } else { | |
1508 block_pitch -= t2; | |
1509 if (block_pitch < t3) { | |
1510 bl_pitch_sh2 = | |
1511 (s->block_conv_table[2] + block_pitch) << 2; | |
1512 } else | |
1513 bl_pitch_sh2 = s->block_conv_table[3] << 2; | |
1514 } | |
1515 } | |
1516 pitch[n] = bl_pitch_sh2 >> 2; | |
1517 break; | |
1518 } | |
1519 | |
1520 case ACB_TYPE_ASYMMETRIC: { | |
1521 bl_pitch_sh2 = pitch[n] << 2; | |
1522 break; | |
1523 } | |
1524 | |
1525 default: // ACB_TYPE_NONE has no pitch | |
1526 bl_pitch_sh2 = 0; | |
1527 break; | |
1528 } | |
1529 | |
1530 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, | |
1531 lsps, prev_lsps, &frame_descs[bd_idx], | |
1532 &excitation[n * block_nsamples], | |
1533 &synth[n * block_nsamples]); | |
1534 } | |
1535 | |
1536 /* Averaging projection filter, if applicable. Else, just copy samples | |
1537 * from synthesis buffer */ | |
1538 if (s->do_apf) { | |
11653 | 1539 double i_lsps[MAX_LSPS]; |
1540 float lpcs[MAX_LSPS]; | |
1541 | |
1542 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1543 i_lsps[n] = cos(0.5 * (prev_lsps[n] + lsps[n])); | |
1544 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1545 postfilter(s, synth, samples, 80, lpcs, | |
1546 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx], | |
1547 frame_descs[bd_idx].fcb_type, pitch[0]); | |
1548 | |
1549 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
1550 i_lsps[n] = cos(lsps[n]); | |
1551 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
1552 postfilter(s, &synth[80], &samples[80], 80, lpcs, | |
1553 &s->zero_exc_pf[s->history_nsamples + MAX_FRAMESIZE * frame_idx + 80], | |
1554 frame_descs[bd_idx].fcb_type, pitch[0]); | |
1555 } else | |
11652
8b6f3d3b55cb
Move clipping of audio samples (for those codecs outputting float) from decoder
rbultje
parents:
11644
diff
changeset
|
1556 memcpy(samples, synth, 160 * sizeof(synth[0])); |
11123 | 1557 |
1558 /* Cache values for next frame */ | |
1559 s->frame_cntr++; | |
1560 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) | |
1561 s->last_acb_type = frame_descs[bd_idx].acb_type; | |
1562 switch (frame_descs[bd_idx].acb_type) { | |
1563 case ACB_TYPE_NONE: | |
1564 s->last_pitch_val = 0; | |
1565 break; | |
1566 case ACB_TYPE_ASYMMETRIC: | |
1567 s->last_pitch_val = cur_pitch_val; | |
1568 break; | |
1569 case ACB_TYPE_HAMMING: | |
1570 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; | |
1571 break; | |
1572 } | |
1573 | |
1574 return 0; | |
1575 } | |
1576 | |
1577 /** | |
1578 * Ensure minimum value for first item, maximum value for last value, | |
1579 * proper spacing between each value and proper ordering. | |
1580 * | |
1581 * @param lsps array of LSPs | |
1582 * @param num size of LSP array | |
1583 * | |
1584 * @note basically a double version of #ff_acelp_reorder_lsf(), might be | |
1585 * useful to put in a generic location later on. Parts are also | |
1586 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), | |
1587 * which is in float. | |
1588 */ | |
1589 static void stabilize_lsps(double *lsps, int num) | |
1590 { | |
1591 int n, m, l; | |
1592 | |
1593 /* set minimum value for first, maximum value for last and minimum | |
1594 * spacing between LSF values. | |
1595 * Very similar to ff_set_min_dist_lsf(), but in double. */ | |
1596 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); | |
1597 for (n = 1; n < num; n++) | |
1598 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); | |
1599 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); | |
1600 | |
1601 /* reorder (looks like one-time / non-recursed bubblesort). | |
1602 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ | |
1603 for (n = 1; n < num; n++) { | |
1604 if (lsps[n] < lsps[n - 1]) { | |
1605 for (m = 1; m < num; m++) { | |
1606 double tmp = lsps[m]; | |
1607 for (l = m - 1; l >= 0; l--) { | |
1608 if (lsps[l] <= tmp) break; | |
1609 lsps[l + 1] = lsps[l]; | |
1610 } | |
1611 lsps[l + 1] = tmp; | |
1612 } | |
1613 break; | |
1614 } | |
1615 } | |
1616 } | |
1617 | |
1618 /** | |
