Mercurial > libavcodec.hg
annotate wmavoice.c @ 11566:214a2a8b9e58 libavcodec
Document API addition of avcodec_copy_context().
author | rbultje |
---|---|
date | Wed, 31 Mar 2010 21:10:52 +0000 |
parents | 8a4984c5cacc |
children | 7dd2a45249a9 |
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 /** | |
23 * @file libavcodec/wmavoice.c | |
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" | |
39 | |
40 #define MAX_BLOCKS 8 ///< maximum number of blocks per frame | |
41 #define MAX_LSPS 16 ///< maximum filter order | |
42 #define MAX_FRAMES 3 ///< maximum number of frames per superframe | |
43 #define MAX_FRAMESIZE 160 ///< maximum number of samples per frame | |
44 #define MAX_SIGNAL_HISTORY 416 ///< maximum excitation signal history | |
45 #define MAX_SFRAMESIZE (MAX_FRAMESIZE * MAX_FRAMES) | |
46 ///< maximum number of samples per superframe | |
47 #define SFRAME_CACHE_MAXSIZE 256 ///< maximum cache size for frame data that | |
48 ///< was split over two packets | |
49 #define VLC_NBITS 6 ///< number of bits to read per VLC iteration | |
50 | |
51 /** | |
52 * Frame type VLC coding. | |
53 */ | |
54 static VLC frame_type_vlc; | |
55 | |
56 /** | |
57 * Adaptive codebook types. | |
58 */ | |
59 enum { | |
60 ACB_TYPE_NONE = 0, ///< no adaptive codebook (only hardcoded fixed) | |
61 ACB_TYPE_ASYMMETRIC = 1, ///< adaptive codebook with per-frame pitch, which | |
62 ///< we interpolate to get a per-sample pitch. | |
63 ///< Signal is generated using an asymmetric sinc | |
64 ///< window function | |
65 ///< @note see #wmavoice_ipol1_coeffs | |
66 ACB_TYPE_HAMMING = 2 ///< Per-block pitch with signal generation using | |
67 ///< a Hamming sinc window function | |
68 ///< @note see #wmavoice_ipol2_coeffs | |
69 }; | |
70 | |
71 /** | |
72 * Fixed codebook types. | |
73 */ | |
74 enum { | |
75 FCB_TYPE_SILENCE = 0, ///< comfort noise during silence | |
76 ///< generated from a hardcoded (fixed) codebook | |
77 ///< with per-frame (low) gain values | |
78 FCB_TYPE_HARDCODED = 1, ///< hardcoded (fixed) codebook with per-block | |
79 ///< gain values | |
80 FCB_TYPE_AW_PULSES = 2, ///< Pitch-adaptive window (AW) pulse signals, | |
81 ///< used in particular for low-bitrate streams | |
82 FCB_TYPE_EXC_PULSES = 3, ///< Innovation (fixed) codebook pulse sets in | |
83 ///< combinations of either single pulses or | |
84 ///< pulse pairs | |
85 }; | |
86 | |
87 /** | |
88 * Description of frame types. | |
89 */ | |
90 static const struct frame_type_desc { | |
91 uint8_t n_blocks; ///< amount of blocks per frame (each block | |
92 ///< (contains 160/#n_blocks samples) | |
93 uint8_t log_n_blocks; ///< log2(#n_blocks) | |
94 uint8_t acb_type; ///< Adaptive codebook type (ACB_TYPE_*) | |
95 uint8_t fcb_type; ///< Fixed codebook type (FCB_TYPE_*) | |
96 uint8_t dbl_pulses; ///< how many pulse vectors have pulse pairs | |
97 ///< (rather than just one single pulse) | |
98 ///< only if #fcb_type == #FCB_TYPE_EXC_PULSES | |
99 uint16_t frame_size; ///< the amount of bits that make up the block | |
100 ///< data (per frame) | |
101 } frame_descs[17] = { | |
102 { 1, 0, ACB_TYPE_NONE, FCB_TYPE_SILENCE, 0, 0 }, | |
103 { 2, 1, ACB_TYPE_NONE, FCB_TYPE_HARDCODED, 0, 28 }, | |
104 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_AW_PULSES, 0, 46 }, | |
105 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 80 }, | |
106 { 2, 1, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 104 }, | |
107 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 0, 108 }, | |
108 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 2, 132 }, | |
109 { 4, 2, ACB_TYPE_ASYMMETRIC, FCB_TYPE_EXC_PULSES, 5, 168 }, | |
110 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 64 }, | |
111 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 80 }, | |
112 { 2, 1, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 104 }, | |
113 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 108 }, | |
114 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 132 }, | |
115 { 4, 2, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 168 }, | |
116 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 0, 176 }, | |
117 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 2, 208 }, | |
118 { 8, 3, ACB_TYPE_HAMMING, FCB_TYPE_EXC_PULSES, 5, 256 } | |
119 }; | |
120 | |
121 /** | |
122 * WMA Voice decoding context. | |
123 */ | |
124 typedef struct { | |
125 /** | |
126 * @defgroup struct_global Global values | |
127 * Global values, specified in the stream header / extradata or used | |
128 * all over. | |
129 * @{ | |
130 */ | |
131 GetBitContext gb; ///< packet bitreader. During decoder init, | |
132 ///< it contains the extradata from the | |
133 ///< demuxer. During decoding, it contains | |
134 ///< packet data. | |
135 int8_t vbm_tree[25]; ///< converts VLC codes to frame type | |
136 | |
137 int spillover_bitsize; ///< number of bits used to specify | |
138 ///< #spillover_nbits in the packet header | |
139 ///< = ceil(log2(ctx->block_align << 3)) | |
140 int history_nsamples; ///< number of samples in history for signal | |
141 ///< prediction (through ACB) | |
142 | |
143 int do_apf; ///< whether to apply the averaged | |
144 ///< projection filter (APF) | |
145 | |
146 int lsps; ///< number of LSPs per frame [10 or 16] | |
147 int lsp_q_mode; ///< defines quantizer defaults [0, 1] | |
148 int lsp_def_mode; ///< defines different sets of LSP defaults | |
149 ///< [0, 1] | |
150 int frame_lsp_bitsize; ///< size (in bits) of LSPs, when encoded | |
151 ///< per-frame (independent coding) | |
152 int sframe_lsp_bitsize; ///< size (in bits) of LSPs, when encoded | |
153 ///< per superframe (residual coding) | |
154 | |
155 int min_pitch_val; ///< base value for pitch parsing code | |
156 int max_pitch_val; ///< max value + 1 for pitch parsing | |
157 int pitch_nbits; ///< number of bits used to specify the | |
158 ///< pitch value in the frame header | |
159 int block_pitch_nbits; ///< number of bits used to specify the | |
160 ///< first block's pitch value | |
161 int block_pitch_range; ///< range of the block pitch | |
162 int block_delta_pitch_nbits; ///< number of bits used to specify the | |
163 ///< delta pitch between this and the last | |
164 ///< block's pitch value, used in all but | |
165 ///< first block | |
166 int block_delta_pitch_hrange; ///< 1/2 range of the delta (full range is | |
167 ///< from -this to +this-1) | |
168 uint16_t block_conv_table[4]; ///< boundaries for block pitch unit/scale | |
169 ///< conversion | |
170 | |
171 /** | |
172 * @} | |
173 * @defgroup struct_packet Packet values | |
174 * Packet values, specified in the packet header or related to a packet. | |
175 * A packet is considered to be a single unit of data provided to this | |
176 * decoder by the demuxer. | |
177 * @{ | |
178 */ | |
179 int spillover_nbits; ///< number of bits of the previous packet's | |
180 ///< last superframe preceeding this | |
181 ///< packet's first full superframe (useful | |
182 ///< for re-synchronization also) | |
183 int has_residual_lsps; ///< if set, superframes contain one set of | |
184 ///< LSPs that cover all frames, encoded as | |
185 ///< independent and residual LSPs; if not | |
186 ///< set, each frame contains its own, fully | |
187 ///< independent, LSPs | |
188 int skip_bits_next; ///< number of bits to skip at the next call | |
189 ///< to #wmavoice_decode_packet() (since | |
190 ///< they're part of the previous superframe) | |
191 | |
192 uint8_t sframe_cache[SFRAME_CACHE_MAXSIZE + FF_INPUT_BUFFER_PADDING_SIZE]; | |
193 ///< cache for superframe data split over | |
194 ///< multiple packets | |
195 int sframe_cache_size; ///< set to >0 if we have data from an | |
196 ///< (incomplete) superframe from a previous | |
