Mercurial > libavcodec.hg
annotate amrnbdec.c @ 12530:63edd10ad4bc libavcodec tip
Try to fix crashes introduced by r25218
r25218 made assumptions about the existence of past reference frames that
weren't necessarily true.
author | darkshikari |
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date | Tue, 28 Sep 2010 09:06:22 +0000 |
parents | b42e02e9bf2b |
children |
rev | line source |
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11235 | 1 /* |
2 * AMR narrowband decoder | |
3 * Copyright (c) 2006-2007 Robert Swain | |
4 * Copyright (c) 2009 Colin McQuillan | |
5 * | |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 | |
24 /** | |
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25 * @file |
11235 | 26 * AMR narrowband decoder |
27 * | |
28 * This decoder uses floats for simplicity and so is not bit-exact. One | |
29 * difference is that differences in phase can accumulate. The test sequences | |
30 * in 3GPP TS 26.074 can still be useful. | |
31 * | |
32 * - Comparing this file's output to the output of the ref decoder gives a | |
33 * PSNR of 30 to 80. Plotting the output samples shows a difference in | |
34 * phase in some areas. | |
35 * | |
36 * - Comparing both decoders against their input, this decoder gives a similar | |
37 * PSNR. If the test sequence homing frames are removed (this decoder does | |
38 * not detect them), the PSNR is at least as good as the reference on 140 | |
39 * out of 169 tests. | |
40 */ | |
41 | |
42 | |
43 #include <string.h> | |
44 #include <math.h> | |
45 | |
46 #include "avcodec.h" | |
47 #include "get_bits.h" | |
48 #include "libavutil/common.h" | |
49 #include "celp_math.h" | |
50 #include "celp_filters.h" | |
51 #include "acelp_filters.h" | |
52 #include "acelp_vectors.h" | |
53 #include "acelp_pitch_delay.h" | |
54 #include "lsp.h" | |
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55 #include "amr.h" |
11235 | 56 |
57 #include "amrnbdata.h" | |
58 | |
59 #define AMR_BLOCK_SIZE 160 ///< samples per frame | |
60 #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow | |
61 | |
62 /** | |
63 * Scale from constructed speech to [-1,1] | |
64 * | |
65 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but | |
66 * upscales by two (section 6.2.2). | |
67 * | |
68 * Fundamentally, this scale is determined by energy_mean through | |
69 * the fixed vector contribution to the excitation vector. | |
70 */ | |
71 #define AMR_SAMPLE_SCALE (2.0 / 32768.0) | |
72 | |
73 /** Prediction factor for 12.2kbit/s mode */ | |
74 #define PRED_FAC_MODE_12k2 0.65 | |
75 | |
76 #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz | |
77 #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter | |
78 #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode | |
79 | |
80 /** Initial energy in dB. Also used for bad frames (unimplemented). */ | |
81 #define MIN_ENERGY -14.0 | |
82 | |
83 /** Maximum sharpening factor | |
84 * | |
85 * The specification says 0.8, which should be 13107, but the reference C code | |
86 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.) | |
87 */ | |
88 #define SHARP_MAX 0.79449462890625 | |
89 | |
90 /** Number of impulse response coefficients used for tilt factor */ | |
91 #define AMR_TILT_RESPONSE 22 | |
92 /** Tilt factor = 1st reflection coefficient * gamma_t */ | |
93 #define AMR_TILT_GAMMA_T 0.8 | |
94 /** Adaptive gain control factor used in post-filter */ | |
95 #define AMR_AGC_ALPHA 0.9 | |
96 | |
97 typedef struct AMRContext { | |
98 AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc) | |
99 uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0 | |
100 enum Mode cur_frame_mode; | |
101 | |
102 int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe | |
103 double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame | |
104 double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame | |
105 | |
106 float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing | |
107 float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector | |
108 | |
109 float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes | |
110 | |
111 uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe | |
112 | |
113 float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history | |
114 float *excitation; ///< pointer to the current excitation vector in excitation_buf | |
115 | |
116 float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector | |
117 float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames) | |
118 | |
119 float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes | |
120 float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes | |
121 float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes | |
122 | |
123 float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX] | |
124 uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65 | |
125 uint8_t hang_count; ///< the number of subframes since a hangover period started | |
126 | |
127 float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset" | |
128 uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none | |
