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libavcodec/amrnbdec.c

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00001 /*
00002  * AMR narrowband decoder
00003  * Copyright (c) 2006-2007 Robert Swain
00004  * Copyright (c) 2009 Colin McQuillan
00005  *
00006  * This file is part of FFmpeg.
00007  *
00008  * FFmpeg is free software; you can redistribute it and/or
00009  * modify it under the terms of the GNU Lesser General Public
00010  * License as published by the Free Software Foundation; either
00011  * version 2.1 of the License, or (at your option) any later version.
00012  *
00013  * FFmpeg is distributed in the hope that it will be useful,
00014  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00015  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00016  * Lesser General Public License for more details.
00017  *
00018  * You should have received a copy of the GNU Lesser General Public
00019  * License along with FFmpeg; if not, write to the Free Software
00020  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00021  */
00022 
00023 
00043 #include <string.h>
00044 #include <math.h>
00045 
00046 #include "avcodec.h"
00047 #include "get_bits.h"
00048 #include "libavutil/common.h"
00049 #include "celp_math.h"
00050 #include "celp_filters.h"
00051 #include "acelp_filters.h"
00052 #include "acelp_vectors.h"
00053 #include "acelp_pitch_delay.h"
00054 #include "lsp.h"
00055 #include "amr.h"
00056 
00057 #include "amrnbdata.h"
00058 
00059 #define AMR_BLOCK_SIZE              160   ///< samples per frame
00060 #define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow
00061 
00071 #define AMR_SAMPLE_SCALE  (2.0 / 32768.0)
00072 
00074 #define PRED_FAC_MODE_12k2             0.65
00075 
00076 #define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
00077 #define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
00078 #define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode
00079 
00081 #define MIN_ENERGY -14.0
00082 
00088 #define SHARP_MAX 0.79449462890625
00089 
00091 #define AMR_TILT_RESPONSE   22
00092 
00093 #define AMR_TILT_GAMMA_T   0.8
00094 
00095 #define AMR_AGC_ALPHA      0.9
00096 
00097 typedef struct AMRContext {
00098     AMRNBFrame                        frame; 
00099     uint8_t             bad_frame_indicator; 
00100     enum Mode                cur_frame_mode;
00101 
00102     int16_t     prev_lsf_r[LP_FILTER_ORDER]; 
00103     double          lsp[4][LP_FILTER_ORDER]; 
00104     double   prev_lsp_sub4[LP_FILTER_ORDER]; 
00105 
00106     float         lsf_q[4][LP_FILTER_ORDER]; 
00107     float          lsf_avg[LP_FILTER_ORDER]; 
00108 
00109     float           lpc[4][LP_FILTER_ORDER]; 
00110 
00111     uint8_t                   pitch_lag_int; 
00112 
00113     float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; 
00114     float                       *excitation; 
00115 
00116     float   pitch_vector[AMR_SUBFRAME_SIZE]; 
00117     float   fixed_vector[AMR_SUBFRAME_SIZE]; 
00118 
00119     float               prediction_error[4]; 
00120     float                     pitch_gain[5]; 
00121     float                     fixed_gain[5]; 
00122 
00123     float                              beta; 
00124     uint8_t                      diff_count; 
00125     uint8_t                      hang_count; 
00126 
00127     float            prev_sparse_fixed_gain; 
00128     uint8_t               prev_ir_filter_nr; 
00129     uint8_t                 ir_filter_onset; 
00130 
00131     float                postfilter_mem[10]; 
00132     float                          tilt_mem; 
00133     float                    postfilter_agc; 
00134     float                  high_pass_mem[2]; 
00135 
00136     float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; 
00137 
00138 } AMRContext;
00139 
00141 static void weighted_vector_sumd(double *out, const double *in_a,
00142                                  const double *in_b, double weight_coeff_a,
00143                                  double weight_coeff_b, int length)
00144 {
00145     int i;
00146 
00147     for (i = 0; i < length; i++)
00148         out[i] = weight_coeff_a * in_a[i]
00149                + weight_coeff_b * in_b[i];
00150 }
00151 
00152 static av_cold int amrnb_decode_init(AVCodecContext *avctx)
00153 {
00154     AMRContext *p = avctx->priv_data;
00155     int i;
00156 
00157     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
00158 
00159     // p->excitation always points to the same position in p->excitation_buf
00160     p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
00161 
00162     for (i = 0; i < LP_FILTER_ORDER; i++) {
00163         p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
00164         p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
00165     }
00166 
00167     for (i = 0; i < 4; i++)
00168         p->prediction_error[i] = MIN_ENERGY;
00169 
00170     return 0;
00171 }
00172 
00173 
00185 static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
00186                                   int buf_size)
00187 {
00188     GetBitContext gb;
00189     enum Mode mode;
00190 
00191     init_get_bits(&gb, buf, buf_size * 8);
00192 
00193     // Decode the first octet.