197 ///< packet that spilled over in the current | |
198 ///< packet; specifies the amount of bits in | |
199 ///< #sframe_cache | |
200 PutBitContext pb; ///< bitstream writer for #sframe_cache | |
201 | |
202 /** | |
203 * @} | |
204 * @defgroup struct_frame Frame and superframe values | |
205 * Superframe and frame data - these can change from frame to frame, | |
206 * although some of them do in that case serve as a cache / history for | |
207 * the next frame or superframe. | |
208 * @{ | |
209 */ | |
210 double prev_lsps[MAX_LSPS]; ///< LSPs of the last frame of the previous | |
211 ///< superframe | |
212 int last_pitch_val; ///< pitch value of the previous frame | |
213 int last_acb_type; ///< frame type [0-2] of the previous frame | |
214 int pitch_diff_sh16; ///< ((cur_pitch_val - #last_pitch_val) | |
215 ///< << 16) / #MAX_FRAMESIZE | |
216 float silence_gain; ///< set for use in blocks if #ACB_TYPE_NONE | |
217 | |
218 int aw_idx_is_ext; ///< whether the AW index was encoded in | |
219 ///< 8 bits (instead of 6) | |
220 int aw_pulse_range; ///< the range over which #aw_pulse_set1() | |
221 ///< can apply the pulse, relative to the | |
222 ///< value in aw_first_pulse_off. The exact | |
223 ///< position of the first AW-pulse is within | |
224 ///< [pulse_off, pulse_off + this], and | |
225 ///< depends on bitstream values; [16 or 24] | |
226 int aw_n_pulses[2]; ///< number of AW-pulses in each block; note | |
227 ///< that this number can be negative (in | |
228 ///< which case it basically means "zero") | |
229 int aw_first_pulse_off[2]; ///< index of first sample to which to | |
230 ///< apply AW-pulses, or -0xff if unset | |
231 int aw_next_pulse_off_cache; ///< the position (relative to start of the | |
232 ///< second block) at which pulses should | |
233 ///< start to be positioned, serves as a | |
234 ///< cache for pitch-adaptive window pulses | |
235 ///< between blocks | |
236 | |
237 int frame_cntr; ///< current frame index [0 - 0xFFFE]; is | |
238 ///< only used for comfort noise in #pRNG() | |
239 float gain_pred_err[6]; ///< cache for gain prediction | |
240 float excitation_history[MAX_SIGNAL_HISTORY]; | |
241 ///< cache of the signal of previous | |
242 ///< superframes, used as a history for | |
243 ///< signal generation | |
244 float synth_history[MAX_LSPS]; ///< see #excitation_history | |
245 /** | |
246 * @} | |
247 */ | |
248 } WMAVoiceContext; | |
249 | |
250 /** | |
251 * Sets up the variable bit mode (VBM) tree from container extradata. | |
252 * @param gb bit I/O context. | |
253 * The bit context (s->gb) should be loaded with byte 23-46 of the | |
254 * container extradata (i.e. the ones containing the VBM tree). | |
255 * @param vbm_tree pointer to array to which the decoded VBM tree will be | |
256 * written. | |
257 * @return 0 on success, <0 on error. | |
258 */ | |
259 static av_cold int decode_vbmtree(GetBitContext *gb, int8_t vbm_tree[25]) | |
260 { | |
261 static const uint8_t bits[] = { | |
262 2, 2, 2, 4, 4, 4, | |
263 6, 6, 6, 8, 8, 8, | |
264 10, 10, 10, 12, 12, 12, | |
265 14, 14, 14, 14 | |
266 }; | |
267 static const uint16_t codes[] = { | |
268 0x0000, 0x0001, 0x0002, // 00/01/10 | |
269 0x000c, 0x000d, 0x000e, // 11+00/01/10 | |
270 0x003c, 0x003d, 0x003e, // 1111+00/01/10 | |
271 0x00fc, 0x00fd, 0x00fe, // 111111+00/01/10 | |
272 0x03fc, 0x03fd, 0x03fe, // 11111111+00/01/10 | |
273 0x0ffc, 0x0ffd, 0x0ffe, // 1111111111+00/01/10 | |
274 0x3ffc, 0x3ffd, 0x3ffe, 0x3fff // 111111111111+xx | |
275 }; | |
276 int cntr[8], n, res; | |
277 | |
278 memset(vbm_tree, 0xff, sizeof(vbm_tree)); | |
279 memset(cntr, 0, sizeof(cntr)); | |
280 for (n = 0; n < 17; n++) { | |
281 res = get_bits(gb, 3); | |
282 if (cntr[res] > 3) // should be >= 3 + (res == 7)) | |
283 return -1; | |
284 vbm_tree[res * 3 + cntr[res]++] = n; | |
285 } | |
286 INIT_VLC_STATIC(&frame_type_vlc, VLC_NBITS, sizeof(bits), | |
287 bits, 1, 1, codes, 2, 2, 132); | |
288 return 0; | |
289 } | |
290 | |
291 /** | |
292 * Set up decoder with parameters from demuxer (extradata etc.). | |
293 */ | |
294 static av_cold int wmavoice_decode_init(AVCodecContext *ctx) | |
295 { | |
296 int n, flags, pitch_range, lsp16_flag; | |
297 WMAVoiceContext *s = ctx->priv_data; | |
298 | |
299 /** | |
300 * Extradata layout: | |
301 * - byte 0-18: WMAPro-in-WMAVoice extradata (see wmaprodec.c), | |
302 * - byte 19-22: flags field (annoyingly in LE; see below for known | |
303 * values), | |
304 * - byte 23-46: variable bitmode tree (really just 17 * 3 bits, | |
305 * rest is 0). | |
306 */ | |
307 if (ctx->extradata_size != 46) { | |
308 av_log(ctx, AV_LOG_ERROR, | |
309 "Invalid extradata size %d (should be 46)\n", | |
310 ctx->extradata_size); | |
311 return -1; | |
312 } | |
313 flags = AV_RL32(ctx->extradata + 18); | |
314 s->spillover_bitsize = 3 + av_ceil_log2(ctx->block_align); | |
315 s->do_apf = flags & 0x1; | |
316 s->lsp_q_mode = !!(flags & 0x2000); | |
317 s->lsp_def_mode = !!(flags & 0x4000); | |
318 lsp16_flag = flags & 0x1000; | |
319 if (lsp16_flag) { | |
320 s->lsps = 16; | |
321 s->frame_lsp_bitsize = 34; | |
322 s->sframe_lsp_bitsize = 60; | |
323 } else { | |
324 s->lsps = 10; | |
325 s->frame_lsp_bitsize = 24; | |
326 s->sframe_lsp_bitsize = 48; | |
327 } | |
328 for (n = 0; n < s->lsps; n++) | |
329 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
330 | |
331 init_get_bits(&s->gb, ctx->extradata + 22, (ctx->extradata_size - 22) << 3); | |
332 if (decode_vbmtree(&s->gb, s->vbm_tree) < 0) { | |
333 av_log(ctx, AV_LOG_ERROR, "Invalid VBM tree; broken extradata?\n"); | |
334 return -1; | |
335 } | |
336 | |
337 s->min_pitch_val = ((ctx->sample_rate << 8) / 400 + 50) >> 8; | |
338 s->max_pitch_val = ((ctx->sample_rate << 8) * 37 / 2000 + 50) >> 8; | |
339 pitch_range = s->max_pitch_val - s->min_pitch_val; | |
340 s->pitch_nbits = av_ceil_log2(pitch_range); | |
341 s->last_pitch_val = 40; | |
342 s->last_acb_type = ACB_TYPE_NONE; | |
343 s->history_nsamples = s->max_pitch_val + 8; | |
344 | |
345 if (s->min_pitch_val < 1 || s->history_nsamples > MAX_SIGNAL_HISTORY) { | |
346 int min_sr = ((((1 << 8) - 50) * 400) + 0xFF) >> 8, | |
347 max_sr = ((((MAX_SIGNAL_HISTORY - 8) << 8) + 205) * 2000 / 37) >> 8; | |
348 | |
349 av_log(ctx, AV_LOG_ERROR, | |
350 "Unsupported samplerate %d (min=%d, max=%d)\n", | |
351 ctx->sample_rate, min_sr, max_sr); // 322-22097 Hz | |
352 | |
353 return -1; | |
354 } | |
355 | |
356 s->block_conv_table[0] = s->min_pitch_val; | |
357 s->block_conv_table[1] = (pitch_range * 25) >> 6; | |
358 s->block_conv_table[2] = (pitch_range * 44) >> 6; | |
359 s->block_conv_table[3] = s->max_pitch_val - 1; | |
360 s->block_delta_pitch_hrange = (pitch_range >> 3) & ~0xF; | |
361 s->block_delta_pitch_nbits = 1 + av_ceil_log2(s->block_delta_pitch_hrange); | |
362 s->block_pitch_range = s->block_conv_table[2] + | |
363 s->block_conv_table[3] + 1 + | |
364 2 * (s->block_conv_table[1] - 2 * s->min_pitch_val); | |
365 s->block_pitch_nbits = av_ceil_log2(s->block_pitch_range); | |
366 | |
367 ctx->sample_fmt = SAMPLE_FMT_FLT; | |
368 | |
369 return 0; | |
370 } | |
371 | |
372 /** | |
373 * Dequantize LSPs | |
374 * @param lsps output pointer to the array that will hold the LSPs | |
375 * @param num number of LSPs to be dequantized | |
376 * @param values quantized values, contains n_stages values | |
377 * @param sizes range (i.e. max value) of each quantized value | |
378 * @param n_stages number of dequantization runs | |
379 * @param table dequantization table to be used | |
380 * @param mul_q LSF multiplier | |
381 * @param base_q base (lowest) LSF values | |
382 */ | |
383 static void dequant_lsps(double *lsps, int num, | |
384 const uint16_t *values, | |
385 const uint16_t *sizes, | |
386 int n_stages, const uint8_t *table, | |