129 uint8_t ir_filter_onset; ///< flag for impulse response filter strength | |
130 | |
131 float postfilter_mem[10]; ///< previous intermediate values in the formant filter | |
132 float tilt_mem; ///< previous input to tilt compensation filter | |
133 float postfilter_agc; ///< previous factor used for adaptive gain control | |
134 float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter | |
135 | |
136 float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples | |
137 | |
138 } AMRContext; | |
139 | |
140 /** Double version of ff_weighted_vector_sumf() */ | |
141 static void weighted_vector_sumd(double *out, const double *in_a, | |
142 const double *in_b, double weight_coeff_a, | |
143 double weight_coeff_b, int length) | |
144 { | |
145 int i; | |
146 | |
147 for (i = 0; i < length; i++) | |
148 out[i] = weight_coeff_a * in_a[i] | |
149 + weight_coeff_b * in_b[i]; | |
150 } | |
151 | |
152 static av_cold int amrnb_decode_init(AVCodecContext *avctx) | |
153 { | |
154 AMRContext *p = avctx->priv_data; | |
155 int i; | |
156 | |
157 avctx->sample_fmt = SAMPLE_FMT_FLT; | |
158 | |
159 // p->excitation always points to the same position in p->excitation_buf | |
160 p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1]; | |
161 | |
162 for (i = 0; i < LP_FILTER_ORDER; i++) { | |
163 p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); | |
164 p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); | |
165 } | |
166 | |
167 for (i = 0; i < 4; i++) | |
168 p->prediction_error[i] = MIN_ENERGY; | |
169 | |
170 return 0; | |
171 } | |
172 | |
173 | |
174 /** | |
175 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters. | |
176 * | |
177 * The order of speech bits is specified by 3GPP TS 26.101. | |
178 * | |
179 * @param p the context | |
180 * @param buf pointer to the input buffer | |
181 * @param buf_size size of the input buffer | |
182 * | |
183 * @return the frame mode | |
184 */ | |
185 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf, | |
186 int buf_size) | |
187 { | |
188 GetBitContext gb; | |
189 enum Mode mode; | |
190 | |
191 init_get_bits(&gb, buf, buf_size * 8); | |
192 | |
193 // Decode the first octet. | |
194 skip_bits(&gb, 1); // padding bit | |
195 mode = get_bits(&gb, 4); // frame type | |
196 p->bad_frame_indicator = !get_bits1(&gb); // quality bit | |
197 skip_bits(&gb, 2); // two padding bits | |
198 | |
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199 if (mode < MODE_DTX) |
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200 ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1, |
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201 amr_unpacking_bitmaps_per_mode[mode]); |
11235 | 202 |
203 return mode; | |
204 } | |
205 | |
206 | |
207 /// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions | |
208 /// @{ | |
209 | |
210 /** | |
211 * Interpolate the LSF vector (used for fixed gain smoothing). | |
212 * The interpolation is done over all four subframes even in MODE_12k2. | |
213 * | |
214 * @param[in,out] lsf_q LSFs in [0,1] for each subframe | |
215 * @param[in] lsf_new New LSFs in [0,1] for subframe 4 | |
216 */ | |
217 static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) | |
218 { | |
219 int i; | |
220 | |
221 for (i = 0; i < 4; i++) | |
222 ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new, | |
223 0.25 * (3 - i), 0.25 * (i + 1), | |
224 LP_FILTER_ORDER); | |
225 } | |
226 | |
227 /** | |
228 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector. | |
229 * | |
230 * @param p the context | |
231 * @param lsp output LSP vector | |
232 * @param lsf_no_r LSF vector without the residual vector added | |
233 * @param lsf_quantizer pointers to LSF dictionary tables | |
234 * @param quantizer_offset offset in tables | |
235 * @param sign for the 3 dictionary table | |
236 * @param update store data for computing the next frame's LSFs | |
237 */ | |
238 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER], | |
239 const float lsf_no_r[LP_FILTER_ORDER], | |
240 const int16_t *lsf_quantizer[5], | |
241 const int quantizer_offset, | |
242 const int sign, const int update) | |
243 { | |
244 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector | |
245 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector | |
246 int i; | |
247 | |
248 for (i = 0; i < LP_FILTER_ORDER >> 1; i++) | |
249 memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset], | |
250 2 * sizeof(*lsf_r)); | |
251 | |
252 if (sign) { | |
253 lsf_r[4] *= -1; | |
254 lsf_r[5] *= -1; | |
255 } | |
256 | |
257 if (update) | |
258 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float)); | |
259 | |
260 for (i = 0; i < LP_FILTER_ORDER; i++) | |
261 lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0); | |
262 | |
263 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); | |
264 | |
265 if (update) | |
266 interpolate_lsf(p->lsf_q, lsf_q); | |
267 | |
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268 ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER); |
11235 | 269 } |
270 | |