00194     skip_bits(&gb, 1);                        // padding bit
00195     mode = get_bits(&gb, 4);                  // frame type
00196     p->bad_frame_indicator = !get_bits1(&gb); // quality bit
00197     skip_bits(&gb, 2);                        // two padding bits
00198 
00199     if (mode < MODE_DTX)
00200         ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
00201                            amr_unpacking_bitmaps_per_mode[mode]);
00202 
00203     return mode;
00204 }
00205 
00206 
00209 
00217 static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
00218 {
00219     int i;
00220 
00221     for (i = 0; i < 4; i++)
00222         ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
00223                                 0.25 * (3 - i), 0.25 * (i + 1),
00224                                 LP_FILTER_ORDER);
00225 }
00226 
00238 static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
00239                                  const float lsf_no_r[LP_FILTER_ORDER],
00240                                  const int16_t *lsf_quantizer[5],
00241                                  const int quantizer_offset,
00242                                  const int sign, const int update)
00243 {
00244     int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
00245     float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
00246     int i;
00247 
00248     for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
00249         memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
00250                2 * sizeof(*lsf_r));
00251 
00252     if (sign) {
00253         lsf_r[4] *= -1;
00254         lsf_r[5] *= -1;
00255     }
00256 
00257     if (update)
00258         memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
00259 
00260     for (i = 0; i < LP_FILTER_ORDER; i++)
00261         lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
00262 
00263     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
00264 
00265     if (update)
00266         interpolate_lsf(p->lsf_q, lsf_q);
00267 
00268     ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
00269 }
00270 
00276 static void lsf2lsp_5(AMRContext *p)
00277 {
00278     const uint16_t *lsf_param = p->frame.lsf;
00279     float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
00280     const int16_t *lsf_quantizer[5];
00281     int i;
00282 
00283     lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
00284     lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
00285     lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
00286     lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
00287     lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
00288 
00289     for (i = 0; i < LP_FILTER_ORDER; i++)
00290         lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
00291 
00292     lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
00293     lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
00294 
00295     // interpolate LSP vectors at subframes 1 and 3
00296     weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
00297     weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
00298 }
00299 
00305 static void lsf2lsp_3(AMRContext *p)
00306 {
00307     const uint16_t *lsf_param = p->frame.lsf;
00308     int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
00309     float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
00310     const int16_t *lsf_quantizer;
00311     int i, j;
00312 
00313     lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
00314     memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
00315 
00316     lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
00317     memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
00318 
00319     lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
00320     memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
00321 
00322     // calculate mean-removed LSF vector and add mean
00323     for (i = 0; i < LP_FILTER_ORDER; i++)
00324         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);
00325 
00326     ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
00327 
00328     // store data for computing the next frame's LSFs
00329     interpolate_lsf(p->lsf_q, lsf_q);
00330     memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
00331 
00332     ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
00333 
00334     // interpolate LSP vectors at subframes 1, 2 and 3
00335     for (i = 1; i <= 3; i++)
00336         for(j = 0; j < LP_FILTER_ORDER; j++)
00337             p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
