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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 | /** | |
25 | * @file | |
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 "libavutil/channel_layout.h" | |
47 | #include "libavutil/float_dsp.h" | |
48 | #include "avcodec.h" | |
49 | #include "libavutil/common.h" | |
50 | #include "libavutil/avassert.h" | |
51 | #include "celp_math.h" | |
52 | #include "celp_filters.h" | |
53 | #include "acelp_filters.h" | |
54 | #include "acelp_vectors.h" | |
55 | #include "acelp_pitch_delay.h" | |
56 | #include "lsp.h" | |
57 | #include "amr.h" | |
58 | #include "internal.h" | |
59 | ||
60 | #include "amrnbdata.h" | |
61 | ||
62 | #define AMR_BLOCK_SIZE 160 ///< samples per frame | |
63 | #define AMR_SAMPLE_BOUND 32768.0 ///< threshold for synthesis overflow | |
64 | ||
65 | /** | |
66 | * Scale from constructed speech to [-1,1] | |
67 | * | |
68 | * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but | |
69 | * upscales by two (section 6.2.2). | |
70 | * | |
71 | * Fundamentally, this scale is determined by energy_mean through | |
72 | * the fixed vector contribution to the excitation vector. | |
73 | */ | |
74 | #define AMR_SAMPLE_SCALE (2.0 / 32768.0) | |
75 | ||
76 | /** Prediction factor for 12.2kbit/s mode */ | |
77 | #define PRED_FAC_MODE_12k2 0.65 | |
78 | ||
79 | #define LSF_R_FAC (8000.0 / 32768.0) ///< LSF residual tables to Hertz | |
80 | #define MIN_LSF_SPACING (50.0488 / 8000.0) ///< Ensures stability of LPC filter | |
81 | #define PITCH_LAG_MIN_MODE_12k2 18 ///< Lower bound on decoded lag search in 12.2kbit/s mode | |
82 | ||
83 | /** Initial energy in dB. Also used for bad frames (unimplemented). */ | |
84 | #define MIN_ENERGY -14.0 | |
85 | ||
86 | /** Maximum sharpening factor | |
87 | * | |
88 | * The specification says 0.8, which should be 13107, but the reference C code | |
89 | * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.) | |
90 | */ | |
91 | #define SHARP_MAX 0.79449462890625 | |
92 | ||
93 | /** Number of impulse response coefficients used for tilt factor */ | |
94 | #define AMR_TILT_RESPONSE 22 | |
95 | /** Tilt factor = 1st reflection coefficient * gamma_t */ | |
96 | #define AMR_TILT_GAMMA_T 0.8 | |
97 | /** Adaptive gain control factor used in post-filter */ | |
98 | #define AMR_AGC_ALPHA 0.9 | |
99 | ||
100 | typedef struct AMRContext { | |
101 | AMRNBFrame frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc) | |
102 | uint8_t bad_frame_indicator; ///< bad frame ? 1 : 0 | |
103 | enum Mode cur_frame_mode; | |
104 | ||
105 | int16_t prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe | |
106 | double lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame | |
107 | double prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame | |
108 | ||
109 | float lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing | |
110 | float lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector | |
111 | ||
112 | float lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes | |
113 | ||
114 | uint8_t pitch_lag_int; ///< integer part of pitch lag from current subframe | |
115 | ||
116 | float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history | |
117 | float *excitation; ///< pointer to the current excitation vector in excitation_buf | |
118 | ||
119 | float pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector | |
120 | float fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames) | |
121 | ||
122 | float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes | |
123 | float pitch_gain[5]; ///< quantified pitch gains for the current and previous four subframes | |
124 | float fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes | |
125 | ||
126 | float beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX] | |
127 | uint8_t diff_count; ///< the number of subframes for which diff has been above 0.65 | |
128 | uint8_t hang_count; ///< the number of subframes since a hangover period started | |
129 | ||
130 | float prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing to determine "onset" | |
131 | uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0 - strong, 1 - medium, 2 - none | |
132 | uint8_t ir_filter_onset; ///< flag for impulse response filter strength | |
133 | ||
134 | float postfilter_mem[10]; ///< previous intermediate values in the formant filter | |
135 | float tilt_mem; ///< previous input to tilt compensation filter | |
136 | float postfilter_agc; ///< previous factor used for adaptive gain control | |
137 | float high_pass_mem[2]; ///< previous intermediate values in the high-pass filter | |
138 | ||
139 | float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples | |
140 | ||
141 | ACELPFContext acelpf_ctx; ///< context for filters for ACELP-based codecs | |
142 | ACELPVContext acelpv_ctx; ///< context for vector operations for ACELP-based codecs | |
143 | CELPFContext celpf_ctx; ///< context for filters for CELP-based codecs | |
144 | CELPMContext celpm_ctx; ///< context for fixed point math operations | |
145 | ||
146 | } AMRContext; | |
147 | ||
148 | /** Double version of ff_weighted_vector_sumf() */ | |
149 | static void weighted_vector_sumd(double *out, const double *in_a, | |
150 | const double *in_b, double weight_coeff_a, | |