387 const double *mul_q, | |
388 const double *base_q) | |
389 { | |
390 int n, m; | |
391 | |
392 memset(lsps, 0, num * sizeof(*lsps)); | |
393 for (n = 0; n < n_stages; n++) { | |
394 const uint8_t *t_off = &table[values[n] * num]; | |
395 double base = base_q[n], mul = mul_q[n]; | |
396 | |
397 for (m = 0; m < num; m++) | |
398 lsps[m] += base + mul * t_off[m]; | |
399 | |
400 table += sizes[n] * num; | |
401 } | |
402 } | |
403 | |
404 /** | |
405 * @defgroup lsp_dequant LSP dequantization routines | |
406 * LSP dequantization routines, for 10/16LSPs and independent/residual coding. | |
407 * @note we assume enough bits are available, caller should check. | |
408 * lsp10i() consumes 24 bits; lsp10r() consumes an additional 24 bits; | |
409 * lsp16i() consumes 34 bits; lsp16r() consumes an additional 26 bits. | |
410 * @{ | |
411 */ | |
412 /** | |
413 * Parse 10 independently-coded LSPs. | |
414 */ | |
415 static void dequant_lsp10i(GetBitContext *gb, double *lsps) | |
416 { | |
417 static const uint16_t vec_sizes[4] = { 256, 64, 32, 32 }; | |
418 static const double mul_lsf[4] = { | |
419 5.2187144800e-3, 1.4626986422e-3, | |
420 9.6179549166e-4, 1.1325736225e-3 | |
421 }; | |
422 static const double base_lsf[4] = { | |
423 M_PI * -2.15522e-1, M_PI * -6.1646e-2, | |
424 M_PI * -3.3486e-2, M_PI * -5.7408e-2 | |
425 }; | |
426 uint16_t v[4]; | |
427 | |
428 v[0] = get_bits(gb, 8); | |
429 v[1] = get_bits(gb, 6); | |
430 v[2] = get_bits(gb, 5); | |
431 v[3] = get_bits(gb, 5); | |
432 | |
433 dequant_lsps(lsps, 10, v, vec_sizes, 4, wmavoice_dq_lsp10i, | |
434 mul_lsf, base_lsf); | |
435 } | |
436 | |
437 /** | |
438 * Parse 10 independently-coded LSPs, and then derive the tables to | |
439 * generate LSPs for the other frames from them (residual coding). | |
440 */ | |
441 static void dequant_lsp10r(GetBitContext *gb, | |
442 double *i_lsps, const double *old, | |
443 double *a1, double *a2, int q_mode) | |
444 { | |
445 static const uint16_t vec_sizes[3] = { 128, 64, 64 }; | |
446 static const double mul_lsf[3] = { | |
447 2.5807601174e-3, 1.2354460219e-3, 1.1763821673e-3 | |
448 }; | |
449 static const double base_lsf[3] = { | |
450 M_PI * -1.07448e-1, M_PI * -5.2706e-2, M_PI * -5.1634e-2 | |
451 }; | |
452 const float (*ipol_tab)[2][10] = q_mode ? | |
453 wmavoice_lsp10_intercoeff_b : wmavoice_lsp10_intercoeff_a; | |
454 uint16_t interpol, v[3]; | |
455 int n; | |
456 | |
457 dequant_lsp10i(gb, i_lsps); | |
458 | |
459 interpol = get_bits(gb, 5); | |
460 v[0] = get_bits(gb, 7); | |
461 v[1] = get_bits(gb, 6); | |
462 v[2] = get_bits(gb, 6); | |
463 | |
464 for (n = 0; n < 10; n++) { | |
465 double delta = old[n] - i_lsps[n]; | |
466 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
467 a1[10 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
468 } | |
469 | |
470 dequant_lsps(a2, 20, v, vec_sizes, 3, wmavoice_dq_lsp10r, | |
471 mul_lsf, base_lsf); | |
472 } | |
473 | |
474 /** | |
475 * Parse 16 independently-coded LSPs. | |
476 */ | |
477 static void dequant_lsp16i(GetBitContext *gb, double *lsps) | |
478 { | |
479 static const uint16_t vec_sizes[5] = { 256, 64, 128, 64, 128 }; | |
480 static const double mul_lsf[5] = { | |
481 3.3439586280e-3, 6.9908173703e-4, | |
482 3.3216608306e-3, 1.0334960326e-3, | |
483 3.1899104283e-3 | |
484 }; | |
485 static const double base_lsf[5] = { | |
486 M_PI * -1.27576e-1, M_PI * -2.4292e-2, | |
487 M_PI * -1.28094e-1, M_PI * -3.2128e-2, | |
488 M_PI * -1.29816e-1 | |
489 }; | |
490 uint16_t v[5]; | |
491 | |
492 v[0] = get_bits(gb, 8); | |
493 v[1] = get_bits(gb, 6); | |
494 v[2] = get_bits(gb, 7); | |
495 v[3] = get_bits(gb, 6); | |
496 v[4] = get_bits(gb, 7); | |
497 | |
498 dequant_lsps( lsps, 5, v, vec_sizes, 2, | |
499 wmavoice_dq_lsp16i1, mul_lsf, base_lsf); | |
500 dequant_lsps(&lsps[5], 5, &v[2], &vec_sizes[2], 2, | |
501 wmavoice_dq_lsp16i2, &mul_lsf[2], &base_lsf[2]); | |
502 dequant_lsps(&lsps[10], 6, &v[4], &vec_sizes[4], 1, | |
503 wmavoice_dq_lsp16i3, &mul_lsf[4], &base_lsf[4]); | |
504 } | |
505 | |
506 /** | |
507 * Parse 16 independently-coded LSPs, and then derive the tables to | |
508 * generate LSPs for the other frames from them (residual coding). | |
509 */ | |
510 static void dequant_lsp16r(GetBitContext *gb, | |
511 double *i_lsps, const double *old, | |
512 double *a1, double *a2, int q_mode) | |
513 { | |
514 static const uint16_t vec_sizes[3] = { 128, 128, 128 }; | |
515 static const double mul_lsf[3] = { | |
516 1.2232979501e-3, 1.4062241527e-3, 1.6114744851e-3 | |
517 }; | |
518 static const double base_lsf[3] = { | |
519 M_PI * -5.5830e-2, M_PI * -5.2908e-2, M_PI * -5.4776e-2 | |
520 }; | |
521 const float (*ipol_tab)[2][16] = q_mode ? | |
522 wmavoice_lsp16_intercoeff_b : wmavoice_lsp16_intercoeff_a; | |
523 uint16_t interpol, v[3]; | |
524 int n; | |
525 | |
526 dequant_lsp16i(gb, i_lsps); | |
527 | |
528 interpol = get_bits(gb, 5); | |
529 v[0] = get_bits(gb, 7); | |
530 v[1] = get_bits(gb, 7); | |
531 v[2] = get_bits(gb, 7); | |
532 | |
533 for (n = 0; n < 16; n++) { | |
534 double delta = old[n] - i_lsps[n]; | |
535 a1[n] = ipol_tab[interpol][0][n] * delta + i_lsps[n]; | |
536 a1[16 + n] = ipol_tab[interpol][1][n] * delta + i_lsps[n]; | |
537 } | |
538 | |
539 dequant_lsps( a2, 10, v, vec_sizes, 1, | |
540 wmavoice_dq_lsp16r1, mul_lsf, base_lsf); | |
541 dequant_lsps(&a2[10], 10, &v[1], &vec_sizes[1], 1, | |
542 wmavoice_dq_lsp16r2, &mul_lsf[1], &base_lsf[1]); | |
543 dequant_lsps(&a2[20], 12, &v[2], &vec_sizes[2], 1, | |
544 wmavoice_dq_lsp16r3, &mul_lsf[2], &base_lsf[2]); | |
545 } | |
546 | |
547 /** | |
548 * @} | |
549 * @defgroup aw Pitch-adaptive window coding functions | |
550 * The next few functions are for pitch-adaptive window coding. | |
551 * @{ | |
552 */ | |
553 /** | |
554 * Parse the offset of the first pitch-adaptive window pulses, and | |
555 * the distribution of pulses between the two blocks in this frame. | |
556 * @param s WMA Voice decoding context private data | |
557 * @param gb bit I/O context | |
558 * @param pitch pitch for each block in this frame | |
559 */ | |
560 static void aw_parse_coords(WMAVoiceContext *s, GetBitContext *gb, | |
561 const int *pitch) | |
562 { | |
563 static const int16_t start_offset[94] = { | |
564 -11, -9, -7, -5, -3, -1, 1, 3, 5, 7, 9, 11, | |
565 13, 15, 18, 17, 19, 20, 21, 22, 23, 24, 25, 26, | |
566 27, 28, 29, 30, 31, 32, 33, 35, 37, 39, 41, 43, | |
567 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, | |
568 69, 71, 73, 75, 77, 79, 81, 83, 85, 87, 89, 91, | |
569 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113, 115, | |
570 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, | |
571 141, 143, 145, 147, 149, 151, 153, 155, 157, 159 | |
572 }; | |
573 int bits, offset; | |
574 | |
575 /* position of pulse */ | |
576 s->aw_idx_is_ext = 0; | |
577 if ((bits = get_bits(gb, 6)) >= 54) { | |
578 s->aw_idx_is_ext = 1; | |
579 bits += (bits - 54) * 3 + get_bits(gb, 2); | |
580 } | |
581 | |
582 /* for a repeated pulse at pulse_off with a pitch_lag of pitch[], count | |
583 * the distribution of the pulses in each block contained in this frame. */ | |
584 s->aw_pulse_range = FFMIN(pitch[0], pitch[1]) > 32 ? 24 : 16; | |
585 for (offset = start_offset[bits]; offset < 0; offset += pitch[0]) ; | |
586 s->aw_n_pulses[0] = (pitch[0] - 1 + MAX_FRAMESIZE / 2 - offset) / pitch[0]; | |
587 s->aw_first_pulse_off[0] = offset - s->aw_pulse_range / 2; | |
588 offset += s->aw_n_pulses[0] * pitch[0]; | |
589 s->aw_n_pulses[1] = (pitch[1] - 1 + MAX_FRAMESIZE - offset) / pitch[1]; | |
590 s->aw_first_pulse_off[1] = offset - (MAX_FRAMESIZE + s->aw_pulse_range) / 2; | |
591 | |
592 /* if continuing from a position before the block, reset position to | |
593 * start of block (when corrected for the range over which it can be | |