00338                 (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
00339 }
00340 
00342 
00343 
00346 
00350 static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
00351                                  const int prev_lag_int, const int subframe)
00352 {
00353     if (subframe == 0 || subframe == 2) {
00354         if (pitch_index < 463) {
00355             *lag_int  = (pitch_index + 107) * 10923 >> 16;
00356             *lag_frac = pitch_index - *lag_int * 6 + 105;
00357         } else {
00358             *lag_int  = pitch_index - 368;
00359             *lag_frac = 0;
00360         }
00361     } else {
00362         *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
00363         *lag_frac = pitch_index - *lag_int * 6 - 3;
00364         *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
00365                             PITCH_DELAY_MAX - 9);
00366     }
00367 }
00368 
00369 static void decode_pitch_vector(AMRContext *p,
00370                                 const AMRNBSubframe *amr_subframe,
00371                                 const int subframe)
00372 {
00373     int pitch_lag_int, pitch_lag_frac;
00374     enum Mode mode = p->cur_frame_mode;
00375 
00376     if (p->cur_frame_mode == MODE_12k2) {
00377         decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
00378                              amr_subframe->p_lag, p->pitch_lag_int,
00379                              subframe);
00380     } else
00381         ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
00382                             amr_subframe->p_lag,
00383                             p->pitch_lag_int, subframe,
00384                             mode != MODE_4k75 && mode != MODE_5k15,
00385                             mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
00386 
00387     p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
00388 
00389     pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
00390 
00391     pitch_lag_int += pitch_lag_frac > 0;
00392 
00393     /* Calculate the pitch vector by interpolating the past excitation at the
00394        pitch lag using a b60 hamming windowed sinc function.   */
00395     ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
00396                           ff_b60_sinc, 6,
00397                           pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
00398                           10, AMR_SUBFRAME_SIZE);
00399 
00400     memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
00401 }
00402 
00404 
00405 
00408 
00412 static void decode_10bit_pulse(int code, int pulse_position[8],
00413                                int i1, int i2, int i3)
00414 {
00415     // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
00416     // the 3 pulses and the upper 7 bits being coded in base 5
00417     const uint8_t *positions = base_five_table[code >> 3];
00418     pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
00419     pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
00420     pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
00421 }
00422 
00430 static void decode_8_pulses_31bits(const int16_t *fixed_index,
00431                                    AMRFixed *fixed_sparse)
00432 {
00433     int pulse_position[8];
00434     int i, temp;
00435 
00436     decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
00437     decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
00438 
00439     // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
00440     // the 2 pulses and the upper 5 bits being coded in base 5
00441     temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
00442     pulse_position[3] = temp % 5;
00443     pulse_position[7] = temp / 5;
00444     if (pulse_position[7] & 1)
00445         pulse_position[3] = 4 - pulse_position[3];
00446     pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
00447     pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
00448 
00449     fixed_sparse->n = 8;
00450     for (i = 0; i < 4; i++) {
00451         const int pos1   = (pulse_position[i]     << 2) + i;
00452         const int pos2   = (pulse_position[i + 4] << 2) + i;
00453         const float sign = fixed_index[i] ? -1.0 : 1.0;
00454         fixed_sparse->x[i    ] = pos1;
00455         fixed_sparse->x[i + 4] = pos2;
00456         fixed_sparse->y[i    ] = sign;
00457         fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
00458     }
00459 }
00460 
00476 static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
00477                                 const enum Mode mode, const int subframe)
00478 {