1619 * Test if there's enough bits to read 1 superframe. | |
1620 * | |
1621 * @param orig_gb bit I/O context used for reading. This function | |
1622 * does not modify the state of the bitreader; it | |
1623 * only uses it to copy the current stream position | |
1624 * @param s WMA Voice decoding context private data | |
11556 | 1625 * @return -1 if unsupported, 1 on not enough bits or 0 if OK. |
11123 | 1626 */ |
1627 static int check_bits_for_superframe(GetBitContext *orig_gb, | |
1628 WMAVoiceContext *s) | |
1629 { | |
1630 GetBitContext s_gb, *gb = &s_gb; | |
1631 int n, need_bits, bd_idx; | |
1632 const struct frame_type_desc *frame_desc; | |
1633 | |
1634 /* initialize a copy */ | |
1635 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); | |
1636 skip_bits_long(gb, get_bits_count(orig_gb)); | |
1637 assert(get_bits_left(gb) == get_bits_left(orig_gb)); | |
1638 | |
1639 /* superframe header */ | |
1640 if (get_bits_left(gb) < 14) | |
1641 return 1; | |
1642 if (!get_bits1(gb)) | |
1643 return -1; // WMAPro-in-WMAVoice superframe | |
1644 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe | |
1645 if (s->has_residual_lsps) { // residual LSPs (for all frames) | |
1646 if (get_bits_left(gb) < s->sframe_lsp_bitsize) | |
1647 return 1; | |
1648 skip_bits_long(gb, s->sframe_lsp_bitsize); | |
1649 } | |
1650 | |
1651 /* frames */ | |
1652 for (n = 0; n < MAX_FRAMES; n++) { | |
1653 int aw_idx_is_ext = 0; | |
1654 | |
1655 if (!s->has_residual_lsps) { // independent LSPs (per-frame) | |
1656 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; | |
1657 skip_bits_long(gb, s->frame_lsp_bitsize); | |
1658 } | |
1659 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; | |
1660 if (bd_idx < 0) | |
1661 return -1; // invalid frame type VLC code | |
1662 frame_desc = &frame_descs[bd_idx]; | |
1663 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1664 if (get_bits_left(gb) < s->pitch_nbits) | |
1665 return 1; | |
1666 skip_bits_long(gb, s->pitch_nbits); | |
1667 } | |
1668 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1669 skip_bits(gb, 8); | |
1670 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1671 int tmp = get_bits(gb, 6); | |
1672 if (tmp >= 0x36) { | |
1673 skip_bits(gb, 2); | |
1674 aw_idx_is_ext = 1; | |
1675 } | |
1676 } | |
1677 | |
1678 /* blocks */ | |
1679 if (frame_desc->acb_type == ACB_TYPE_HAMMING) { | |
1680 need_bits = s->block_pitch_nbits + | |
1681 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; | |
1682 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1683 need_bits = 2 * !aw_idx_is_ext; | |
1684 } else | |
1685 need_bits = 0; | |
1686 need_bits += frame_desc->frame_size; | |
1687 if (get_bits_left(gb) < need_bits) | |
1688 return 1; | |
1689 skip_bits_long(gb, need_bits); | |
1690 } | |
1691 | |
1692 return 0; | |
1693 } | |
1694 | |
1695 /** | |
1696 * Synthesize output samples for a single superframe. If we have any data | |
1697 * cached in s->sframe_cache, that will be used instead of whatever is loaded | |
1698 * in s->gb. | |
1699 * | |
1700 * WMA Voice superframes contain 3 frames, each containing 160 audio samples, | |
1701 * to give a total of 480 samples per frame. See #synth_frame() for frame | |
1702 * parsing. In addition to 3 frames, superframes can also contain the LSPs | |
1703 * (if these are globally specified for all frames (residually); they can | |
1704 * also be specified individually per-frame. See the s->has_residual_lsps | |
1705 * option), and can specify the number of samples encoded in this superframe | |
1706 * (if less than 480), usually used to prevent blanks at track boundaries. | |
1707 * | |
1708 * @param ctx WMA Voice decoder context | |
1709 * @param samples pointer to output buffer for voice samples | |
1710 * @param data_size pointer containing the size of #samples on input, and the | |
1711 * amount of #samples filled on output | |
1712 * @return 0 on success, <0 on error or 1 if there was not enough data to | |
1713 * fully parse the superframe | |
1714 */ | |
1715 static int synth_superframe(AVCodecContext *ctx, | |
