Go to most recent revision | Details | Last modification | View Log | RSS feed
Rev | Author | Line No. | Line |
---|---|---|---|
4349 | Serge | 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 |
||
44 | #include |
||
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 | };>>>>>>>>>>>>>>>>>><>>>>>><>>>=>>>=>><>><>=>=>=>>><>><>>><>><>><>><>><>=><=>=>>>=>>=>=>><>>>><>>>>>>><>><>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> |