00479     assert(MODE_4k75 <= mode && mode <= MODE_12k2);
00480 
00481     if (mode == MODE_12k2) {
00482         ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
00483     } else if (mode == MODE_10k2) {
00484         decode_8_pulses_31bits(pulses, fixed_sparse);
00485     } else {
00486         int *pulse_position = fixed_sparse->x;
00487         int i, pulse_subset;
00488         const int fixed_index = pulses[0];
00489 
00490         if (mode <= MODE_5k15) {
00491             pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
00492             pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
00493             pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
00494             fixed_sparse->n = 2;
00495         } else if (mode == MODE_5k9) {
00496             pulse_subset      = ((fixed_index & 1) << 1) + 1;
00497             pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
00498             pulse_subset      = (fixed_index  >> 4) & 3;
00499             pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
00500             fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
00501         } else if (mode == MODE_6k7) {
00502             pulse_position[0] = (fixed_index        & 7) * 5;
00503             pulse_subset      = (fixed_index  >> 2) & 2;
00504             pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
00505             pulse_subset      = (fixed_index  >> 6) & 2;
00506             pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
00507             fixed_sparse->n = 3;
00508         } else { // mode <= MODE_7k95
00509             pulse_position[0] = gray_decode[ fixed_index        & 7];
00510             pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
00511             pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
00512             pulse_subset      = (fixed_index >> 9) & 1;
00513             pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
00514             fixed_sparse->n = 4;
00515         }
00516         for (i = 0; i < fixed_sparse->n; i++)
00517             fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
00518     }
00519 }
00520 
00529 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
00530                              AMRFixed *fixed_sparse)
00531 {
00532     // The spec suggests the current pitch gain is always used, but in other
00533     // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
00534     // so the codebook gain cannot depend on the quantized pitch gain.
00535     if (mode == MODE_12k2)
00536         p->beta = FFMIN(p->pitch_gain[4], 1.0);
00537 
00538     fixed_sparse->pitch_lag  = p->pitch_lag_int;
00539     fixed_sparse->pitch_fac  = p->beta;
00540 
00541     // Save pitch sharpening factor for the next subframe
00542     // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
00543     // the fact that the gains for two subframes are jointly quantized.
00544     if (mode != MODE_4k75 || subframe & 1)
00545         p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
00546 }
00548 
00549 
00552 
00565 static float fixed_gain_smooth(AMRContext *p , const float *lsf,
00566                                const float *lsf_avg, const enum Mode mode)
00567 {
00568     float diff = 0.0;
00569     int i;
00570 
00571     for (i = 0; i < LP_FILTER_ORDER; i++)
00572         diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
00573 
00574     // If diff is large for ten subframes, disable smoothing for a 40-subframe
00575     // hangover period.
00576     p->diff_count++;
00577     if (diff <= 0.65)
00578         p->diff_count = 0;
00579 
00580     if (p->diff_count > 10) {
00581         p->hang_count = 0;
00582         p->diff_count--; // don't let diff_count overflow
00583     }
00584 
00585     if (p->hang_count < 40) {
00586         p->hang_count++;
00587     } else if (mode < MODE_7k4 || mode == MODE_10k2) {
00588         const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
00589         const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
00590                                        p->fixed_gain[2] + p->fixed_gain[3] +
00591                                        p->fixed_gain[4]) * 0.2;
00592         return smoothing_factor * p->fixed_gain[4] +
00593                (1.0 - smoothing_factor) * fixed_gain_mean;
00594     }
00595     return p->fixed_gain[4];
00596 }
00597 
00607 static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
00608                          const enum Mode mode, const int subframe,
00609                          float *fixed_gain_factor)
00610 {