594 * spread in aw_pulse_set1()). */ | |
595 if (start_offset[bits] < MAX_FRAMESIZE / 2) { | |
596 while (s->aw_first_pulse_off[1] - pitch[1] + s->aw_pulse_range > 0) | |
597 s->aw_first_pulse_off[1] -= pitch[1]; | |
598 if (start_offset[bits] < 0) | |
599 while (s->aw_first_pulse_off[0] - pitch[0] + s->aw_pulse_range > 0) | |
600 s->aw_first_pulse_off[0] -= pitch[0]; | |
601 } | |
602 } | |
603 | |
604 /** | |
605 * Apply second set of pitch-adaptive window pulses. | |
606 * @param s WMA Voice decoding context private data | |
607 * @param gb bit I/O context | |
608 * @param block_idx block index in frame [0, 1] | |
609 * @param fcb structure containing fixed codebook vector info | |
610 */ | |
611 static void aw_pulse_set2(WMAVoiceContext *s, GetBitContext *gb, | |
612 int block_idx, AMRFixed *fcb) | |
613 { | |
614 uint16_t use_mask[7]; // only 5 are used, rest is padding | |
615 /* in this function, idx is the index in the 80-bit (+ padding) use_mask | |
616 * bit-array. Since use_mask consists of 16-bit values, the lower 4 bits | |
617 * of idx are the position of the bit within a particular item in the | |
618 * array (0 being the most significant bit, and 15 being the least | |
619 * significant bit), and the remainder (>> 4) is the index in the | |
620 * use_mask[]-array. This is faster and uses less memory than using a | |
621 * 80-byte/80-int array. */ | |
622 int pulse_off = s->aw_first_pulse_off[block_idx], | |
623 pulse_start, n, idx, range, aidx, start_off = 0; | |
624 | |
625 /* set offset of first pulse to within this block */ | |
626 if (s->aw_n_pulses[block_idx] > 0) | |
627 while (pulse_off + s->aw_pulse_range < 1) | |
628 pulse_off += fcb->pitch_lag; | |
629 | |
630 /* find range per pulse */ | |
631 if (s->aw_n_pulses[0] > 0) { | |
632 if (block_idx == 0) { | |
633 range = 32; | |
634 } else /* block_idx = 1 */ { | |
635 range = 8; | |
636 if (s->aw_n_pulses[block_idx] > 0) | |
637 pulse_off = s->aw_next_pulse_off_cache; | |
638 } | |
639 } else | |
640 range = 16; | |
641 pulse_start = s->aw_n_pulses[block_idx] > 0 ? pulse_off - range / 2 : 0; | |
642 | |
643 /* aw_pulse_set1() already applies pulses around pulse_off (to be exactly, | |
644 * in the range of [pulse_off, pulse_off + s->aw_pulse_range], and thus | |
645 * we exclude that range from being pulsed again in this function. */ | |
646 memset( use_mask, -1, 5 * sizeof(use_mask[0])); | |
647 memset(&use_mask[5], 0, 2 * sizeof(use_mask[0])); | |
648 if (s->aw_n_pulses[block_idx] > 0) | |
649 for (idx = pulse_off; idx < MAX_FRAMESIZE / 2; idx += fcb->pitch_lag) { | |
650 int excl_range = s->aw_pulse_range; // always 16 or 24 | |
651 uint16_t *use_mask_ptr = &use_mask[idx >> 4]; | |
652 int first_sh = 16 - (idx & 15); | |
653 *use_mask_ptr++ &= 0xFFFF << first_sh; | |
654 excl_range -= first_sh; | |
655 if (excl_range >= 16) { | |
656 *use_mask_ptr++ = 0; | |
657 *use_mask_ptr &= 0xFFFF >> (excl_range - 16); | |
658 } else | |
659 *use_mask_ptr &= 0xFFFF >> excl_range; | |
660 } | |
661 | |
662 /* find the 'aidx'th offset that is not excluded */ | |
663 aidx = get_bits(gb, s->aw_n_pulses[0] > 0 ? 5 - 2 * block_idx : 4); | |
664 for (n = 0; n <= aidx; pulse_start++) { | |
665 for (idx = pulse_start; idx < 0; idx += fcb->pitch_lag) ; | |
666 if (idx >= MAX_FRAMESIZE / 2) { // find from zero | |
667 if (use_mask[0]) idx = 0x0F; | |
668 else if (use_mask[1]) idx = 0x1F; | |
669 else if (use_mask[2]) idx = 0x2F; | |
670 else if (use_mask[3]) idx = 0x3F; | |
671 else if (use_mask[4]) idx = 0x4F; | |
672 else return; | |
673 idx -= av_log2_16bit(use_mask[idx >> 4]); | |
674 } | |
675 if (use_mask[idx >> 4] & (0x8000 >> (idx & 15))) { | |
676 use_mask[idx >> 4] &= ~(0x8000 >> (idx & 15)); | |
677 n++; | |
678 start_off = idx; | |
679 } | |
680 } | |
681 | |
682 fcb->x[fcb->n] = start_off; | |
683 fcb->y[fcb->n] = get_bits1(gb) ? -1.0 : 1.0; | |
684 fcb->n++; | |
685 | |
686 /* set offset for next block, relative to start of that block */ | |
687 n = (MAX_FRAMESIZE / 2 - start_off) % fcb->pitch_lag; | |
688 s->aw_next_pulse_off_cache = n ? fcb->pitch_lag - n : 0; | |
689 } | |
690 | |
691 /** | |
692 * Apply first set of pitch-adaptive window pulses. | |
693 * @param s WMA Voice decoding context private data | |
694 * @param gb bit I/O context | |
695 * @param block_idx block index in frame [0, 1] | |
696 * @param fcb storage location for fixed codebook pulse info | |
697 */ | |
698 static void aw_pulse_set1(WMAVoiceContext *s, GetBitContext *gb, | |
699 int block_idx, AMRFixed *fcb) | |
700 { | |
701 int val = get_bits(gb, 12 - 2 * (s->aw_idx_is_ext && !block_idx)); | |
702 float v; | |
703 | |
704 if (s->aw_n_pulses[block_idx] > 0) { | |
705 int n, v_mask, i_mask, sh, n_pulses; | |
706 | |
707 if (s->aw_pulse_range == 24) { // 3 pulses, 1:sign + 3:index each | |
708 n_pulses = 3; | |
709 v_mask = 8; | |
710 i_mask = 7; | |
711 sh = 4; | |
712 } else { // 4 pulses, 1:sign + 2:index each | |
713 n_pulses = 4; | |
714 v_mask = 4; | |
715 i_mask = 3; | |
716 sh = 3; | |
717 } | |
718 | |
719 for (n = n_pulses - 1; n >= 0; n--, val >>= sh) { | |
720 fcb->y[fcb->n] = (val & v_mask) ? -1.0 : 1.0; | |
721 fcb->x[fcb->n] = (val & i_mask) * n_pulses + n + | |
722 s->aw_first_pulse_off[block_idx]; | |
723 while (fcb->x[fcb->n] < 0) | |
724 fcb->x[fcb->n] += fcb->pitch_lag; | |
725 if (fcb->x[fcb->n] < MAX_FRAMESIZE / 2) | |
726 fcb->n++; | |
727 } | |
728 } else { | |
729 int num2 = (val & 0x1FF) >> 1, delta, idx; | |
730 | |
731 if (num2 < 1 * 79) { delta = 1; idx = num2 + 1; } | |
732 else if (num2 < 2 * 78) { delta = 3; idx = num2 + 1 - 1 * 77; } | |
733 else if (num2 < 3 * 77) { delta = 5; idx = num2 + 1 - 2 * 76; } | |
734 else { delta = 7; idx = num2 + 1 - 3 * 75; } | |
735 v = (val & 0x200) ? -1.0 : 1.0; | |
736 | |
737 fcb->no_repeat_mask |= 3 << fcb->n; | |
738 fcb->x[fcb->n] = idx - delta; | |
739 fcb->y[fcb->n] = v; | |
740 fcb->x[fcb->n + 1] = idx; | |
741 fcb->y[fcb->n + 1] = (val & 1) ? -v : v; | |
742 fcb->n += 2; | |
743 } | |
744 } | |
745 | |
746 /** | |
747 * @} | |
748 * | |
749 * Generate a random number from frame_cntr and block_idx, which will lief | |
750 * in the range [0, 1000 - block_size] (so it can be used as an index in a | |
751 * table of size 1000 of which you want to read block_size entries). | |
752 * | |
753 * @param frame_cntr current frame number | |
754 * @param block_num current block index | |
755 * @param block_size amount of entries we want to read from a table | |
756 * that has 1000 entries | |
11556 | 757 * @return a (non-)random number in the [0, 1000 - block_size] range. |
11123 | 758 */ |
759 static int pRNG(int frame_cntr, int block_num, int block_size) | |
760 { | |
761 /* array to simplify the calculation of z: | |
762 * y = (x % 9) * 5 + 6; | |
763 * z = (49995 * x) / y; | |
764 * Since y only has 9 values, we can remove the division by using a | |
765 * LUT and using FASTDIV-style divisions. For each of the 9 values | |
766 * of y, we can rewrite z as: | |
767 * z = x * (49995 / y) + x * ((49995 % y) / y) | |
768 * In this table, each col represents one possible value of y, the | |
769 * first number is 49995 / y, and the second is the FASTDIV variant | |
770 * of 49995 % y / y. */ | |
771 static const unsigned int div_tbl[9][2] = { | |
772 { 8332, 3 * 715827883U }, // y = 6 | |
773 { 4545, 0 * 390451573U }, // y = 11 | |
774 { 3124, 11 * 268435456U }, // y = 16 | |
775 { 2380, 15 * 204522253U }, // y = 21 | |
776 { 1922, 23 * 165191050U }, // y = 26 | |
777 { 1612, 23 * 138547333U }, // y = 31 | |
778 { 1388, 27 * 119304648U }, // y = 36 | |
779 { 1219, 16 * 104755300U }, // y = 41 | |
780 { 1086, 39 * 93368855U } // y = 46 | |
781 }; | |
782 unsigned int z, y, x = MUL16(block_num, 1877) + frame_cntr; | |