271 /** | |
272 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors. | |
273 * | |
274 * @param p pointer to the AMRContext | |
275 */ | |
276 static void lsf2lsp_5(AMRContext *p) | |
277 { | |
278 const uint16_t *lsf_param = p->frame.lsf; | |
279 float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector | |
280 const int16_t *lsf_quantizer[5]; | |
281 int i; | |
282 | |
283 lsf_quantizer[0] = lsf_5_1[lsf_param[0]]; | |
284 lsf_quantizer[1] = lsf_5_2[lsf_param[1]]; | |
285 lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1]; | |
286 lsf_quantizer[3] = lsf_5_4[lsf_param[3]]; | |
287 lsf_quantizer[4] = lsf_5_5[lsf_param[4]]; | |
288 | |
289 for (i = 0; i < LP_FILTER_ORDER; i++) | |
290 lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i]; | |
291 | |
292 lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0); | |
293 lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1); | |
294 | |
295 // interpolate LSP vectors at subframes 1 and 3 | |
296 weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER); | |
297 weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER); | |
298 } | |
299 | |
300 /** | |
301 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector. | |
302 * | |
303 * @param p pointer to the AMRContext | |
304 */ | |
305 static void lsf2lsp_3(AMRContext *p) | |
306 { | |
307 const uint16_t *lsf_param = p->frame.lsf; | |
308 int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector | |
309 float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector | |
310 const int16_t *lsf_quantizer; | |
311 int i, j; | |
312 | |
313 lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]]; | |
314 memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r)); | |
315 | |
316 lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)]; | |
317 memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r)); | |
318 | |
319 lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]]; | |
320 memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r)); | |
321 | |
322 // calculate mean-removed LSF vector and add mean | |
323 for (i = 0; i < LP_FILTER_ORDER; i++) | |
324 lsf_q[i] = (lsf_r[i] + p->prev_lsf_r[i] * pred_fac[i]) * (LSF_R_FAC / 8000.0) + lsf_3_mean[i] * (1.0 / 8000.0); | |
325 | |
326 ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); | |
327 | |
328 // store data for computing the next frame's LSFs | |
329 interpolate_lsf(p->lsf_q, lsf_q); | |
330 memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); | |
331 | |
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332 ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER); |
11235 | 333 |
334 // interpolate LSP vectors at subframes 1, 2 and 3 | |
335 for (i = 1; i <= 3; i++) | |
336 for(j = 0; j < LP_FILTER_ORDER; j++) | |
337 p->lsp[i-1][j] = p->prev_lsp_sub4[j] + | |
338 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i; | |
339 } | |
340 | |
341 /// @} | |
342 | |
343 | |
344 /// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions | |
345 /// @{ | |
346 | |
347 /** | |
348 * Like ff_decode_pitch_lag(), but with 1/6 resolution | |
349 */ | |
350 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index, | |
351 const int prev_lag_int, const int subframe) | |
352 { | |
353 if (subframe == 0 || subframe == 2) { | |
354 if (pitch_index < 463) { | |
355 *lag_int = (pitch_index + 107) * 10923 >> 16; | |
356 *lag_frac = pitch_index - *lag_int * 6 + 105; | |
357 } else { | |
358 *lag_int = pitch_index - 368; | |
359 *lag_frac = 0; | |
360 } | |
361 } else { | |
362 *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1; | |
363 *lag_frac = pitch_index - *lag_int * 6 - 3; | |
364 *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2, | |
365 PITCH_DELAY_MAX - 9); | |
366 } | |
367 } | |
368 | |
369 static void decode_pitch_vector(AMRContext *p, | |
370 const AMRNBSubframe *amr_subframe, | |
371 const int subframe) | |
372 { | |
373 int pitch_lag_int, pitch_lag_frac; | |
374 enum Mode mode = p->cur_frame_mode; | |
375 | |
376 if (p->cur_frame_mode == MODE_12k2) { | |
377 decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac, | |
378 amr_subframe->p_lag, p->pitch_lag_int, | |
379 subframe); | |
380 } else | |
381 ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac, | |
382 amr_subframe->p_lag, | |
383 p->pitch_lag_int, subframe, | |
384 mode != MODE_4k75 && mode != MODE_5k15, | |
385 mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6)); | |
386 | |
387 p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t | |
388 | |
389 pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2); | |
390 | |
391 pitch_lag_int += pitch_lag_frac > 0; | |
392 | |
393 /* Calculate the pitch vector by interpolating the past excitation at the | |
394 pitch lag using a b60 hamming windowed sinc function. */ | |
395 ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int, | |
396 ff_b60_sinc, 6, | |
397 pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0), | |
398 10, AMR_SUBFRAME_SIZE); | |
399 | |
400 memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float)); | |
401 } | |