1716 float *samples, int *data_size) | |
1717 { | |
1718 WMAVoiceContext *s = ctx->priv_data; | |
1719 GetBitContext *gb = &s->gb, s_gb; | |
1720 int n, res, n_samples = 480; | |
1721 double lsps[MAX_FRAMES][MAX_LSPS]; | |
1722 const double *mean_lsf = s->lsps == 16 ? | |
1723 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; | |
1724 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; | |
1725 float synth[MAX_LSPS + MAX_SFRAMESIZE]; | |
1726 | |
1727 memcpy(synth, s->synth_history, | |
1728 s->lsps * sizeof(*synth)); | |
1729 memcpy(excitation, s->excitation_history, | |
1730 s->history_nsamples * sizeof(*excitation)); | |
1731 | |
1732 if (s->sframe_cache_size > 0) { | |
1733 gb = &s_gb; | |
1734 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); | |
1735 s->sframe_cache_size = 0; | |
1736 } | |
1737 | |
1738 if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; | |
1739 | |
1740 /* First bit is speech/music bit, it differentiates between WMAVoice | |
1741 * speech samples (the actual codec) and WMAVoice music samples, which | |
1742 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in | |
1743 * the wild yet. */ | |
1744 if (!get_bits1(gb)) { | |
1745 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1); | |
1746 return -1; | |
1747 } | |
1748 | |
1749 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ | |
1750 if (get_bits1(gb)) { | |
1751 if ((n_samples = get_bits(gb, 12)) > 480) { | |
1752 av_log(ctx, AV_LOG_ERROR, | |
1753 "Superframe encodes >480 samples (%d), not allowed\n", | |
1754 n_samples); | |
1755 return -1; | |
1756 } | |
1757 } | |
1758 /* Parse LSPs, if global for the superframe (can also be per-frame). */ | |
1759 if (s->has_residual_lsps) { | |
1760 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; | |
1761 | |
1762 for (n = 0; n < s->lsps; n++) | |
1763 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; | |
1764 | |
1765 if (s->lsps == 10) { | |
1766 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1767 } else /* s->lsps == 16 */ | |
1768 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1769 | |
1770 for (n = 0; n < s->lsps; n++) { | |
1771 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); | |
1772 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); | |
1773 lsps[2][n] += mean_lsf[n]; | |
1774 } | |
1775 for (n = 0; n < 3; n++) | |
1776 stabilize_lsps(lsps[n], s->lsps); | |
1777 } | |
1778 | |
1779 /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ | |
1780 for (n = 0; n < 3; n++) { | |
1781 if (!s->has_residual_lsps) { | |
1782 int m; | |
1783 | |
1784 if (s->lsps == 10) { | |
1785 dequant_lsp10i(gb, lsps[n]); | |
1786 } else /* s->lsps == 16 */ | |
1787 dequant_lsp16i(gb, lsps[n]); | |
1788 | |
1789 for (m = 0; m < s->lsps; m++) | |
1790 lsps[n][m] += mean_lsf[m]; | |
1791 stabilize_lsps(lsps[n], s->lsps); | |
1792 } | |
1793 | |
11653 | 1794 if ((res = synth_frame(ctx, gb, n, |
11123 | 1795 &samples[n * MAX_FRAMESIZE], |
1796 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], | |
1797 &excitation[s->history_nsamples + n * MAX_FRAMESIZE], | |
1798 &synth[s->lsps + n * MAX_FRAMESIZE]))) | |
1799 return res; | |
1800 } | |
1801 | |
1802 /* Statistics? FIXME - we don't check for length, a slight overrun | |
1803 * will be caught by internal buffer padding, and anything else | |
1804 * will be skipped, not read. */ | |
1805 if (get_bits1(gb)) { | |
1806 res = get_bits(gb, 4); | |
1807 skip_bits(gb, 10 * (res + 1)); | |
1808 } | |
1809 | |
1810 /* Specify nr. of output samples */ | |
1811 *data_size = n_samples * sizeof(float); | |
1812 | |
1813 /* Update history */ | |
1814 memcpy(s->prev_lsps, lsps[2], | |
1815 s->lsps * sizeof(*s->prev_lsps)); | |
1816 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], | |
1817 s->lsps * sizeof(*synth)); | |
1818 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], | |
1819 s->history_nsamples * sizeof(*excitation)); | |
11653 | 1820 if (s->do_apf) |
1821 memmove(s->zero_exc_pf, &s->zero_exc_pf[MAX_SFRAMESIZE], | |