151 | double weight_coeff_b, int length) | |
152 | { | |
153 | int i; | |
154 | ||
155 | for (i = 0; i < length; i++) | |
156 | out[i] = weight_coeff_a * in_a[i] | |
157 | + weight_coeff_b * in_b[i]; | |
158 | } | |
159 | ||
160 | static av_cold int amrnb_decode_init(AVCodecContext *avctx) | |
161 | { | |
162 | AMRContext *p = avctx->priv_data; | |
163 | int i; | |
164 | ||
165 | if (avctx->channels > 1) { | |
166 | avpriv_report_missing_feature(avctx, "multi-channel AMR"); | |
167 | return AVERROR_PATCHWELCOME; | |
168 | } | |
169 | ||
170 | avctx->channels = 1; | |
171 | avctx->channel_layout = AV_CH_LAYOUT_MONO; | |
172 | if (!avctx->sample_rate) | |
173 | avctx->sample_rate = 8000; | |
174 | avctx->sample_fmt = AV_SAMPLE_FMT_FLT; | |
175 | ||
176 | // p->excitation always points to the same position in p->excitation_buf | |
177 | p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1]; | |
178 | ||
179 | for (i = 0; i < LP_FILTER_ORDER; i++) { | |
180 | p->prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); | |
181 | p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); | |
182 | } | |
183 | ||
184 | for (i = 0; i < 4; i++) | |
185 | p->prediction_error[i] = MIN_ENERGY; | |
186 | ||
187 | ff_acelp_filter_init(&p->acelpf_ctx); | |
188 | ff_acelp_vectors_init(&p->acelpv_ctx); | |
189 | ff_celp_filter_init(&p->celpf_ctx); | |
190 | ff_celp_math_init(&p->celpm_ctx); | |
191 | ||
192 | return 0; | |
193 | } | |
194 | ||
195 | ||
196 | /** | |
197 | * Unpack an RFC4867 speech frame into the AMR frame mode and parameters. | |
198 | * | |
199 | * The order of speech bits is specified by 3GPP TS 26.101. | |
200 | * | |
201 | * @param p the context | |
202 | * @param buf pointer to the input buffer | |
203 | * @param buf_size size of the input buffer | |
204 | * | |
205 | * @return the frame mode | |
206 | */ | |
207 | static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf, | |
208 | int buf_size) | |
209 | { | |
210 | enum Mode mode; | |
211 | ||
212 | // Decode the first octet. | |
213 | mode = buf[0] >> 3 & 0x0F; // frame type | |
214 | p->bad_frame_indicator = (buf[0] & 0x4) != 0x4; // quality bit | |
215 | ||
216 | if (mode >= N_MODES || buf_size < frame_sizes_nb[mode] + 1) { | |
217 | return NO_DATA; | |
218 | } | |
219 | ||
220 | if (mode < MODE_DTX) | |
221 | ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1, | |
222 | amr_unpacking_bitmaps_per_mode[mode]); | |
223 | ||
224 | return mode; | |
225 | } | |
226 | ||
227 | ||
228 | /// @name AMR pitch LPC coefficient decoding functions | |
229 | /// @{ | |
230 | ||
231 | /** | |
232 | * Interpolate the LSF vector (used for fixed gain smoothing). | |
233 | * The interpolation is done over all four subframes even in MODE_12k2. | |
234 | * | |
235 | * @param[in] ctx The Context | |
236 | * @param[in,out] lsf_q LSFs in [0,1] for each subframe | |
237 | * @param[in] lsf_new New LSFs in [0,1] for subframe 4 | |
238 | */ | |
239 | static void interpolate_lsf(ACELPVContext *ctx, float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) | |
240 | { | |
241 | int i; | |
242 | ||
243 | for (i = 0; i < 4; i++) | |
244 | ctx->weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new, | |
245 | 0.25 * (3 - i), 0.25 * (i + 1), | |
246 | LP_FILTER_ORDER); | |
247 | } | |
248 | ||
249 | /** | |
250 | * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector. | |
251 | * | |
252 | * @param p the context | |
253 | * @param lsp output LSP vector | |
254 | * @param lsf_no_r LSF vector without the residual vector added | |
255 | * @param lsf_quantizer pointers to LSF dictionary tables | |
256 | * @param quantizer_offset offset in tables | |
257 | * @param sign for the 3 dictionary table | |
258 | * @param update store data for computing the next frame's LSFs | |
259 | */ | |
260 | static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER], | |
261 | const float lsf_no_r[LP_FILTER_ORDER], | |
262 | const int16_t *lsf_quantizer[5], | |
263 | const int quantizer_offset, | |
264 | const int sign, const int update) | |
265 | { | |
266 | int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector | |
267 | float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector | |
268 | int i; | |
269 | ||
270 | for (i = 0; i < LP_FILTER_ORDER >> 1; i++) | |
271 | memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset], | |
272 | 2 * sizeof(*lsf_r)); | |
273 | ||
274 | if (sign) { | |
275 | lsf_r[4] *= -1; | |
276 | lsf_r[5] *= -1; | |
277 | } | |
278 | ||
279 | if (update) | |
280 | memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); | |
281 | ||
282 | for (i = 0; i < LP_FILTER_ORDER; i++) | |
283 | lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0); | |
284 | ||
285 | ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); | |
286 | ||
287 | if (update) | |
288 | interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); | |
289 | ||
290 | ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER); | |
291 | } | |
292 | ||
293 | /** | |
294 | * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors. | |
295 | * | |
296 | * @param p pointer to the AMRContext | |
297 | */ | |
298 | static void lsf2lsp_5(AMRContext *p) | |
299 | { | |
300 | const uint16_t *lsf_param = p->frame.lsf; | |
301 | float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector | |
302 | const int16_t *lsf_quantizer[5]; | |
303 | int i; | |
304 | ||
305 | lsf_quantizer[0] = lsf_5_1[lsf_param[0]]; | |
306 | lsf_quantizer[1] = lsf_5_2[lsf_param[1]]; | |
307 | lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1]; | |
308 | lsf_quantizer[3] = lsf_5_4[lsf_param[3]]; | |
309 | lsf_quantizer[4] = lsf_5_5[lsf_param[4]]; | |
310 | ||
311 | for (i = 0; i < LP_FILTER_ORDER; i++) | |
312 | lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i]; | |
313 | ||
314 | lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0); | |
315 | lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1); | |
316 | ||
317 | // interpolate LSP vectors at subframes 1 and 3 | |
318 | weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER); | |
319 | weighted_vector_sumd(p->lsp[2], p->lsp[1] , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER); | |
320 | } | |
321 | ||
322 | /** | |
323 | * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector. | |
324 | * | |
325 | * @param p pointer to the AMRContext | |
326 | */ | |
327 | static void lsf2lsp_3(AMRContext *p) | |
328 | { | |
329 | const uint16_t *lsf_param = p->frame.lsf; | |
330 | int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector | |
331 | float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector | |
332 | const int16_t *lsf_quantizer; | |
333 | int i, j; | |
334 | ||
335 | lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]]; | |
336 | memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r)); | |
337 | ||
338 | lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)]; | |
339 | memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r)); | |
340 | ||
341 | lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]]; | |
342 | memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r)); | |
343 | ||
344 | // calculate mean-removed LSF vector and add mean | |
345 | for (i = 0; i < LP_FILTER_ORDER; i++) | |
346 | 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); | |
347 | ||
348 | ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER); | |
349 | ||
350 | // store data for computing the next frame's LSFs | |
351 | interpolate_lsf(&p->acelpv_ctx, p->lsf_q, lsf_q); | |
352 | memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r)); | |
353 | ||
354 | ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER); | |
355 | ||
356 | // interpolate LSP vectors at subframes 1, 2 and 3 | |
357 | for (i = 1; i <= 3; i++) | |
358 | for(j = 0; j < LP_FILTER_ORDER; j++) | |
359 | p->lsp[i-1][j] = p->prev_lsp_sub4[j] + | |
360 | (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i; | |
361 | } | |
362 | ||
363 | /// @} | |
364 | ||
365 | ||
366 | /// @name AMR pitch vector decoding functions | |
367 | /// @{ | |
368 | ||
369 | /** | |
370 | * Like ff_decode_pitch_lag(), but with 1/6 resolution | |
371 | */ | |
372 | static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index, | |
373 | const int prev_lag_int, const int subframe) | |
374 | { | |
375 | if (subframe == 0 || subframe == 2) { | |
376 | if (pitch_index < 463) { | |
377 | *lag_int = (pitch_index + 107) * 10923 >> 16; | |
378 | *lag_frac = pitch_index - *lag_int * 6 + 105; | |
379 | } else { | |
380 | *lag_int = pitch_index - 368; | |
381 | *lag_frac = 0; | |
382 | } | |
383 | } else { | |
384 | *lag_int = ((pitch_index + 5) * 10923 >> 16) - 1; | |
385 | *lag_frac = pitch_index - *lag_int * 6 - 3; | |
386 | *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2, | |
387 | PITCH_DELAY_MAX - 9); | |
388 | } | |
389 | } | |
390 | ||
391 | static void decode_pitch_vector(AMRContext *p, | |
392 | const AMRNBSubframe *amr_subframe, | |
393 | const int subframe) | |
394 | { | |
395 | int pitch_lag_int, pitch_lag_frac; | |
396 | enum Mode mode = p->cur_frame_mode; | |
397 | ||
398 | if (p->cur_frame_mode == MODE_12k2) { | |
399 | decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac, | |
400 | amr_subframe->p_lag, p->pitch_lag_int, | |
401 | subframe); | |
402 | } else | |
403 | ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac, | |
404 | amr_subframe->p_lag, | |
405 | p->pitch_lag_int, subframe, | |
406 | mode != MODE_4k75 && mode != MODE_5k15, | |
407 | mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6)); | |
408 | ||
409 | p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t | |
410 | ||
411 | pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2); | |
412 | ||
413 | pitch_lag_int += pitch_lag_frac > 0; | |
414 | ||
415 | /* Calculate the pitch vector by interpolating the past excitation at the | |
416 | pitch lag using a b60 hamming windowed sinc function. */ | |
417 | p->acelpf_ctx.acelp_interpolatef(p->excitation, | |
418 | p->excitation + 1 - pitch_lag_int, | |
419 | ff_b60_sinc, 6, | |
420 | pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0), | |
421 | 10, AMR_SUBFRAME_SIZE); | |
422 | ||
423 | memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float)); | |
424 | } | |
425 | ||
426 | /// @} | |
427 | ||
428 | ||
429 | /// @name AMR algebraic code book (fixed) vector decoding functions | |
430 | /// @{ | |
431 | ||
432 | /** | |
433 | * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame. | |
434 | */ | |
435 | static void decode_10bit_pulse(int code, int pulse_position[8], | |
436 | int i1, int i2, int i3) | |