402 | |
403 /// @} | |
404 | |
405 | |
406 /// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions | |
407 /// @{ | |
408 | |
409 /** | |
410 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame. | |
411 */ | |
412 static void decode_10bit_pulse(int code, int pulse_position[8], | |
413 int i1, int i2, int i3) | |
414 { | |
415 // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of | |
416 // the 3 pulses and the upper 7 bits being coded in base 5 | |
417 const uint8_t *positions = base_five_table[code >> 3]; | |
418 pulse_position[i1] = (positions[2] << 1) + ( code & 1); | |
419 pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1); | |
420 pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1); | |
421 } | |
422 | |
423 /** | |
424 * Decode the algebraic codebook index to pulse positions and signs and | |
425 * construct the algebraic codebook vector for MODE_10k2. | |
426 * | |
427 * @param fixed_index positions of the eight pulses | |
428 * @param fixed_sparse pointer to the algebraic codebook vector | |
429 */ | |
430 static void decode_8_pulses_31bits(const int16_t *fixed_index, | |
431 AMRFixed *fixed_sparse) | |
432 { | |
433 int pulse_position[8]; | |
434 int i, temp; | |
435 | |
436 decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1); | |
437 decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5); | |
438 | |
439 // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of | |
440 // the 2 pulses and the upper 5 bits being coded in base 5 | |
441 temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5; | |
442 pulse_position[3] = temp % 5; | |
443 pulse_position[7] = temp / 5; | |
444 if (pulse_position[7] & 1) | |
445 pulse_position[3] = 4 - pulse_position[3]; | |
446 pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1); | |
447 pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1); | |
448 | |
449 fixed_sparse->n = 8; | |
450 for (i = 0; i < 4; i++) { | |
451 const int pos1 = (pulse_position[i] << 2) + i; | |
452 const int pos2 = (pulse_position[i + 4] << 2) + i; | |
453 const float sign = fixed_index[i] ? -1.0 : 1.0; | |
454 fixed_sparse->x[i ] = pos1; | |
455 fixed_sparse->x[i + 4] = pos2; | |
456 fixed_sparse->y[i ] = sign; | |
457 fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign; | |
458 } | |
459 } | |
460 | |
461 /** | |
462 * Decode the algebraic codebook index to pulse positions and signs, | |
463 * then construct the algebraic codebook vector. | |
464 * | |
465 * nb of pulses | bits encoding pulses | |
466 * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7 | |
467 * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9 | |
468 * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11 | |
469 * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13 | |
470 * | |
471 * @param fixed_sparse pointer to the algebraic codebook vector | |
472 * @param pulses algebraic codebook indexes | |
473 * @param mode mode of the current frame | |
474 * @param subframe current subframe number | |
475 */ | |
476 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses, | |
477 const enum Mode mode, const int subframe) | |
478 { | |
479 assert(MODE_4k75 <= mode && mode <= MODE_12k2); | |
480 | |
481 if (mode == MODE_12k2) { | |
482 ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3); | |
483 } else if (mode == MODE_10k2) { | |
484 decode_8_pulses_31bits(pulses, fixed_sparse); | |
485 } else { | |
486 int *pulse_position = fixed_sparse->x; | |
487 int i, pulse_subset; | |
488 const int fixed_index = pulses[0]; | |
489 | |
490 if (mode <= MODE_5k15) { | |
491 pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1); | |
492 pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset]; | |
493 pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1]; | |
494 fixed_sparse->n = 2; | |
495 } else if (mode == MODE_5k9) { | |
496 pulse_subset = ((fixed_index & 1) << 1) + 1; | |
497 pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset; | |
498 pulse_subset = (fixed_index >> 4) & 3; | |
499 pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0); | |
500 fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2; | |
501 } else if (mode == MODE_6k7) { | |
502 pulse_position[0] = (fixed_index & 7) * 5; | |
503 pulse_subset = (fixed_index >> 2) & 2; | |
504 pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1; | |
505 pulse_subset = (fixed_index >> 6) & 2; | |
506 pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2; | |
507 fixed_sparse->n = 3; | |
508 } else { // mode <= MODE_7k95 | |
509 pulse_position[0] = gray_decode[ fixed_index & 7]; | |
510 pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1; | |
511 pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2; | |
512 pulse_subset = (fixed_index >> 9) & 1; | |
513 pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3; | |
514 fixed_sparse->n = 4; | |
515 } | |
516 for (i = 0; i < fixed_sparse->n; i++) | |
517 fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0; | |
518 } | |
519 } | |
520 | |
521 /** | |
522 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2) | |