783 if (x >= 0xFFFF) x -= 0xFFFF; // max value of x is 8*1877+0xFFFE=0x13AA6, | |
784 // so this is effectively a modulo (%) | |
785 y = x - 9 * MULH(477218589, x); // x % 9 | |
786 z = (uint16_t) (x * div_tbl[y][0] + UMULH(x, div_tbl[y][1])); | |
787 // z = x * 49995 / (y * 5 + 6) | |
788 return z % (1000 - block_size); | |
789 } | |
790 | |
791 /** | |
792 * Parse hardcoded signal for a single block. | |
793 * @note see #synth_block(). | |
794 */ | |
795 static void synth_block_hardcoded(WMAVoiceContext *s, GetBitContext *gb, | |
796 int block_idx, int size, | |
797 const struct frame_type_desc *frame_desc, | |
798 float *excitation) | |
799 { | |
800 float gain; | |
801 int n, r_idx; | |
802 | |
803 assert(size <= MAX_FRAMESIZE); | |
804 | |
805 /* Set the offset from which we start reading wmavoice_std_codebook */ | |
806 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
807 r_idx = pRNG(s->frame_cntr, block_idx, size); | |
808 gain = s->silence_gain; | |
809 } else /* FCB_TYPE_HARDCODED */ { | |
810 r_idx = get_bits(gb, 8); | |
811 gain = wmavoice_gain_universal[get_bits(gb, 6)]; | |
812 } | |
813 | |
814 /* Clear gain prediction parameters */ | |
815 memset(s->gain_pred_err, 0, sizeof(s->gain_pred_err)); | |
816 | |
817 /* Apply gain to hardcoded codebook and use that as excitation signal */ | |
818 for (n = 0; n < size; n++) | |
819 excitation[n] = wmavoice_std_codebook[r_idx + n] * gain; | |
820 } | |
821 | |
822 /** | |
823 * Parse FCB/ACB signal for a single block. | |
824 * @note see #synth_block(). | |
825 */ | |
826 static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb, | |
827 int block_idx, int size, | |
828 int block_pitch_sh2, | |
829 const struct frame_type_desc *frame_desc, | |
830 float *excitation) | |
831 { | |
832 static const float gain_coeff[6] = { | |
833 0.8169, -0.06545, 0.1726, 0.0185, -0.0359, 0.0458 | |
834 }; | |
835 float pulses[MAX_FRAMESIZE / 2], pred_err, acb_gain, fcb_gain; | |
836 int n, idx, gain_weight; | |
837 AMRFixed fcb; | |
838 | |
839 assert(size <= MAX_FRAMESIZE / 2); | |
840 memset(pulses, 0, sizeof(*pulses) * size); | |
841 | |
842 fcb.pitch_lag = block_pitch_sh2 >> 2; | |
843 fcb.pitch_fac = 1.0; | |
844 fcb.no_repeat_mask = 0; | |
845 fcb.n = 0; | |
846 | |
847 /* For the other frame types, this is where we apply the innovation | |
848 * (fixed) codebook pulses of the speech signal. */ | |
849 if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
850 aw_pulse_set1(s, gb, block_idx, &fcb); | |
851 aw_pulse_set2(s, gb, block_idx, &fcb); | |
852 } else /* FCB_TYPE_EXC_PULSES */ { | |
853 int offset_nbits = 5 - frame_desc->log_n_blocks; | |
854 | |
855 fcb.no_repeat_mask = -1; | |
856 /* similar to ff_decode_10_pulses_35bits(), but with single pulses | |
857 * (instead of double) for a subset of pulses */ | |
858 for (n = 0; n < 5; n++) { | |
859 float sign; | |
860 int pos1, pos2; | |
861 | |
862 sign = get_bits1(gb) ? 1.0 : -1.0; | |
863 pos1 = get_bits(gb, offset_nbits); | |
864 fcb.x[fcb.n] = n + 5 * pos1; | |
865 fcb.y[fcb.n++] = sign; | |
866 if (n < frame_desc->dbl_pulses) { | |
867 pos2 = get_bits(gb, offset_nbits); | |
868 fcb.x[fcb.n] = n + 5 * pos2; | |
869 fcb.y[fcb.n++] = (pos1 < pos2) ? -sign : sign; | |
870 } | |
871 } | |
872 } | |
873 ff_set_fixed_vector(pulses, &fcb, 1.0, size); | |
874 | |
875 /* Calculate gain for adaptive & fixed codebook signal. | |
876 * see ff_amr_set_fixed_gain(). */ | |
877 idx = get_bits(gb, 7); | |
878 fcb_gain = expf(ff_dot_productf(s->gain_pred_err, gain_coeff, 6) - | |
879 5.2409161640 + wmavoice_gain_codebook_fcb[idx]); | |
880 acb_gain = wmavoice_gain_codebook_acb[idx]; | |
881 pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx], | |
882 -2.9957322736 /* log(0.05) */, | |
883 1.6094379124 /* log(5.0) */); | |
884 | |
885 gain_weight = 8 >> frame_desc->log_n_blocks; | |
886 memmove(&s->gain_pred_err[gain_weight], s->gain_pred_err, | |
887 sizeof(*s->gain_pred_err) * (6 - gain_weight)); | |
888 for (n = 0; n < gain_weight; n++) | |
889 s->gain_pred_err[n] = pred_err; | |
890 | |
891 /* Calculation of adaptive codebook */ | |
892 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
893 int len; | |
894 for (n = 0; n < size; n += len) { | |
895 int next_idx_sh16; | |
896 int abs_idx = block_idx * size + n; | |
897 int pitch_sh16 = (s->last_pitch_val << 16) + | |
898 s->pitch_diff_sh16 * abs_idx; | |
899 int pitch = (pitch_sh16 + 0x6FFF) >> 16; | |
900 int idx_sh16 = ((pitch << 16) - pitch_sh16) * 8 + 0x58000; | |
901 idx = idx_sh16 >> 16; | |
902 if (s->pitch_diff_sh16) { | |
903 if (s->pitch_diff_sh16 > 0) { | |
904 next_idx_sh16 = (idx_sh16) &~ 0xFFFF; | |
905 } else | |
906 next_idx_sh16 = (idx_sh16 + 0x10000) &~ 0xFFFF; | |
907 len = av_clip((idx_sh16 - next_idx_sh16) / s->pitch_diff_sh16 / 8, | |
908 1, size - n); | |
909 } else | |
910 len = size; | |
911 | |
912 ff_acelp_interpolatef(&excitation[n], &excitation[n - pitch], | |
913 wmavoice_ipol1_coeffs, 17, | |
914 idx, 9, len); | |
915 } | |
916 } else /* ACB_TYPE_HAMMING */ { | |
917 int block_pitch = block_pitch_sh2 >> 2; | |
918 idx = block_pitch_sh2 & 3; | |
919 if (idx) { | |
920 ff_acelp_interpolatef(excitation, &excitation[-block_pitch], | |
921 wmavoice_ipol2_coeffs, 4, | |
922 idx, 8, size); | |
923 } else | |
924 av_memcpy_backptr(excitation, sizeof(float) * block_pitch, | |
925 sizeof(float) * size); | |
926 } | |
927 | |
928 /* Interpolate ACB/FCB and use as excitation signal */ | |
929 ff_weighted_vector_sumf(excitation, excitation, pulses, | |
930 acb_gain, fcb_gain, size); | |
931 } | |
932 | |
933 /** | |
934 * Parse data in a single block. | |
935 * @note we assume enough bits are available, caller should check. | |
936 * | |
937 * @param s WMA Voice decoding context private data | |
938 * @param gb bit I/O context | |
939 * @param block_idx index of the to-be-read block | |
940 * @param size amount of samples to be read in this block | |
941 * @param block_pitch_sh2 pitch for this block << 2 | |
942 * @param lsps LSPs for (the end of) this frame | |
943 * @param prev_lsps LSPs for the last frame | |
944 * @param frame_desc frame type descriptor | |
945 * @param excitation target memory for the ACB+FCB interpolated signal | |
946 * @param synth target memory for the speech synthesis filter output | |
947 * @return 0 on success, <0 on error. | |
948 */ | |
949 static void synth_block(WMAVoiceContext *s, GetBitContext *gb, | |
950 int block_idx, int size, | |
951 int block_pitch_sh2, | |
952 const double *lsps, const double *prev_lsps, | |
953 const struct frame_type_desc *frame_desc, | |
954 float *excitation, float *synth) | |
955 { | |
956 double i_lsps[MAX_LSPS]; | |
957 float lpcs[MAX_LSPS]; | |
958 float fac; | |
959 int n; | |
960 | |
961 if (frame_desc->acb_type == ACB_TYPE_NONE) | |
962 synth_block_hardcoded(s, gb, block_idx, size, frame_desc, excitation); | |
963 else | |
964 synth_block_fcb_acb(s, gb, block_idx, size, block_pitch_sh2, | |
965 frame_desc, excitation); | |
966 | |
967 /* convert interpolated LSPs to LPCs */ | |
968 fac = (block_idx + 0.5) / frame_desc->n_blocks; | |
969 for (n = 0; n < s->lsps; n++) // LSF -> LSP | |
970 i_lsps[n] = cos(prev_lsps[n] + fac * (lsps[n] - prev_lsps[n])); | |
971 ff_acelp_lspd2lpc(i_lsps, lpcs, s->lsps >> 1); | |
972 | |
973 /* Speech synthesis */ | |
974 ff_celp_lp_synthesis_filterf(synth, lpcs, excitation, size, s->lsps); | |
975 } | |
976 | |
977 /** | |
978 * Synthesize output samples for a single frame. | |
979 * @note we assume enough bits are available, caller should check. | |
980 * | |
981 * @param ctx WMA Voice decoder context | |
982 * @param gb bit I/O context (s->gb or one for cross-packet superframes) | |
983 * @param samples pointer to output sample buffer, has space for at least 160 | |