437 | { | |
438 | // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of | |
439 | // the 3 pulses and the upper 7 bits being coded in base 5 | |
440 | const uint8_t *positions = base_five_table[code >> 3]; | |
441 | pulse_position[i1] = (positions[2] << 1) + ( code & 1); | |
442 | pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1); | |
443 | pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1); | |
444 | } | |
445 | ||
446 | /** | |
447 | * Decode the algebraic codebook index to pulse positions and signs and | |
448 | * construct the algebraic codebook vector for MODE_10k2. | |
449 | * | |
450 | * @param fixed_index positions of the eight pulses | |
451 | * @param fixed_sparse pointer to the algebraic codebook vector | |
452 | */ | |
453 | static void decode_8_pulses_31bits(const int16_t *fixed_index, | |
454 | AMRFixed *fixed_sparse) | |
455 | { | |
456 | int pulse_position[8]; | |
457 | int i, temp; | |
458 | ||
459 | decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1); | |
460 | decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5); | |
461 | ||
462 | // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of | |
463 | // the 2 pulses and the upper 5 bits being coded in base 5 | |
464 | temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5; | |
465 | pulse_position[3] = temp % 5; | |
466 | pulse_position[7] = temp / 5; | |
467 | if (pulse_position[7] & 1) | |
468 | pulse_position[3] = 4 - pulse_position[3]; | |
469 | pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1); | |
470 | pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1); | |
471 | ||
472 | fixed_sparse->n = 8; | |
473 | for (i = 0; i < 4; i++) { | |
474 | const int pos1 = (pulse_position[i] << 2) + i; | |
475 | const int pos2 = (pulse_position[i + 4] << 2) + i; | |
476 | const float sign = fixed_index[i] ? -1.0 : 1.0; | |
477 | fixed_sparse->x[i ] = pos1; | |
478 | fixed_sparse->x[i + 4] = pos2; | |
479 | fixed_sparse->y[i ] = sign; | |
480 | fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign; | |
481 | } | |
482 | } | |
483 | ||
484 | /** | |
485 | * Decode the algebraic codebook index to pulse positions and signs, | |
486 | * then construct the algebraic codebook vector. | |
487 | * | |
488 | * nb of pulses | bits encoding pulses | |
489 | * For MODE_4k75 or MODE_5k15, 2 | 1-3, 4-6, 7 | |
490 | * MODE_5k9, 2 | 1, 2-4, 5-6, 7-9 | |
491 | * MODE_6k7, 3 | 1-3, 4, 5-7, 8, 9-11 | |
492 | * MODE_7k4 or MODE_7k95, 4 | 1-3, 4-6, 7-9, 10, 11-13 | |
493 | * | |
494 | * @param fixed_sparse pointer to the algebraic codebook vector | |
495 | * @param pulses algebraic codebook indexes | |
496 | * @param mode mode of the current frame | |
497 | * @param subframe current subframe number | |
498 | */ | |
499 | static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses, | |
500 | const enum Mode mode, const int subframe) | |
501 | { | |
502 | av_assert1(MODE_4k75 <= (signed)mode && mode <= MODE_12k2); | |
503 | ||
504 | if (mode == MODE_12k2) { | |
505 | ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3); | |
506 | } else if (mode == MODE_10k2) { | |
507 | decode_8_pulses_31bits(pulses, fixed_sparse); | |
508 | } else { | |
509 | int *pulse_position = fixed_sparse->x; | |
510 | int i, pulse_subset; | |
511 | const int fixed_index = pulses[0]; | |
512 | ||
513 | if (mode <= MODE_5k15) { | |
514 | pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1); | |
515 | pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset]; | |
516 | pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1]; | |
517 | fixed_sparse->n = 2; | |
518 | } else if (mode == MODE_5k9) { | |
519 | pulse_subset = ((fixed_index & 1) << 1) + 1; | |
520 | pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset; | |
521 | pulse_subset = (fixed_index >> 4) & 3; | |
522 | pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0); | |
523 | fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2; | |
524 | } else if (mode == MODE_6k7) { | |
525 | pulse_position[0] = (fixed_index & 7) * 5; | |
526 | pulse_subset = (fixed_index >> 2) & 2; | |
527 | pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1; | |
528 | pulse_subset = (fixed_index >> 6) & 2; | |
529 | pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2; | |
530 | fixed_sparse->n = 3; | |
531 | } else { // mode <= MODE_7k95 | |
532 | pulse_position[0] = gray_decode[ fixed_index & 7]; | |
533 | pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1; | |
534 | pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2; | |
535 | pulse_subset = (fixed_index >> 9) & 1; | |
536 | pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3; | |
537 | fixed_sparse->n = 4; | |
538 | } | |
539 | for (i = 0; i < fixed_sparse->n; i++) | |
540 | fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0; | |
541 | } | |
542 | } | |
543 | ||
544 | /** | |
545 | * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2) | |
546 | * | |
547 | * @param p the context | |
548 | * @param subframe unpacked amr subframe | |
549 | * @param mode mode of the current frame | |
550 | * @param fixed_sparse sparse respresentation of the fixed vector | |
551 | */ | |
552 | static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode, | |
553 | AMRFixed *fixed_sparse) | |
554 | { | |