523 * | |
524 * @param p the context | |
525 * @param subframe unpacked amr subframe | |
526 * @param mode mode of the current frame | |
527 * @param fixed_sparse sparse respresentation of the fixed vector | |
528 */ | |
529 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode, | |
530 AMRFixed *fixed_sparse) | |
531 { | |
532 // The spec suggests the current pitch gain is always used, but in other | |
533 // modes the pitch and codebook gains are joinly quantized (sec 5.8.2) | |
534 // so the codebook gain cannot depend on the quantized pitch gain. | |
535 if (mode == MODE_12k2) | |
536 p->beta = FFMIN(p->pitch_gain[4], 1.0); | |
537 | |
538 fixed_sparse->pitch_lag = p->pitch_lag_int; | |
539 fixed_sparse->pitch_fac = p->beta; | |
540 | |
541 // Save pitch sharpening factor for the next subframe | |
542 // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from | |
543 // the fact that the gains for two subframes are jointly quantized. | |
544 if (mode != MODE_4k75 || subframe & 1) | |
545 p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX); | |
546 } | |
547 /// @} | |
548 | |
549 | |
550 /// @defgroup amr_gain_decoding AMR gain decoding functions | |
551 /// @{ | |
552 | |
553 /** | |
554 * fixed gain smoothing | |
555 * Note that where the spec specifies the "spectrum in the q domain" | |
556 * in section 6.1.4, in fact frequencies should be used. | |
557 * | |
558 * @param p the context | |
559 * @param lsf LSFs for the current subframe, in the range [0,1] | |
560 * @param lsf_avg averaged LSFs | |
561 * @param mode mode of the current frame | |
562 * | |
563 * @return fixed gain smoothed | |
564 */ | |
565 static float fixed_gain_smooth(AMRContext *p , const float *lsf, | |
566 const float *lsf_avg, const enum Mode mode) | |
567 { | |
568 float diff = 0.0; | |
569 int i; | |
570 | |
571 for (i = 0; i < LP_FILTER_ORDER; i++) | |
572 diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i]; | |
573 | |
574 // If diff is large for ten subframes, disable smoothing for a 40-subframe | |
575 // hangover period. | |
576 p->diff_count++; | |
577 if (diff <= 0.65) | |
578 p->diff_count = 0; | |
579 | |
580 if (p->diff_count > 10) { | |
581 p->hang_count = 0; | |
582 p->diff_count--; // don't let diff_count overflow | |
583 } | |
584 | |
585 if (p->hang_count < 40) { | |
586 p->hang_count++; | |
587 } else if (mode < MODE_7k4 || mode == MODE_10k2) { | |
588 const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0); | |
589 const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + | |
590 p->fixed_gain[2] + p->fixed_gain[3] + | |
591 p->fixed_gain[4]) * 0.2; | |
592 return smoothing_factor * p->fixed_gain[4] + | |
593 (1.0 - smoothing_factor) * fixed_gain_mean; | |
594 } | |
595 return p->fixed_gain[4]; | |
596 } | |
597 | |
598 /** | |
599 * Decode pitch gain and fixed gain factor (part of section 6.1.3). | |
600 * | |
601 * @param p the context | |
602 * @param amr_subframe unpacked amr subframe | |
603 * @param mode mode of the current frame | |
604 * @param subframe current subframe number | |
605 * @param fixed_gain_factor decoded gain correction factor | |
606 */ | |
607 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, | |
608 const enum Mode mode, const int subframe, | |
609 float *fixed_gain_factor) | |
610 { | |
611 if (mode == MODE_12k2 || mode == MODE_7k95) { | |
612 p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ] | |
613 * (1.0 / 16384.0); | |
614 *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain] | |
615 * (1.0 / 2048.0); | |
616 } else { | |
617 const uint16_t *gains; | |
618 | |
619 if (mode >= MODE_6k7) { | |
620 gains = gains_high[amr_subframe->p_gain]; | |
621 } else if (mode >= MODE_5k15) { | |
622 gains = gains_low [amr_subframe->p_gain]; | |
623 } else { | |
624 // gain index is only coded in subframes 0,2 for MODE_4k75 | |
625 gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)]; | |
626 } | |
627 | |
628 p->pitch_gain[4] = gains[0] * (1.0 / 16384.0); | |
629 *fixed_gain_factor = gains[1] * (1.0 / 4096.0); | |
630 } | |
631 } | |
632 | |
633 /// @} | |
634 | |
635 | |
636 /// @defgroup amr_pre_processing AMR pre-processing functions | |
637 /// @{ | |
638 | |
639 /** | |
640 * Circularly convolve a sparse fixed vector with a phase dispersion impulse | |
641 * response filter (D.6.2 of G.729 and 6.1.5 of AMR). | |
642 * | |
643 * @param out vector with filter applied | |
644 * @param in source vector | |
645 * @param filter phase filter coefficients | |
646 * | |
647 * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] } | |
648 */ | |
649 static void apply_ir_filter(float *out, const AMRFixed *in, | |
650 const float *filter) | |
651 { | |
652 float filter1[AMR_SUBFRAME_SIZE], //!< filters at pitch lag*1 and *2 | |
653 filter2[AMR_SUBFRAME_SIZE]; | |
654 int lag = in->pitch_lag; | |
655 float fac = in->pitch_fac; | |
656 int i; | |
657 | |
658 if (lag < AMR_SUBFRAME_SIZE) { | |
659 ff_celp_circ_addf(filter1, filter, filter, lag, fac, | |
660 AMR_SUBFRAME_SIZE); | |
661 | |
662 if (lag < AMR_SUBFRAME_SIZE >> 1) | |