984 * samples | |
985 * @param lsps LSP array | |
986 * @param prev_lsps array of previous frame's LSPs | |
987 * @param excitation target buffer for excitation signal | |
988 * @param synth target buffer for synthesized speech data | |
989 * @return 0 on success, <0 on error. | |
990 */ | |
991 static int synth_frame(AVCodecContext *ctx, GetBitContext *gb, | |
992 float *samples, | |
993 const double *lsps, const double *prev_lsps, | |
994 float *excitation, float *synth) | |
995 { | |
996 WMAVoiceContext *s = ctx->priv_data; | |
997 int n, n_blocks_x2, log_n_blocks_x2, cur_pitch_val; | |
998 int pitch[MAX_BLOCKS], last_block_pitch; | |
999 | |
1000 /* Parse frame type ("frame header"), see frame_descs */ | |
1001 int bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)], | |
1002 block_nsamples = MAX_FRAMESIZE / frame_descs[bd_idx].n_blocks; | |
1003 | |
1004 if (bd_idx < 0) { | |
1005 av_log(ctx, AV_LOG_ERROR, | |
1006 "Invalid frame type VLC code, skipping\n"); | |
1007 return -1; | |
1008 } | |
1009 | |
1010 /* Pitch calculation for ACB_TYPE_ASYMMETRIC ("pitch-per-frame") */ | |
1011 if (frame_descs[bd_idx].acb_type == ACB_TYPE_ASYMMETRIC) { | |
1012 /* Pitch is provided per frame, which is interpreted as the pitch of | |
1013 * the last sample of the last block of this frame. We can interpolate | |
1014 * the pitch of other blocks (and even pitch-per-sample) by gradually | |
1015 * incrementing/decrementing prev_frame_pitch to cur_pitch_val. */ | |
1016 n_blocks_x2 = frame_descs[bd_idx].n_blocks << 1; | |
1017 log_n_blocks_x2 = frame_descs[bd_idx].log_n_blocks + 1; | |
1018 cur_pitch_val = s->min_pitch_val + get_bits(gb, s->pitch_nbits); | |
1019 cur_pitch_val = FFMIN(cur_pitch_val, s->max_pitch_val - 1); | |
1020 if (s->last_acb_type == ACB_TYPE_NONE || | |
1021 20 * abs(cur_pitch_val - s->last_pitch_val) > | |
1022 (cur_pitch_val + s->last_pitch_val)) | |
1023 s->last_pitch_val = cur_pitch_val; | |
1024 | |
1025 /* pitch per block */ | |
1026 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1027 int fac = n * 2 + 1; | |
1028 | |
1029 pitch[n] = (MUL16(fac, cur_pitch_val) + | |
1030 MUL16((n_blocks_x2 - fac), s->last_pitch_val) + | |
1031 frame_descs[bd_idx].n_blocks) >> log_n_blocks_x2; | |
1032 } | |
1033 | |
1034 /* "pitch-diff-per-sample" for calculation of pitch per sample */ | |
1035 s->pitch_diff_sh16 = | |
1036 ((cur_pitch_val - s->last_pitch_val) << 16) / MAX_FRAMESIZE; | |
1037 } | |
1038 | |
1039 /* Global gain (if silence) and pitch-adaptive window coordinates */ | |
1040 switch (frame_descs[bd_idx].fcb_type) { | |
1041 case FCB_TYPE_SILENCE: | |
1042 s->silence_gain = wmavoice_gain_silence[get_bits(gb, 8)]; | |
1043 break; | |
1044 case FCB_TYPE_AW_PULSES: | |
1045 aw_parse_coords(s, gb, pitch); | |
1046 break; | |
1047 } | |
1048 | |
1049 for (n = 0; n < frame_descs[bd_idx].n_blocks; n++) { | |
1050 int bl_pitch_sh2; | |
1051 | |
1052 /* Pitch calculation for ACB_TYPE_HAMMING ("pitch-per-block") */ | |
1053 switch (frame_descs[bd_idx].acb_type) { | |
1054 case ACB_TYPE_HAMMING: { | |
1055 /* Pitch is given per block. Per-block pitches are encoded as an | |
1056 * absolute value for the first block, and then delta values | |
1057 * relative to this value) for all subsequent blocks. The scale of | |
1058 * this pitch value is semi-logaritmic compared to its use in the | |
1059 * decoder, so we convert it to normal scale also. */ | |
1060 int block_pitch, | |
1061 t1 = (s->block_conv_table[1] - s->block_conv_table[0]) << 2, | |
1062 t2 = (s->block_conv_table[2] - s->block_conv_table[1]) << 1, | |
1063 t3 = s->block_conv_table[3] - s->block_conv_table[2] + 1; | |
1064 | |
1065 if (n == 0) { | |
1066 block_pitch = get_bits(gb, s->block_pitch_nbits); | |
1067 } else | |
1068 block_pitch = last_block_pitch - s->block_delta_pitch_hrange + | |
1069 get_bits(gb, s->block_delta_pitch_nbits); | |
1070 /* Convert last_ so that any next delta is within _range */ | |
1071 last_block_pitch = av_clip(block_pitch, | |
1072 s->block_delta_pitch_hrange, | |
1073 s->block_pitch_range - | |
1074 s->block_delta_pitch_hrange); | |
1075 | |
1076 /* Convert semi-log-style scale back to normal scale */ | |
1077 if (block_pitch < t1) { | |
1078 bl_pitch_sh2 = (s->block_conv_table[0] << 2) + block_pitch; | |
1079 } else { | |
1080 block_pitch -= t1; | |
1081 if (block_pitch < t2) { | |
1082 bl_pitch_sh2 = | |
1083 (s->block_conv_table[1] << 2) + (block_pitch << 1); | |
1084 } else { | |
1085 block_pitch -= t2; | |
1086 if (block_pitch < t3) { | |
1087 bl_pitch_sh2 = | |
1088 (s->block_conv_table[2] + block_pitch) << 2; | |
1089 } else | |
1090 bl_pitch_sh2 = s->block_conv_table[3] << 2; | |
1091 } | |
1092 } | |
1093 pitch[n] = bl_pitch_sh2 >> 2; | |
1094 break; | |
1095 } | |
1096 | |
1097 case ACB_TYPE_ASYMMETRIC: { | |
1098 bl_pitch_sh2 = pitch[n] << 2; | |
1099 break; | |
1100 } | |
1101 | |
1102 default: // ACB_TYPE_NONE has no pitch | |
1103 bl_pitch_sh2 = 0; | |
1104 break; | |
1105 } | |
1106 | |
1107 synth_block(s, gb, n, block_nsamples, bl_pitch_sh2, | |
1108 lsps, prev_lsps, &frame_descs[bd_idx], | |
1109 &excitation[n * block_nsamples], | |
1110 &synth[n * block_nsamples]); | |
1111 } | |
1112 | |
1113 /* Averaging projection filter, if applicable. Else, just copy samples | |
1114 * from synthesis buffer */ | |
1115 if (s->do_apf) { | |
1116 // FIXME this is where APF would take place, currently not implemented | |
1117 av_log_missing_feature(ctx, "APF", 0); | |
1118 s->do_apf = 0; | |
1119 } //else | |
1120 for (n = 0; n < 160; n++) | |
1121 samples[n] = av_clipf(synth[n], -1.0, 1.0); | |
1122 | |
1123 /* Cache values for next frame */ | |
1124 s->frame_cntr++; | |
1125 if (s->frame_cntr >= 0xFFFF) s->frame_cntr -= 0xFFFF; // i.e. modulo (%) | |
1126 s->last_acb_type = frame_descs[bd_idx].acb_type; | |
1127 switch (frame_descs[bd_idx].acb_type) { | |
1128 case ACB_TYPE_NONE: | |
1129 s->last_pitch_val = 0; | |
1130 break; | |
1131 case ACB_TYPE_ASYMMETRIC: | |
1132 s->last_pitch_val = cur_pitch_val; | |
1133 break; | |
1134 case ACB_TYPE_HAMMING: | |
1135 s->last_pitch_val = pitch[frame_descs[bd_idx].n_blocks - 1]; | |
1136 break; | |
1137 } | |
1138 | |
1139 return 0; | |
1140 } | |
1141 | |
1142 /** | |
1143 * Ensure minimum value for first item, maximum value for last value, | |
1144 * proper spacing between each value and proper ordering. | |
1145 * | |
1146 * @param lsps array of LSPs | |
1147 * @param num size of LSP array | |
1148 * | |
1149 * @note basically a double version of #ff_acelp_reorder_lsf(), might be | |
1150 * useful to put in a generic location later on. Parts are also | |
1151 * present in #ff_set_min_dist_lsf() + #ff_sort_nearly_sorted_floats(), | |
1152 * which is in float. | |
1153 */ | |
1154 static void stabilize_lsps(double *lsps, int num) | |
1155 { | |
1156 int n, m, l; | |
1157 | |
1158 /* set minimum value for first, maximum value for last and minimum | |
1159 * spacing between LSF values. | |
1160 * Very similar to ff_set_min_dist_lsf(), but in double. */ | |
1161 lsps[0] = FFMAX(lsps[0], 0.0015 * M_PI); | |
1162 for (n = 1; n < num; n++) | |
1163 lsps[n] = FFMAX(lsps[n], lsps[n - 1] + 0.0125 * M_PI); | |
1164 lsps[num - 1] = FFMIN(lsps[num - 1], 0.9985 * M_PI); | |
1165 | |
1166 /* reorder (looks like one-time / non-recursed bubblesort). | |
1167 * Very similar to ff_sort_nearly_sorted_floats(), but in double. */ | |
1168 for (n = 1; n < num; n++) { | |
1169 if (lsps[n] < lsps[n - 1]) { | |
1170 for (m = 1; m < num; m++) { | |
1171 double tmp = lsps[m]; | |
1172 for (l = m - 1; l >= 0; l--) { | |
1173 if (lsps[l] <= tmp) break; | |
1174 lsps[l + 1] = lsps[l]; | |
1175 } | |
1176 lsps[l + 1] = tmp; | |
1177 } | |
1178 break; | |
1179 } | |
1180 } | |
1181 } | |
1182 | |