1822 s->history_nsamples * sizeof(*s->zero_exc_pf)); | |
11123 | 1823 |
1824 return 0; | |
1825 } | |
1826 | |
1827 /** | |
1828 * Parse the packet header at the start of each packet (input data to this | |
1829 * decoder). | |
1830 * | |
1831 * @param s WMA Voice decoding context private data | |
11556 | 1832 * @return 1 if not enough bits were available, or 0 on success. |
11123 | 1833 */ |
1834 static int parse_packet_header(WMAVoiceContext *s) | |
1835 { | |
1836 GetBitContext *gb = &s->gb; | |
1837 unsigned int res; | |
1838 | |
1839 if (get_bits_left(gb) < 11) | |
1840 return 1; | |
1841 skip_bits(gb, 4); // packet sequence number | |
1842 s->has_residual_lsps = get_bits1(gb); | |
1843 do { | |
1844 res = get_bits(gb, 6); // number of superframes per packet | |
1845 // (minus first one if there is spillover) | |
1846 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) | |
1847 return 1; | |
1848 } while (res == 0x3F); | |
1849 s->spillover_nbits = get_bits(gb, s->spillover_bitsize); | |
1850 | |
1851 return 0; | |
1852 } | |
1853 | |
1854 /** | |
1855 * Copy (unaligned) bits from gb/data/size to pb. | |
1856 * | |
1857 * @param pb target buffer to copy bits into | |
1858 * @param data source buffer to copy bits from | |
1859 * @param size size of the source data, in bytes | |
1860 * @param gb bit I/O context specifying the current position in the source. | |
1861 * data. This function might use this to align the bit position to | |
1862 * a whole-byte boundary before calling #ff_copy_bits() on aligned | |
1863 * source data | |
1864 * @param nbits the amount of bits to copy from source to target | |
1865 * | |
1866 * @note after calling this function, the current position in the input bit | |
1867 * I/O context is undefined. | |
1868 */ | |
1869 static void copy_bits(PutBitContext *pb, | |
1870 const uint8_t *data, int size, | |
1871 GetBitContext *gb, int nbits) | |
1872 { | |
1873 int rmn_bytes, rmn_bits; | |
1874 | |
1875 rmn_bits = rmn_bytes = get_bits_left(gb); | |
1876 if (rmn_bits < nbits) | |
1877 return; | |
1878 rmn_bits &= 7; rmn_bytes >>= 3; | |
1879 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) | |
1880 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); | |
1881 ff_copy_bits(pb, data + size - rmn_bytes, | |
1882 FFMIN(nbits - rmn_bits, rmn_bytes << 3)); | |
1883 } | |
1884 | |
1885 /** | |
1886 * Packet decoding: a packet is anything that the (ASF) demuxer contains, | |
1887 * and we expect that the demuxer / application provides it to us as such | |
1888 * (else you'll probably get garbage as output). Every packet has a size of | |
1889 * ctx->block_align bytes, starts with a packet header (see | |
1890 * #parse_packet_header()), and then a series of superframes. Superframe | |
1891 * boundaries may exceed packets, i.e. superframes can split data over | |
1892 * multiple (two) packets. | |
1893 * | |
1894 * For more information about frames, see #synth_superframe(). | |
1895 */ | |
1896 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, | |
1897 int *data_size, AVPacket *avpkt) | |
1898 { | |
1899 WMAVoiceContext *s = ctx->priv_data; | |
1900 GetBitContext *gb = &s->gb; | |
1901 int size, res, pos; | |
1902 | |
1903 if (*data_size < 480 * sizeof(float)) { | |
1904 av_log(ctx, AV_LOG_ERROR, | |
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1905 "Output buffer too small (%d given - %zu needed)\n", |
11123 | 1906 *data_size, 480 * sizeof(float)); |
1907 return -1; | |
1908 } | |
1909 *data_size = 0; | |
1910 | |
1911 /* Packets are sometimes a multiple of ctx->block_align, with a packet | |
1912 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer | |
1913 * feeds us ASF packets, which may concatenate multiple "codec" packets | |
1914 * in a single "muxer" packet, so we artificially emulate that by | |
1915 * capping the packet size at ctx->block_align. */ | |
1916 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); | |
1917 if (!size) | |
1918 return 0; | |
1919 init_get_bits(&s->gb, avpkt->data, size << 3); | |