663 ff_celp_circ_addf(filter2, filter, filter1, lag, fac, | |
664 AMR_SUBFRAME_SIZE); | |
665 } | |
666 | |
667 memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE); | |
668 for (i = 0; i < in->n; i++) { | |
669 int x = in->x[i]; | |
670 float y = in->y[i]; | |
671 const float *filterp; | |
672 | |
673 if (x >= AMR_SUBFRAME_SIZE - lag) { | |
674 filterp = filter; | |
675 } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) { | |
676 filterp = filter1; | |
677 } else | |
678 filterp = filter2; | |
679 | |
680 ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE); | |
681 } | |
682 } | |
683 | |
684 /** | |
685 * Reduce fixed vector sparseness by smoothing with one of three IR filters. | |
686 * Also know as "adaptive phase dispersion". | |
687 * | |
688 * This implements 3GPP TS 26.090 section 6.1(5). | |
689 * | |
690 * @param p the context | |
691 * @param fixed_sparse algebraic codebook vector | |
692 * @param fixed_vector unfiltered fixed vector | |
693 * @param fixed_gain smoothed gain | |
694 * @param out space for modified vector if necessary | |
695 */ | |
696 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse, | |
697 const float *fixed_vector, | |
698 float fixed_gain, float *out) | |
699 { | |
700 int ir_filter_nr; | |
701 | |
702 if (p->pitch_gain[4] < 0.6) { | |
703 ir_filter_nr = 0; // strong filtering | |
704 } else if (p->pitch_gain[4] < 0.9) { | |
705 ir_filter_nr = 1; // medium filtering | |
706 } else | |
707 ir_filter_nr = 2; // no filtering | |
708 | |
709 // detect 'onset' | |
710 if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) { | |
711 p->ir_filter_onset = 2; | |
712 } else if (p->ir_filter_onset) | |
713 p->ir_filter_onset--; | |
714 | |
715 if (!p->ir_filter_onset) { | |
716 int i, count = 0; | |
717 | |
718 for (i = 0; i < 5; i++) | |
719 if (p->pitch_gain[i] < 0.6) | |
720 count++; | |
721 if (count > 2) | |
722 ir_filter_nr = 0; | |
723 | |
724 if (ir_filter_nr > p->prev_ir_filter_nr + 1) | |
725 ir_filter_nr--; | |
726 } else if (ir_filter_nr < 2) | |
727 ir_filter_nr++; | |
728 | |
729 // Disable filtering for very low level of fixed_gain. | |
730 // Note this step is not specified in the technical description but is in | |
731 // the reference source in the function Ph_disp. | |
732 if (fixed_gain < 5.0) | |
733 ir_filter_nr = 2; | |
734 | |
735 if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2 | |
736 && ir_filter_nr < 2) { | |
737 apply_ir_filter(out, fixed_sparse, | |
738 (p->cur_frame_mode == MODE_7k95 ? | |
739 ir_filters_lookup_MODE_7k95 : | |
740 ir_filters_lookup)[ir_filter_nr]); | |
741 fixed_vector = out; | |
742 } | |
743 | |
744 // update ir filter strength history | |
745 p->prev_ir_filter_nr = ir_filter_nr; | |
746 p->prev_sparse_fixed_gain = fixed_gain; | |
747 | |
748 return fixed_vector; | |
749 } | |
750 | |
751 /// @} | |
752 | |
753 | |
754 /// @defgroup amr_synthesis AMR synthesis functions | |
755 /// @{ | |
756 | |
757 /** | |
758 * Conduct 10th order linear predictive coding synthesis. | |
759 * | |
760 * @param p pointer to the AMRContext | |
761 * @param lpc pointer to the LPC coefficients | |
762 * @param fixed_gain fixed codebook gain for synthesis | |
763 * @param fixed_vector algebraic codebook vector | |
764 * @param samples pointer to the output speech samples | |
765 * @param overflow 16-bit overflow flag | |
766 */ | |
767 static int synthesis(AMRContext *p, float *lpc, | |
768 float fixed_gain, const float *fixed_vector, | |
769 float *samples, uint8_t overflow) | |
770 { | |
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771 int i; |
11235 | 772 float excitation[AMR_SUBFRAME_SIZE]; |
773 | |
774 // if an overflow has been detected, the pitch vector is scaled down by a | |
775 // factor of 4 | |
776 if (overflow) | |
777 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
778 p->pitch_vector[i] *= 0.25; | |
779 | |
780 ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector, | |
781 p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE); | |
782 | |
783 // emphasize pitch vector contribution | |
784 if (p->pitch_gain[4] > 0.5 && !overflow) { | |
785 float energy = ff_dot_productf(excitation, excitation, | |
786 AMR_SUBFRAME_SIZE); | |
787 float pitch_factor = | |
788 p->pitch_gain[4] * | |
789 (p->cur_frame_mode == MODE_12k2 ? | |
790 0.25 * FFMIN(p->pitch_gain[4], 1.0) : | |
791 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX)); | |
792 | |
793 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
794 excitation[i] += pitch_factor * p->pitch_vector[i]; | |
795 | |
796 ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy, | |
797 AMR_SUBFRAME_SIZE); | |
798 } | |
799 | |
800 ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE, | |
801 LP_FILTER_ORDER); | |
802 | |
803 // detect overflow | |
804 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
805 if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) { | |
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806 return 1; |
11235 | 807 } |
808 | |