555 | // The spec suggests the current pitch gain is always used, but in other | |
556 | // modes the pitch and codebook gains are joinly quantized (sec 5.8.2) | |
557 | // so the codebook gain cannot depend on the quantized pitch gain. | |
558 | if (mode == MODE_12k2) | |
559 | p->beta = FFMIN(p->pitch_gain[4], 1.0); | |
560 | ||
561 | fixed_sparse->pitch_lag = p->pitch_lag_int; | |
562 | fixed_sparse->pitch_fac = p->beta; | |
563 | ||
564 | // Save pitch sharpening factor for the next subframe | |
565 | // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from | |
566 | // the fact that the gains for two subframes are jointly quantized. | |
567 | if (mode != MODE_4k75 || subframe & 1) | |
568 | p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX); | |
569 | } | |
570 | /// @} | |
571 | ||
572 | ||
573 | /// @name AMR gain decoding functions | |
574 | /// @{ | |
575 | ||
576 | /** | |
577 | * fixed gain smoothing | |
578 | * Note that where the spec specifies the "spectrum in the q domain" | |
579 | * in section 6.1.4, in fact frequencies should be used. | |
580 | * | |
581 | * @param p the context | |
582 | * @param lsf LSFs for the current subframe, in the range [0,1] | |
583 | * @param lsf_avg averaged LSFs | |
584 | * @param mode mode of the current frame | |
585 | * | |
586 | * @return fixed gain smoothed | |
587 | */ | |
588 | static float fixed_gain_smooth(AMRContext *p , const float *lsf, | |
589 | const float *lsf_avg, const enum Mode mode) | |
590 | { | |
591 | float diff = 0.0; | |
592 | int i; | |
593 | ||
594 | for (i = 0; i < LP_FILTER_ORDER; i++) | |
595 | diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i]; | |
596 | ||
597 | // If diff is large for ten subframes, disable smoothing for a 40-subframe | |
598 | // hangover period. | |
599 | p->diff_count++; | |
600 | if (diff <= 0.65) | |
601 | p->diff_count = 0; | |
602 | ||
603 | if (p->diff_count > 10) { | |
604 | p->hang_count = 0; | |
605 | p->diff_count--; // don't let diff_count overflow | |
606 | } | |
607 | ||
608 | if (p->hang_count < 40) { | |
609 | p->hang_count++; | |
610 | } else if (mode < MODE_7k4 || mode == MODE_10k2) { | |
611 | const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0); | |
612 | const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] + | |
613 | p->fixed_gain[2] + p->fixed_gain[3] + | |
614 | p->fixed_gain[4]) * 0.2; | |
615 | return smoothing_factor * p->fixed_gain[4] + | |
616 | (1.0 - smoothing_factor) * fixed_gain_mean; | |
617 | } | |
618 | return p->fixed_gain[4]; | |
619 | } | |
620 | ||
621 | /** | |
622 | * Decode pitch gain and fixed gain factor (part of section 6.1.3). | |
623 | * | |
624 | * @param p the context | |
625 | * @param amr_subframe unpacked amr subframe | |
626 | * @param mode mode of the current frame | |
627 | * @param subframe current subframe number | |
628 | * @param fixed_gain_factor decoded gain correction factor | |
629 | */ | |
630 | static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, | |
631 | const enum Mode mode, const int subframe, | |
632 | float *fixed_gain_factor) | |
633 | { | |
634 | if (mode == MODE_12k2 || mode == MODE_7k95) { | |
635 | p->pitch_gain[4] = qua_gain_pit [amr_subframe->p_gain ] | |
636 | * (1.0 / 16384.0); | |
637 | *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain] | |
638 | * (1.0 / 2048.0); | |
639 | } else { | |
640 | const uint16_t *gains; | |
641 | ||
642 | if (mode >= MODE_6k7) { | |
643 | gains = gains_high[amr_subframe->p_gain]; | |
644 | } else if (mode >= MODE_5k15) { | |
645 | gains = gains_low [amr_subframe->p_gain]; | |
646 | } else { | |
647 | // gain index is only coded in subframes 0,2 for MODE_4k75 | |
648 | gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)]; | |
649 | } | |
650 | ||
651 | p->pitch_gain[4] = gains[0] * (1.0 / 16384.0); | |
652 | *fixed_gain_factor = gains[1] * (1.0 / 4096.0); | |
653 | } | |
654 | } | |
655 | ||
656 | /// @} | |
657 | ||
658 | ||
659 | /// @name AMR preprocessing functions | |
660 | /// @{ | |
661 | ||
662 | /** | |
663 | * Circularly convolve a sparse fixed vector with a phase dispersion impulse | |
664 | * response filter (D.6.2 of G.729 and 6.1.5 of AMR). | |
665 | * | |
666 | * @param out vector with filter applied | |
667 | * @param in source vector | |
668 | * @param filter phase filter coefficients | |
669 | * | |
670 | * out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] } | |
671 | */ | |
672 | static void apply_ir_filter(float *out, const AMRFixed *in, | |
673 | const float *filter) | |
674 | { | |
675 | float filter1[AMR_SUBFRAME_SIZE], ///< filters at pitch lag*1 and *2 | |
676 | filter2[AMR_SUBFRAME_SIZE]; | |
677 | int lag = in->pitch_lag; | |
678 | float fac = in->pitch_fac; | |
679 | int i; | |
680 | ||
681 | if (lag < AMR_SUBFRAME_SIZE) { | |
682 | ff_celp_circ_addf(filter1, filter, filter, lag, fac, | |
683 | AMR_SUBFRAME_SIZE); | |
684 | ||
685 | if (lag < AMR_SUBFRAME_SIZE >> 1) | |
686 | ff_celp_circ_addf(filter2, filter, filter1, lag, fac, | |
687 | AMR_SUBFRAME_SIZE); | |
688 | } | |
689 | ||
690 | memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE); | |
691 | for (i = 0; i < in->n; i++) { | |
692 | int x = in->x[i]; | |
693 | float y = in->y[i]; | |
694 | const float *filterp; | |
695 | ||
696 | if (x >= AMR_SUBFRAME_SIZE - lag) { | |