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 if (size == ctx->block_align) { // new packet header | |
1925 if ((res = parse_packet_header(s)) < 0) | |
1926 return res; | |
1927 | |
1928 /* If the packet header specifies a s->spillover_nbits, then we want | |
1929 * to push out all data of the previous packet (+ spillover) before | |
1930 * continuing to parse new superframes in the current packet. */ | |
1931 if (s->spillover_nbits > 0) { | |
1932 if (s->sframe_cache_size > 0) { | |
1933 int cnt = get_bits_count(gb); | |
1934 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); | |
1935 flush_put_bits(&s->pb); | |
1936 s->sframe_cache_size += s->spillover_nbits; | |
1937 if ((res = synth_superframe(ctx, data, data_size)) == 0 && | |
1938 *data_size > 0) { | |
1939 cnt += s->spillover_nbits; | |
1940 s->skip_bits_next = cnt & 7; | |
1941 return cnt >> 3; | |
1942 } else | |
1943 skip_bits_long (gb, s->spillover_nbits - cnt + | |
1944 get_bits_count(gb)); // resync | |
1945 } else | |
1946 skip_bits_long(gb, s->spillover_nbits); // resync | |
1947 } | |
1948 } else if (s->skip_bits_next) | |
1949 skip_bits(gb, s->skip_bits_next); | |
1950 | |
1951 /* Try parsing superframes in current packet */ | |
1952 s->sframe_cache_size = 0; | |
1953 s->skip_bits_next = 0; | |
1954 pos = get_bits_left(gb); | |
1955 if ((res = synth_superframe(ctx, data, data_size)) < 0) { | |
1956 return res; | |
1957 } else if (*data_size > 0) { | |
1958 int cnt = get_bits_count(gb); | |
1959 s->skip_bits_next = cnt & 7; | |
1960 return cnt >> 3; | |
1961 } else if ((s->sframe_cache_size = pos) > 0) { | |
1962 /* rewind bit reader to start of last (incomplete) superframe... */ | |
1963 init_get_bits(gb, avpkt->data, size << 3); | |
1964 skip_bits_long(gb, (size << 3) - pos); | |
1965 assert(get_bits_left(gb) == pos); | |
1966 | |
1967 /* ...and cache it for spillover in next packet */ | |
1968 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); | |
1969 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); | |
1970 // FIXME bad - just copy bytes as whole and add use the | |
1971 // skip_bits_next field | |
1972 } | |
1973 | |
1974 return size; | |
1975 } | |
1976 | |
11653 | 1977 static av_cold int wmavoice_decode_end(AVCodecContext *ctx) |
1978 { | |
1979 WMAVoiceContext *s = ctx->priv_data; | |
1980 | |
1981 if (s->do_apf) { | |
1982 ff_rdft_end(&s->rdft); | |
1983 ff_rdft_end(&s->irdft); | |
1984 ff_dct_end(&s->dct); | |
1985 ff_dct_end(&s->dst); | |
1986 } | |
1987 | |
1988 return 0; | |
1989 } | |
1990 | |
11123 | 1991 static av_cold void wmavoice_flush(AVCodecContext *ctx) |
1992 { | |
1993 WMAVoiceContext *s = ctx->priv_data; | |
1994 int n; | |
1995 | |
11653 | 1996 s->postfilter_agc = 0; |
11123 | 1997 s->sframe_cache_size = 0; |
1998 s->skip_bits_next = 0; | |
1999 for (n = 0; n < s->lsps; n++) | |
2000 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
2001 memset(s->excitation_history, 0, | |
2002 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); | |
2003 memset(s->synth_history, 0, | |
2004 sizeof(*s->synth_history) * MAX_LSPS); | |
2005 memset(s->gain_pred_err, 0, | |
2006 sizeof(s->gain_pred_err)); | |
11653 | 2007 |
2008 if (s->do_apf) { | |
2009 memset(&s->synth_filter_out_buf[MAX_LSPS_ALIGN16 - s->lsps], 0, | |
2010 sizeof(*s->synth_filter_out_buf) * s->lsps); | |
2011 memset(s->dcf_mem, 0, | |
2012 sizeof(*s->dcf_mem) * 2); | |
2013 memset(s->zero_exc_pf, 0, | |
2014 sizeof(*s->zero_exc_pf) * s->history_nsamples); | |
2015 memset(s->denoise_filter_cache, 0, sizeof(s->denoise_filter_cache)); | |
2016 } | |
11123 | 2017 } |
2018 | |
2019 AVCodec wmavoice_decoder = { | |
2020 "wmavoice", | |
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2021 AVMEDIA_TYPE_AUDIO, |
11123 | 2022 CODEC_ID_WMAVOICE, |
2023 sizeof(WMAVoiceContext), | |
2024 wmavoice_decode_init, | |
2025 NULL, | |
11653 | 2026 wmavoice_decode_end, |
11123 | 2027 wmavoice_decode_packet, |
2028 CODEC_CAP_SUBFRAMES, | |
2029 .flush = wmavoice_flush, | |
2030 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), | |
2031 }; |