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809 return 0; |
11235 | 810 } |
811 | |
812 /// @} | |
813 | |
814 | |
815 /// @defgroup amr_update AMR update functions | |
816 /// @{ | |
817 | |
818 /** | |
819 * Update buffers and history at the end of decoding a subframe. | |
820 * | |
821 * @param p pointer to the AMRContext | |
822 */ | |
823 static void update_state(AMRContext *p) | |
824 { | |
825 memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0])); | |
826 | |
827 memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE], | |
828 (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float)); | |
829 | |
830 memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float)); | |
831 memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float)); | |
832 | |
833 memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE], | |
834 LP_FILTER_ORDER * sizeof(float)); | |
835 } | |
836 | |
837 /// @} | |
838 | |
839 | |
840 /// @defgroup amr_postproc AMR Post processing functions | |
841 /// @{ | |
842 | |
843 /** | |
844 * Get the tilt factor of a formant filter from its transfer function | |
845 * | |
846 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator | |
847 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator | |
848 */ | |
849 static float tilt_factor(float *lpc_n, float *lpc_d) | |
850 { | |
851 float rh0, rh1; // autocorrelation at lag 0 and 1 | |
852 | |
853 // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf | |
854 float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 }; | |
855 float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response | |
856 | |
857 hf[0] = 1.0; | |
858 memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER); | |
859 ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE, | |
860 LP_FILTER_ORDER); | |
861 | |
862 rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE); | |
863 rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1); | |
864 | |
865 // The spec only specifies this check for 12.2 and 10.2 kbit/s | |
866 // modes. But in the ref source the tilt is always non-negative. | |
867 return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0; | |
868 } | |
869 | |
870 /** | |
871 * Perform adaptive post-filtering to enhance the quality of the speech. | |
872 * See section 6.2.1. | |
873 * | |
874 * @param p pointer to the AMRContext | |
875 * @param lpc interpolated LP coefficients for this subframe | |
876 * @param buf_out output of the filter | |
877 */ | |
878 static void postfilter(AMRContext *p, float *lpc, float *buf_out) | |
879 { | |
880 int i; | |
881 float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input | |
882 | |
883 float speech_gain = ff_dot_productf(samples, samples, | |
884 AMR_SUBFRAME_SIZE); | |
885 | |
886 float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter | |
887 const float *gamma_n, *gamma_d; // Formant filter factor table | |
888 float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients | |
889 | |
890 if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) { | |
891 gamma_n = ff_pow_0_7; | |
892 gamma_d = ff_pow_0_75; | |
893 } else { | |
894 gamma_n = ff_pow_0_55; | |
895 gamma_d = ff_pow_0_7; | |
896 } | |
897 | |
898 for (i = 0; i < LP_FILTER_ORDER; i++) { | |
899 lpc_n[i] = lpc[i] * gamma_n[i]; | |
900 lpc_d[i] = lpc[i] * gamma_d[i]; | |
901 } | |
902 | |
903 memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER); | |
904 ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, | |
905 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); | |
906 memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE, | |
907 sizeof(float) * LP_FILTER_ORDER); | |
908 | |
909 ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n, | |
910 pole_out + LP_FILTER_ORDER, | |
911 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); | |
912 | |
913 ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out, | |
914 AMR_SUBFRAME_SIZE); | |
915 | |
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916 ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE, |
11462 | 917 AMR_AGC_ALPHA, &p->postfilter_agc); |
11235 | 918 } |
919 | |
920 /// @} | |
921 | |
922 static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, | |
923 AVPacket *avpkt) | |
924 { | |
925 | |
926 AMRContext *p = avctx->priv_data; // pointer to private data | |
927 const uint8_t *buf = avpkt->data; | |
928 int buf_size = avpkt->size; | |
929 float *buf_out = data; // pointer to the output data buffer | |
930 int i, subframe; | |
931 float fixed_gain_factor; | |
932 AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing | |
933 float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing | |
934 float synth_fixed_gain; // the fixed gain that synthesis should use | |
935 const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use | |
936 | |
937 p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); | |
938 if (p->cur_frame_mode == MODE_DTX) { | |
939 av_log_missing_feature(avctx, "dtx mode", 1); | |
940 return -1; | |
941 } | |
942 | |
943 if (p->cur_frame_mode == MODE_12k2) { | |
944 lsf2lsp_5(p); | |