697 | filterp = filter; | |
698 | } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) { | |
699 | filterp = filter1; | |
700 | } else | |
701 | filterp = filter2; | |
702 | ||
703 | ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE); | |
704 | } | |
705 | } | |
706 | ||
707 | /** | |
708 | * Reduce fixed vector sparseness by smoothing with one of three IR filters. | |
709 | * Also know as "adaptive phase dispersion". | |
710 | * | |
711 | * This implements 3GPP TS 26.090 section 6.1(5). | |
712 | * | |
713 | * @param p the context | |
714 | * @param fixed_sparse algebraic codebook vector | |
715 | * @param fixed_vector unfiltered fixed vector | |
716 | * @param fixed_gain smoothed gain | |
717 | * @param out space for modified vector if necessary | |
718 | */ | |
719 | static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse, | |
720 | const float *fixed_vector, | |
721 | float fixed_gain, float *out) | |
722 | { | |
723 | int ir_filter_nr; | |
724 | ||
725 | if (p->pitch_gain[4] < 0.6) { | |
726 | ir_filter_nr = 0; // strong filtering | |
727 | } else if (p->pitch_gain[4] < 0.9) { | |
728 | ir_filter_nr = 1; // medium filtering | |
729 | } else | |
730 | ir_filter_nr = 2; // no filtering | |
731 | ||
732 | // detect 'onset' | |
733 | if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) { | |
734 | p->ir_filter_onset = 2; | |
735 | } else if (p->ir_filter_onset) | |
736 | p->ir_filter_onset--; | |
737 | ||
738 | if (!p->ir_filter_onset) { | |
739 | int i, count = 0; | |
740 | ||
741 | for (i = 0; i < 5; i++) | |
742 | if (p->pitch_gain[i] < 0.6) | |
743 | count++; | |
744 | if (count > 2) | |
745 | ir_filter_nr = 0; | |
746 | ||
747 | if (ir_filter_nr > p->prev_ir_filter_nr + 1) | |
748 | ir_filter_nr--; | |
749 | } else if (ir_filter_nr < 2) | |
750 | ir_filter_nr++; | |
751 | ||
752 | // Disable filtering for very low level of fixed_gain. | |
753 | // Note this step is not specified in the technical description but is in | |
754 | // the reference source in the function Ph_disp. | |
755 | if (fixed_gain < 5.0) | |
756 | ir_filter_nr = 2; | |
757 | ||
758 | if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2 | |
759 | && ir_filter_nr < 2) { | |
760 | apply_ir_filter(out, fixed_sparse, | |
761 | (p->cur_frame_mode == MODE_7k95 ? | |
762 | ir_filters_lookup_MODE_7k95 : | |
763 | ir_filters_lookup)[ir_filter_nr]); | |
764 | fixed_vector = out; | |
765 | } | |
766 | ||
767 | // update ir filter strength history | |
768 | p->prev_ir_filter_nr = ir_filter_nr; | |
769 | p->prev_sparse_fixed_gain = fixed_gain; | |
770 | ||
771 | return fixed_vector; | |
772 | } | |
773 | ||
774 | /// @} | |
775 | ||
776 | ||
777 | /// @name AMR synthesis functions | |
778 | /// @{ | |
779 | ||
780 | /** | |
781 | * Conduct 10th order linear predictive coding synthesis. | |
782 | * | |
783 | * @param p pointer to the AMRContext | |
784 | * @param lpc pointer to the LPC coefficients | |
785 | * @param fixed_gain fixed codebook gain for synthesis | |
786 | * @param fixed_vector algebraic codebook vector | |
787 | * @param samples pointer to the output speech samples | |
788 | * @param overflow 16-bit overflow flag | |
789 | */ | |
790 | static int synthesis(AMRContext *p, float *lpc, | |
791 | float fixed_gain, const float *fixed_vector, | |
792 | float *samples, uint8_t overflow) | |
793 | { | |
794 | int i; | |
795 | float excitation[AMR_SUBFRAME_SIZE]; | |
796 | ||
797 | // if an overflow has been detected, the pitch vector is scaled down by a | |
798 | // factor of 4 | |
799 | if (overflow) | |
800 | for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
801 | p->pitch_vector[i] *= 0.25; | |
802 | ||
803 | p->acelpv_ctx.weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector, | |
804 | p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE); | |
805 | ||
806 | // emphasize pitch vector contribution | |
807 | if (p->pitch_gain[4] > 0.5 && !overflow) { | |
808 | float energy = p->celpm_ctx.dot_productf(excitation, excitation, | |
809 | AMR_SUBFRAME_SIZE); | |
810 | float pitch_factor = | |
811 | p->pitch_gain[4] * | |
812 | (p->cur_frame_mode == MODE_12k2 ? | |
813 | 0.25 * FFMIN(p->pitch_gain[4], 1.0) : | |
814 | 0.5 * FFMIN(p->pitch_gain[4], SHARP_MAX)); | |
815 | ||
816 | for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
817 | excitation[i] += pitch_factor * p->pitch_vector[i]; | |
818 | ||
819 | ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy, | |
820 | AMR_SUBFRAME_SIZE); | |
821 | } | |
822 | ||
823 | p->celpf_ctx.celp_lp_synthesis_filterf(samples, lpc, excitation, | |
824 | AMR_SUBFRAME_SIZE, | |
825 | LP_FILTER_ORDER); | |
826 | ||
827 | // detect overflow | |
828 | for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
829 | if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) { | |
830 | return 1; | |
831 | } | |
832 | ||
833 | return 0; | |
834 | } | |
835 | ||
836 | /// @} | |
837 | ||
838 | ||
839 | /// @name AMR update functions | |
840 | /// @{ | |
841 | ||
842 | /** | |
843 | * Update buffers and history at the end of decoding a subframe. | |
844 | * | |
845 | * @param p pointer to the AMRContext | |
846 | */ | |
847 | static void update_state(AMRContext *p) | |
848 | { | |
849 | memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0])); | |