1183 /** | |
1184 * Test if there's enough bits to read 1 superframe. | |
1185 * | |
1186 * @param orig_gb bit I/O context used for reading. This function | |
1187 * does not modify the state of the bitreader; it | |
1188 * only uses it to copy the current stream position | |
1189 * @param s WMA Voice decoding context private data | |
11556 | 1190 * @return -1 if unsupported, 1 on not enough bits or 0 if OK. |
11123 | 1191 */ |
1192 static int check_bits_for_superframe(GetBitContext *orig_gb, | |
1193 WMAVoiceContext *s) | |
1194 { | |
1195 GetBitContext s_gb, *gb = &s_gb; | |
1196 int n, need_bits, bd_idx; | |
1197 const struct frame_type_desc *frame_desc; | |
1198 | |
1199 /* initialize a copy */ | |
1200 init_get_bits(gb, orig_gb->buffer, orig_gb->size_in_bits); | |
1201 skip_bits_long(gb, get_bits_count(orig_gb)); | |
1202 assert(get_bits_left(gb) == get_bits_left(orig_gb)); | |
1203 | |
1204 /* superframe header */ | |
1205 if (get_bits_left(gb) < 14) | |
1206 return 1; | |
1207 if (!get_bits1(gb)) | |
1208 return -1; // WMAPro-in-WMAVoice superframe | |
1209 if (get_bits1(gb)) skip_bits(gb, 12); // number of samples in superframe | |
1210 if (s->has_residual_lsps) { // residual LSPs (for all frames) | |
1211 if (get_bits_left(gb) < s->sframe_lsp_bitsize) | |
1212 return 1; | |
1213 skip_bits_long(gb, s->sframe_lsp_bitsize); | |
1214 } | |
1215 | |
1216 /* frames */ | |
1217 for (n = 0; n < MAX_FRAMES; n++) { | |
1218 int aw_idx_is_ext = 0; | |
1219 | |
1220 if (!s->has_residual_lsps) { // independent LSPs (per-frame) | |
1221 if (get_bits_left(gb) < s->frame_lsp_bitsize) return 1; | |
1222 skip_bits_long(gb, s->frame_lsp_bitsize); | |
1223 } | |
1224 bd_idx = s->vbm_tree[get_vlc2(gb, frame_type_vlc.table, 6, 3)]; | |
1225 if (bd_idx < 0) | |
1226 return -1; // invalid frame type VLC code | |
1227 frame_desc = &frame_descs[bd_idx]; | |
1228 if (frame_desc->acb_type == ACB_TYPE_ASYMMETRIC) { | |
1229 if (get_bits_left(gb) < s->pitch_nbits) | |
1230 return 1; | |
1231 skip_bits_long(gb, s->pitch_nbits); | |
1232 } | |
1233 if (frame_desc->fcb_type == FCB_TYPE_SILENCE) { | |
1234 skip_bits(gb, 8); | |
1235 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1236 int tmp = get_bits(gb, 6); | |
1237 if (tmp >= 0x36) { | |
1238 skip_bits(gb, 2); | |
1239 aw_idx_is_ext = 1; | |
1240 } | |
1241 } | |
1242 | |
1243 /* blocks */ | |
1244 if (frame_desc->acb_type == ACB_TYPE_HAMMING) { | |
1245 need_bits = s->block_pitch_nbits + | |
1246 (frame_desc->n_blocks - 1) * s->block_delta_pitch_nbits; | |
1247 } else if (frame_desc->fcb_type == FCB_TYPE_AW_PULSES) { | |
1248 need_bits = 2 * !aw_idx_is_ext; | |
1249 } else | |
1250 need_bits = 0; | |
1251 need_bits += frame_desc->frame_size; | |
1252 if (get_bits_left(gb) < need_bits) | |
1253 return 1; | |
1254 skip_bits_long(gb, need_bits); | |
1255 } | |
1256 | |
1257 return 0; | |
1258 } | |
1259 | |
1260 /** | |
1261 * Synthesize output samples for a single superframe. If we have any data | |
1262 * cached in s->sframe_cache, that will be used instead of whatever is loaded | |
1263 * in s->gb. | |
1264 * | |
1265 * WMA Voice superframes contain 3 frames, each containing 160 audio samples, | |
1266 * to give a total of 480 samples per frame. See #synth_frame() for frame | |
1267 * parsing. In addition to 3 frames, superframes can also contain the LSPs | |
1268 * (if these are globally specified for all frames (residually); they can | |
1269 * also be specified individually per-frame. See the s->has_residual_lsps | |
1270 * option), and can specify the number of samples encoded in this superframe | |
1271 * (if less than 480), usually used to prevent blanks at track boundaries. | |
1272 * | |
1273 * @param ctx WMA Voice decoder context | |
1274 * @param samples pointer to output buffer for voice samples | |
1275 * @param data_size pointer containing the size of #samples on input, and the | |
1276 * amount of #samples filled on output | |
1277 * @return 0 on success, <0 on error or 1 if there was not enough data to | |
1278 * fully parse the superframe | |
1279 */ | |
1280 static int synth_superframe(AVCodecContext *ctx, | |
1281 float *samples, int *data_size) | |
1282 { | |
1283 WMAVoiceContext *s = ctx->priv_data; | |
1284 GetBitContext *gb = &s->gb, s_gb; | |
1285 int n, res, n_samples = 480; | |
1286 double lsps[MAX_FRAMES][MAX_LSPS]; | |
1287 const double *mean_lsf = s->lsps == 16 ? | |
1288 wmavoice_mean_lsf16[s->lsp_def_mode] : wmavoice_mean_lsf10[s->lsp_def_mode]; | |
1289 float excitation[MAX_SIGNAL_HISTORY + MAX_SFRAMESIZE + 12]; | |
1290 float synth[MAX_LSPS + MAX_SFRAMESIZE]; | |
1291 | |
1292 memcpy(synth, s->synth_history, | |
1293 s->lsps * sizeof(*synth)); | |
1294 memcpy(excitation, s->excitation_history, | |
1295 s->history_nsamples * sizeof(*excitation)); | |
1296 | |
1297 if (s->sframe_cache_size > 0) { | |
1298 gb = &s_gb; | |
1299 init_get_bits(gb, s->sframe_cache, s->sframe_cache_size); | |
1300 s->sframe_cache_size = 0; | |
1301 } | |
1302 | |
1303 if ((res = check_bits_for_superframe(gb, s)) == 1) return 1; | |
1304 | |
1305 /* First bit is speech/music bit, it differentiates between WMAVoice | |
1306 * speech samples (the actual codec) and WMAVoice music samples, which | |
1307 * are really WMAPro-in-WMAVoice-superframes. I've never seen those in | |
1308 * the wild yet. */ | |
1309 if (!get_bits1(gb)) { | |
1310 av_log_missing_feature(ctx, "WMAPro-in-WMAVoice support", 1); | |
1311 return -1; | |
1312 } | |
1313 | |
1314 /* (optional) nr. of samples in superframe; always <= 480 and >= 0 */ | |
1315 if (get_bits1(gb)) { | |
1316 if ((n_samples = get_bits(gb, 12)) > 480) { | |
1317 av_log(ctx, AV_LOG_ERROR, | |
1318 "Superframe encodes >480 samples (%d), not allowed\n", | |
1319 n_samples); | |
1320 return -1; | |
1321 } | |
1322 } | |
1323 /* Parse LSPs, if global for the superframe (can also be per-frame). */ | |
1324 if (s->has_residual_lsps) { | |
1325 double prev_lsps[MAX_LSPS], a1[MAX_LSPS * 2], a2[MAX_LSPS * 2]; | |
1326 | |
1327 for (n = 0; n < s->lsps; n++) | |
1328 prev_lsps[n] = s->prev_lsps[n] - mean_lsf[n]; | |
1329 | |
1330 if (s->lsps == 10) { | |
1331 dequant_lsp10r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1332 } else /* s->lsps == 16 */ | |
1333 dequant_lsp16r(gb, lsps[2], prev_lsps, a1, a2, s->lsp_q_mode); | |
1334 | |
1335 for (n = 0; n < s->lsps; n++) { | |
1336 lsps[0][n] = mean_lsf[n] + (a1[n] - a2[n * 2]); | |
1337 lsps[1][n] = mean_lsf[n] + (a1[s->lsps + n] - a2[n * 2 + 1]); | |
1338 lsps[2][n] += mean_lsf[n]; | |
1339 } | |
1340 for (n = 0; n < 3; n++) | |
1341 stabilize_lsps(lsps[n], s->lsps); | |
1342 } | |
1343 | |
1344 /* Parse frames, optionally preceeded by per-frame (independent) LSPs. */ | |
1345 for (n = 0; n < 3; n++) { | |
1346 if (!s->has_residual_lsps) { | |
1347 int m; | |
1348 | |
1349 if (s->lsps == 10) { | |
1350 dequant_lsp10i(gb, lsps[n]); | |
1351 } else /* s->lsps == 16 */ | |
1352 dequant_lsp16i(gb, lsps[n]); | |
1353 | |
1354 for (m = 0; m < s->lsps; m++) | |
1355 lsps[n][m] += mean_lsf[m]; | |
1356 stabilize_lsps(lsps[n], s->lsps); | |
1357 } | |
1358 | |
1359 if ((res = synth_frame(ctx, gb, | |
1360 &samples[n * MAX_FRAMESIZE], | |
1361 lsps[n], n == 0 ? s->prev_lsps : lsps[n - 1], | |
1362 &excitation[s->history_nsamples + n * MAX_FRAMESIZE], | |
1363 &synth[s->lsps + n * MAX_FRAMESIZE]))) | |
1364 return res; | |
1365 } | |
1366 | |
1367 /* Statistics? FIXME - we don't check for length, a slight overrun | |
1368 * will be caught by internal buffer padding, and anything else | |
1369 * will be skipped, not read. */ | |
1370 if (get_bits1(gb)) { | |
1371 res = get_bits(gb, 4); | |
1372 skip_bits(gb, 10 * (res + 1)); | |
1373 } | |
1374 | |
1375 /* Specify nr. of output samples */ | |