945 } else | |
946 lsf2lsp_3(p); | |
947 | |
948 for (i = 0; i < 4; i++) | |
949 ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5); | |
950 | |
951 for (subframe = 0; subframe < 4; subframe++) { | |
952 const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe]; | |
953 | |
954 decode_pitch_vector(p, amr_subframe, subframe); | |
955 | |
956 decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses, | |
957 p->cur_frame_mode, subframe); | |
958 | |
959 // The fixed gain (section 6.1.3) depends on the fixed vector | |
960 // (section 6.1.2), but the fixed vector calculation uses | |
961 // pitch sharpening based on the on the pitch gain (section 6.1.3). | |
962 // So the correct order is: pitch gain, pitch sharpening, fixed gain. | |
963 decode_gains(p, amr_subframe, p->cur_frame_mode, subframe, | |
964 &fixed_gain_factor); | |
965 | |
966 pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse); | |
967 | |
968 ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0, | |
969 AMR_SUBFRAME_SIZE); | |
970 | |
971 p->fixed_gain[4] = | |
972 ff_amr_set_fixed_gain(fixed_gain_factor, | |
973 ff_dot_productf(p->fixed_vector, p->fixed_vector, | |
974 AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE, | |
975 p->prediction_error, | |
976 energy_mean[p->cur_frame_mode], energy_pred_fac); | |
977 | |
978 // The excitation feedback is calculated without any processing such | |
979 // as fixed gain smoothing. This isn't mentioned in the specification. | |
980 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
981 p->excitation[i] *= p->pitch_gain[4]; | |
982 ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4], | |
983 AMR_SUBFRAME_SIZE); | |
984 | |
985 // In the ref decoder, excitation is stored with no fractional bits. | |
986 // This step prevents buzz in silent periods. The ref encoder can | |
987 // emit long sequences with pitch factor greater than one. This | |
988 // creates unwanted feedback if the excitation vector is nonzero. | |
989 // (e.g. test sequence T19_795.COD in 3GPP TS 26.074) | |
990 for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
991 p->excitation[i] = truncf(p->excitation[i]); | |
992 | |
993 // Smooth fixed gain. | |
994 // The specification is ambiguous, but in the reference source, the | |
995 // smoothed value is NOT fed back into later fixed gain smoothing. | |
996 synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe], | |
997 p->lsf_avg, p->cur_frame_mode); | |
998 | |
999 synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector, | |
1000 synth_fixed_gain, spare_vector); | |
1001 | |
1002 if (synthesis(p, p->lpc[subframe], synth_fixed_gain, | |
1003 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0)) | |
1004 // overflow detected -> rerun synthesis scaling pitch vector down | |
1005 // by a factor of 4, skipping pitch vector contribution emphasis | |
1006 // and adaptive gain control | |
1007 synthesis(p, p->lpc[subframe], synth_fixed_gain, | |
1008 synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1); | |
1009 | |
1010 postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); | |
1011 | |
1012 // update buffers and history | |
1013 ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); | |
1014 update_state(p); | |
1015 } | |
1016 | |
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1017 ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros, |
11676
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amrnbdec: Apply AMR_SAMPLE_SCALE when finishing the decoder output
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1018 highpass_poles, |
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amrnbdec: Apply AMR_SAMPLE_SCALE when finishing the decoder output
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1019 highpass_gain * AMR_SAMPLE_SCALE, |
11235 | 1020 p->high_pass_mem, AMR_BLOCK_SIZE); |
1021 | |
1022 /* Update averaged lsf vector (used for fixed gain smoothing). | |
1023 * | |
1024 * Note that lsf_avg should not incorporate the current frame's LSFs | |
1025 * for fixed_gain_smooth. | |
1026 * The specification has an incorrect formula: the reference decoder uses | |
1027 * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ | |
1028 ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], | |
1029 0.84, 0.16, LP_FILTER_ORDER); | |
1030 | |
1031 /* report how many samples we got */ | |
1032 *data_size = AMR_BLOCK_SIZE * sizeof(float); | |
1033 | |
1034 /* return the amount of bytes consumed if everything was OK */ | |
1035 return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC | |
1036 } | |
1037 | |
1038 | |
1039 AVCodec amrnb_decoder = { | |
1040 .name = "amrnb", | |
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1041 .type = AVMEDIA_TYPE_AUDIO, |
11235 | 1042 .id = CODEC_ID_AMR_NB, |
1043 .priv_data_size = sizeof(AMRContext), | |
1044 .init = amrnb_decode_init, | |
1045 .decode = amrnb_decode_frame, | |
1046 .long_name = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"), | |
1047 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE}, | |
1048 }; |