850 | ||
851 | memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE], | |
852 | (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float)); | |
853 | ||
854 | memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float)); | |
855 | memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float)); | |
856 | ||
857 | memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE], | |
858 | LP_FILTER_ORDER * sizeof(float)); | |
859 | } | |
860 | ||
861 | /// @} | |
862 | ||
863 | ||
864 | /// @name AMR Postprocessing functions | |
865 | /// @{ | |
866 | ||
867 | /** | |
868 | * Get the tilt factor of a formant filter from its transfer function | |
869 | * | |
870 | * @param p The Context | |
871 | * @param lpc_n LP_FILTER_ORDER coefficients of the numerator | |
872 | * @param lpc_d LP_FILTER_ORDER coefficients of the denominator | |
873 | */ | |
874 | static float tilt_factor(AMRContext *p, float *lpc_n, float *lpc_d) | |
875 | { | |
876 | float rh0, rh1; // autocorrelation at lag 0 and 1 | |
877 | ||
878 | // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf | |
879 | float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 }; | |
880 | float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response | |
881 | ||
882 | hf[0] = 1.0; | |
883 | memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER); | |
884 | p->celpf_ctx.celp_lp_synthesis_filterf(hf, lpc_d, hf, | |
885 | AMR_TILT_RESPONSE, | |
886 | LP_FILTER_ORDER); | |
887 | ||
888 | rh0 = p->celpm_ctx.dot_productf(hf, hf, AMR_TILT_RESPONSE); | |
889 | rh1 = p->celpm_ctx.dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1); | |
890 | ||
891 | // The spec only specifies this check for 12.2 and 10.2 kbit/s | |
892 | // modes. But in the ref source the tilt is always non-negative. | |
893 | return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0; | |
894 | } | |
895 | ||
896 | /** | |
897 | * Perform adaptive post-filtering to enhance the quality of the speech. | |
898 | * See section 6.2.1. | |
899 | * | |
900 | * @param p pointer to the AMRContext | |
901 | * @param lpc interpolated LP coefficients for this subframe | |
902 | * @param buf_out output of the filter | |
903 | */ | |
904 | static void postfilter(AMRContext *p, float *lpc, float *buf_out) | |
905 | { | |
906 | int i; | |
907 | float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input | |
908 | ||
909 | float speech_gain = p->celpm_ctx.dot_productf(samples, samples, | |
910 | AMR_SUBFRAME_SIZE); | |
911 | ||
912 | float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter | |
913 | const float *gamma_n, *gamma_d; // Formant filter factor table | |
914 | float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients | |
915 | ||
916 | if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) { | |
917 | gamma_n = ff_pow_0_7; | |
918 | gamma_d = ff_pow_0_75; | |
919 | } else { | |
920 | gamma_n = ff_pow_0_55; | |
921 | gamma_d = ff_pow_0_7; | |
922 | } | |
923 | ||
924 | for (i = 0; i < LP_FILTER_ORDER; i++) { | |
925 | lpc_n[i] = lpc[i] * gamma_n[i]; | |
926 | lpc_d[i] = lpc[i] * gamma_d[i]; | |
927 | } | |
928 | ||
929 | memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER); | |
930 | p->celpf_ctx.celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, | |
931 | AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); | |
932 | memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE, | |
933 | sizeof(float) * LP_FILTER_ORDER); | |
934 | ||
935 | p->celpf_ctx.celp_lp_zero_synthesis_filterf(buf_out, lpc_n, | |
936 | pole_out + LP_FILTER_ORDER, | |
937 | AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); | |
938 | ||
939 | ff_tilt_compensation(&p->tilt_mem, tilt_factor(p, lpc_n, lpc_d), buf_out, | |
940 | AMR_SUBFRAME_SIZE); | |
941 | ||
942 | ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE, | |
943 | AMR_AGC_ALPHA, &p->postfilter_agc); | |
944 | } | |
945 | ||
946 | /// @} | |
947 | ||
948 | static int amrnb_decode_frame(AVCodecContext *avctx, void *data, | |
949 | int *got_frame_ptr, AVPacket *avpkt) | |
950 | { | |
951 | ||
952 | AMRContext *p = avctx->priv_data; // pointer to private data | |
953 | AVFrame *frame = data; | |
954 | const uint8_t *buf = avpkt->data; | |
955 | int buf_size = avpkt->size; | |
956 | float *buf_out; // pointer to the output data buffer | |
957 | int i, subframe, ret; | |
958 | float fixed_gain_factor; | |
959 | AMRFixed fixed_sparse = {0}; // fixed vector up to anti-sparseness processing | |
960 | float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from anti-sparseness processing | |
961 | float synth_fixed_gain; // the fixed gain that synthesis should use | |
962 | const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use | |
963 | ||
964 | /* get output buffer */ | |
965 | frame->nb_samples = AMR_BLOCK_SIZE; | |
966 | if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) | |
967 | return ret; | |
968 | buf_out = (float *)frame->data[0]; | |
969 | ||
970 | p->cur_frame_mode = unpack_bitstream(p, buf, buf_size); | |
971 | if (p->cur_frame_mode == NO_DATA) { | |
972 | av_log(avctx, AV_LOG_ERROR, "Corrupt bitstream\n"); | |
973 | return AVERROR_INVALIDDATA; | |
974 | } | |
975 | if (p->cur_frame_mode == MODE_DTX) { | |
976 | avpriv_report_missing_feature(avctx, "dtx mode"); | |