1376 *data_size = n_samples * sizeof(float); | |
1377 | |
1378 /* Update history */ | |
1379 memcpy(s->prev_lsps, lsps[2], | |
1380 s->lsps * sizeof(*s->prev_lsps)); | |
1381 memcpy(s->synth_history, &synth[MAX_SFRAMESIZE], | |
1382 s->lsps * sizeof(*synth)); | |
1383 memcpy(s->excitation_history, &excitation[MAX_SFRAMESIZE], | |
1384 s->history_nsamples * sizeof(*excitation)); | |
1385 | |
1386 return 0; | |
1387 } | |
1388 | |
1389 /** | |
1390 * Parse the packet header at the start of each packet (input data to this | |
1391 * decoder). | |
1392 * | |
1393 * @param s WMA Voice decoding context private data | |
11556 | 1394 * @return 1 if not enough bits were available, or 0 on success. |
11123 | 1395 */ |
1396 static int parse_packet_header(WMAVoiceContext *s) | |
1397 { | |
1398 GetBitContext *gb = &s->gb; | |
1399 unsigned int res; | |
1400 | |
1401 if (get_bits_left(gb) < 11) | |
1402 return 1; | |
1403 skip_bits(gb, 4); // packet sequence number | |
1404 s->has_residual_lsps = get_bits1(gb); | |
1405 do { | |
1406 res = get_bits(gb, 6); // number of superframes per packet | |
1407 // (minus first one if there is spillover) | |
1408 if (get_bits_left(gb) < 6 * (res == 0x3F) + s->spillover_bitsize) | |
1409 return 1; | |
1410 } while (res == 0x3F); | |
1411 s->spillover_nbits = get_bits(gb, s->spillover_bitsize); | |
1412 | |
1413 return 0; | |
1414 } | |
1415 | |
1416 /** | |
1417 * Copy (unaligned) bits from gb/data/size to pb. | |
1418 * | |
1419 * @param pb target buffer to copy bits into | |
1420 * @param data source buffer to copy bits from | |
1421 * @param size size of the source data, in bytes | |
1422 * @param gb bit I/O context specifying the current position in the source. | |
1423 * data. This function might use this to align the bit position to | |
1424 * a whole-byte boundary before calling #ff_copy_bits() on aligned | |
1425 * source data | |
1426 * @param nbits the amount of bits to copy from source to target | |
1427 * | |
1428 * @note after calling this function, the current position in the input bit | |
1429 * I/O context is undefined. | |
1430 */ | |
1431 static void copy_bits(PutBitContext *pb, | |
1432 const uint8_t *data, int size, | |
1433 GetBitContext *gb, int nbits) | |
1434 { | |
1435 int rmn_bytes, rmn_bits; | |
1436 | |
1437 rmn_bits = rmn_bytes = get_bits_left(gb); | |
1438 if (rmn_bits < nbits) | |
1439 return; | |
1440 rmn_bits &= 7; rmn_bytes >>= 3; | |
1441 if ((rmn_bits = FFMIN(rmn_bits, nbits)) > 0) | |
1442 put_bits(pb, rmn_bits, get_bits(gb, rmn_bits)); | |
1443 ff_copy_bits(pb, data + size - rmn_bytes, | |
1444 FFMIN(nbits - rmn_bits, rmn_bytes << 3)); | |
1445 } | |
1446 | |
1447 /** | |
1448 * Packet decoding: a packet is anything that the (ASF) demuxer contains, | |
1449 * and we expect that the demuxer / application provides it to us as such | |
1450 * (else you'll probably get garbage as output). Every packet has a size of | |
1451 * ctx->block_align bytes, starts with a packet header (see | |
1452 * #parse_packet_header()), and then a series of superframes. Superframe | |
1453 * boundaries may exceed packets, i.e. superframes can split data over | |
1454 * multiple (two) packets. | |
1455 * | |
1456 * For more information about frames, see #synth_superframe(). | |
1457 */ | |
1458 static int wmavoice_decode_packet(AVCodecContext *ctx, void *data, | |
1459 int *data_size, AVPacket *avpkt) | |
1460 { | |
1461 WMAVoiceContext *s = ctx->priv_data; | |
1462 GetBitContext *gb = &s->gb; | |
1463 int size, res, pos; | |
1464 | |
1465 if (*data_size < 480 * sizeof(float)) { | |
1466 av_log(ctx, AV_LOG_ERROR, | |
1467 "Output buffer too small (%d given - %lu needed)\n", | |
1468 *data_size, 480 * sizeof(float)); | |
1469 return -1; | |
1470 } | |
1471 *data_size = 0; | |
1472 | |
1473 /* Packets are sometimes a multiple of ctx->block_align, with a packet | |
1474 * header at each ctx->block_align bytes. However, FFmpeg's ASF demuxer | |
1475 * feeds us ASF packets, which may concatenate multiple "codec" packets | |
1476 * in a single "muxer" packet, so we artificially emulate that by | |
1477 * capping the packet size at ctx->block_align. */ | |
1478 for (size = avpkt->size; size > ctx->block_align; size -= ctx->block_align); | |
1479 if (!size) | |
1480 return 0; | |
1481 init_get_bits(&s->gb, avpkt->data, size << 3); | |
1482 | |
1483 /* size == ctx->block_align is used to indicate whether we are dealing with | |
1484 * a new packet or a packet of which we already read the packet header | |
1485 * previously. */ | |
1486 if (size == ctx->block_align) { // new packet header | |
1487 if ((res = parse_packet_header(s)) < 0) | |
1488 return res; | |
1489 | |
1490 /* If the packet header specifies a s->spillover_nbits, then we want | |
1491 * to push out all data of the previous packet (+ spillover) before | |
1492 * continuing to parse new superframes in the current packet. */ | |
1493 if (s->spillover_nbits > 0) { | |
1494 if (s->sframe_cache_size > 0) { | |
1495 int cnt = get_bits_count(gb); | |
1496 copy_bits(&s->pb, avpkt->data, size, gb, s->spillover_nbits); | |
1497 flush_put_bits(&s->pb); | |
1498 s->sframe_cache_size += s->spillover_nbits; | |
1499 if ((res = synth_superframe(ctx, data, data_size)) == 0 && | |
1500 *data_size > 0) { | |
1501 cnt += s->spillover_nbits; | |
1502 s->skip_bits_next = cnt & 7; | |
1503 return cnt >> 3; | |
1504 } else | |
1505 skip_bits_long (gb, s->spillover_nbits - cnt + | |
1506 get_bits_count(gb)); // resync | |
1507 } else | |
1508 skip_bits_long(gb, s->spillover_nbits); // resync | |
1509 } | |
1510 } else if (s->skip_bits_next) | |
1511 skip_bits(gb, s->skip_bits_next); | |
1512 | |
1513 /* Try parsing superframes in current packet */ | |
1514 s->sframe_cache_size = 0; | |
1515 s->skip_bits_next = 0; | |
1516 pos = get_bits_left(gb); | |
1517 if ((res = synth_superframe(ctx, data, data_size)) < 0) { | |
1518 return res; | |
1519 } else if (*data_size > 0) { | |
1520 int cnt = get_bits_count(gb); | |
1521 s->skip_bits_next = cnt & 7; | |
1522 return cnt >> 3; | |
1523 } else if ((s->sframe_cache_size = pos) > 0) { | |
1524 /* rewind bit reader to start of last (incomplete) superframe... */ | |
1525 init_get_bits(gb, avpkt->data, size << 3); | |
1526 skip_bits_long(gb, (size << 3) - pos); | |
1527 assert(get_bits_left(gb) == pos); | |
1528 | |
1529 /* ...and cache it for spillover in next packet */ | |
1530 init_put_bits(&s->pb, s->sframe_cache, SFRAME_CACHE_MAXSIZE); | |
1531 copy_bits(&s->pb, avpkt->data, size, gb, s->sframe_cache_size); | |
1532 // FIXME bad - just copy bytes as whole and add use the | |
1533 // skip_bits_next field | |
1534 } | |
1535 | |
1536 return size; | |
1537 } | |
1538 | |
1539 static av_cold void wmavoice_flush(AVCodecContext *ctx) | |
1540 { | |
1541 WMAVoiceContext *s = ctx->priv_data; | |
1542 int n; | |
1543 | |
1544 s->sframe_cache_size = 0; | |
1545 s->skip_bits_next = 0; | |
1546 for (n = 0; n < s->lsps; n++) | |
1547 s->prev_lsps[n] = M_PI * (n + 1.0) / (s->lsps + 1.0); | |
1548 memset(s->excitation_history, 0, | |
1549 sizeof(*s->excitation_history) * MAX_SIGNAL_HISTORY); | |
1550 memset(s->synth_history, 0, | |
1551 sizeof(*s->synth_history) * MAX_LSPS); | |
1552 memset(s->gain_pred_err, 0, | |
1553 sizeof(s->gain_pred_err)); | |
1554 } | |
1555 | |
1556 AVCodec wmavoice_decoder = { | |
1557 "wmavoice", | |
11560
8a4984c5cacc
Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents:
11556
diff
changeset
|
1558 AVMEDIA_TYPE_AUDIO, |
11123 | 1559 CODEC_ID_WMAVOICE, |
1560 sizeof(WMAVoiceContext), | |
1561 wmavoice_decode_init, | |
1562 NULL, | |
1563 NULL, | |
1564 wmavoice_decode_packet, | |
1565 CODEC_CAP_SUBFRAMES, | |
1566 .flush = wmavoice_flush, | |
1567 .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio Voice"), | |
1568 }; |