977 | av_log(avctx, AV_LOG_INFO, "Note: libopencore_amrnb supports dtx\n"); | |
978 | return AVERROR_PATCHWELCOME; | |
979 | } | |
980 | ||
981 | if (p->cur_frame_mode == MODE_12k2) { | |
982 | lsf2lsp_5(p); | |
983 | } else | |
984 | lsf2lsp_3(p); | |
985 | ||
986 | for (i = 0; i < 4; i++) | |
987 | ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5); | |
988 | ||
989 | for (subframe = 0; subframe < 4; subframe++) { | |
990 | const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe]; | |
991 | ||
992 | decode_pitch_vector(p, amr_subframe, subframe); | |
993 | ||
994 | decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses, | |
995 | p->cur_frame_mode, subframe); | |
996 | ||
997 | // The fixed gain (section 6.1.3) depends on the fixed vector | |
998 | // (section 6.1.2), but the fixed vector calculation uses | |
999 | // pitch sharpening based on the on the pitch gain (section 6.1.3). | |
1000 | // So the correct order is: pitch gain, pitch sharpening, fixed gain. | |
1001 | decode_gains(p, amr_subframe, p->cur_frame_mode, subframe, | |
1002 | &fixed_gain_factor); | |
1003 | ||
1004 | pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse); | |
1005 | ||
1006 | if (fixed_sparse.pitch_lag == 0) { | |
1007 | av_log(avctx, AV_LOG_ERROR, "The file is corrupted, pitch_lag = 0 is not allowed\n"); | |
1008 | return AVERROR_INVALIDDATA; | |
1009 | } | |
1010 | ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0, | |
1011 | AMR_SUBFRAME_SIZE); | |
1012 | ||
1013 | p->fixed_gain[4] = | |
1014 | ff_amr_set_fixed_gain(fixed_gain_factor, | |
1015 | p->celpm_ctx.dot_productf(p->fixed_vector, | |
1016 | p->fixed_vector, | |
1017 | AMR_SUBFRAME_SIZE) / | |
1018 | AMR_SUBFRAME_SIZE, | |
1019 | p->prediction_error, | |
1020 | energy_mean[p->cur_frame_mode], energy_pred_fac); | |
1021 | ||
1022 | // The excitation feedback is calculated without any processing such | |
1023 | // as fixed gain smoothing. This isn't mentioned in the specification. | |
1024 | for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
1025 | p->excitation[i] *= p->pitch_gain[4]; | |
1026 | ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4], | |
1027 | AMR_SUBFRAME_SIZE); | |
1028 | ||
1029 | // In the ref decoder, excitation is stored with no fractional bits. | |
1030 | // This step prevents buzz in silent periods. The ref encoder can | |
1031 | // emit long sequences with pitch factor greater than one. This | |
1032 | // creates unwanted feedback if the excitation vector is nonzero. | |
1033 | // (e.g. test sequence T19_795.COD in 3GPP TS 26.074) | |
1034 | for (i = 0; i < AMR_SUBFRAME_SIZE; i++) | |
1035 | p->excitation[i] = truncf(p->excitation[i]); | |
1036 | ||
1037 | // Smooth fixed gain. | |
1038 | // The specification is ambiguous, but in the reference source, the | |
1039 | // smoothed value is NOT fed back into later fixed gain smoothing. | |
1040 | synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe], | |
1041 | p->lsf_avg, p->cur_frame_mode); | |
1042 | ||
1043 | synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector, | |
1044 | synth_fixed_gain, spare_vector); | |
1045 | ||
1046 | if (synthesis(p, p->lpc[subframe], synth_fixed_gain, | |
1047 | synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0)) | |
1048 | // overflow detected -> rerun synthesis scaling pitch vector down | |
1049 | // by a factor of 4, skipping pitch vector contribution emphasis | |
1050 | // and adaptive gain control | |
1051 | synthesis(p, p->lpc[subframe], synth_fixed_gain, | |
1052 | synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1); | |
1053 | ||
1054 | postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); | |
1055 | ||
1056 | // update buffers and history | |
1057 | ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); | |
1058 | update_state(p); | |
1059 | } | |
1060 | ||
1061 | p->acelpf_ctx.acelp_apply_order_2_transfer_function(buf_out, | |
1062 | buf_out, highpass_zeros, | |
1063 | highpass_poles, | |
1064 | highpass_gain * AMR_SAMPLE_SCALE, | |
1065 | p->high_pass_mem, AMR_BLOCK_SIZE); | |
1066 | ||
1067 | /* Update averaged lsf vector (used for fixed gain smoothing). | |
1068 | * | |
1069 | * Note that lsf_avg should not incorporate the current frame's LSFs | |
1070 | * for fixed_gain_smooth. | |
1071 | * The specification has an incorrect formula: the reference decoder uses | |
1072 | * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */ | |
1073 | p->acelpv_ctx.weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3], | |
1074 | 0.84, 0.16, LP_FILTER_ORDER); | |
1075 | ||
1076 | *got_frame_ptr = 1; | |
1077 | ||
1078 | /* return the amount of bytes consumed if everything was OK */ | |
1079 | return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC | |
1080 | } | |
1081 | ||
1082 | ||
1083 | AVCodec ff_amrnb_decoder = { | |
1084 | .name = "amrnb", | |
1085 | .long_name = NULL_IF_CONFIG_SMALL("AMR-NB (Adaptive Multi-Rate NarrowBand)"), | |
1086 | .type = AVMEDIA_TYPE_AUDIO, | |
1087 | .id = AV_CODEC_ID_AMR_NB, | |
1088 | .priv_data_size = sizeof(AMRContext), | |
1089 | .init = amrnb_decode_init, | |
1090 | .decode = amrnb_decode_frame, | |
1091 | .capabilities = CODEC_CAP_DR1, | |
1092 | .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, | |
1093 | AV_SAMPLE_FMT_NONE }, | |
1094 | }; |