00611     if (mode == MODE_12k2 || mode == MODE_7k95) {
00612         p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
00613             * (1.0 / 16384.0);
00614         *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
00615             * (1.0 /  2048.0);
00616     } else {
00617         const uint16_t *gains;
00618 
00619         if (mode >= MODE_6k7) {
00620             gains = gains_high[amr_subframe->p_gain];
00621         } else if (mode >= MODE_5k15) {
00622             gains = gains_low [amr_subframe->p_gain];
00623         } else {
00624             // gain index is only coded in subframes 0,2 for MODE_4k75
00625             gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
00626         }
00627 
00628         p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
00629         *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
00630     }
00631 }
00632 
00634 
00635 
00638 
00649 static void apply_ir_filter(float *out, const AMRFixed *in,
00650                             const float *filter)
00651 {
00652     float filter1[AMR_SUBFRAME_SIZE],     
00653           filter2[AMR_SUBFRAME_SIZE];
00654     int   lag = in->pitch_lag;
00655     float fac = in->pitch_fac;
00656     int i;
00657 
00658     if (lag < AMR_SUBFRAME_SIZE) {
00659         ff_celp_circ_addf(filter1, filter, filter, lag, fac,
00660                           AMR_SUBFRAME_SIZE);
00661 
00662         if (lag < AMR_SUBFRAME_SIZE >> 1)
00663             ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
00664                               AMR_SUBFRAME_SIZE);
00665     }
00666 
00667     memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
00668     for (i = 0; i < in->n; i++) {
00669         int   x = in->x[i];
00670         float y = in->y[i];
00671         const float *filterp;
00672 
00673         if (x >= AMR_SUBFRAME_SIZE - lag) {
00674             filterp = filter;
00675         } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
00676             filterp = filter1;
00677         } else
00678             filterp = filter2;
00679 
00680         ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
00681     }
00682 }
00683 
00696 static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
00697                                     const float *fixed_vector,
00698                                     float fixed_gain, float *out)
00699 {
00700     int ir_filter_nr;
00701 
00702     if (p->pitch_gain[4] < 0.6) {
00703         ir_filter_nr = 0;      // strong filtering
00704     } else if (p->pitch_gain[4] < 0.9) {
00705         ir_filter_nr = 1;      // medium filtering
00706     } else
00707         ir_filter_nr = 2;      // no filtering
00708 
00709     // detect 'onset'
00710     if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
00711         p->ir_filter_onset = 2;
00712     } else if (p->ir_filter_onset)
00713         p->ir_filter_onset--;
00714 
00715     if (!p->ir_filter_onset) {
00716         int i, count = 0;
00717 
00718         for (i = 0; i < 5; i++)
00719             if (p->pitch_gain[i] < 0.6)
00720                 count++;
00721         if (count > 2)
00722             ir_filter_nr = 0;
00723 
00724         if (ir_filter_nr > p->prev_ir_filter_nr + 1)
00725             ir_filter_nr--;
00726     } else if (ir_filter_nr < 2)
00727         ir_filter_nr++;
00728 
00729     // Disable filtering for very low level of fixed_gain.
00730     // Note this step is not specified in the technical description but is in
00731     // the reference source in the function Ph_disp.
00732     if (fixed_gain < 5.0)
00733         ir_filter_nr = 2;
00734 
00735     if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
00736          && ir_filter_nr < 2) {
00737         apply_ir_filter(out, fixed_sparse,
00738                         (p->cur_frame_mode == MODE_7k95 ?
00739                              ir_filters_lookup_MODE_7k95 :
00740                              ir_filters_lookup)[ir_filter_nr]);
00741         fixed_vector = out;
00742     }
00743 
00744     // update ir filter strength history
00745     p->prev_ir_filter_nr       = ir_filter_nr;
00746     p->prev_sparse_fixed_gain  = fixed_gain;
00747 
00748     return fixed_vector;
00749 }
00750 
00752 
00753 
00756 
00767 static int synthesis(AMRContext *p, float *lpc,
00768                      float fixed_gain, const float *fixed_vector,
00769                      float *samples, uint8_t overflow)
00770 {
00771     int i;
00772     float excitation[AMR_SUBFRAME_SIZE];
00773 
00774     // if an overflow has been detected, the pitch vector is scaled down by a
00775     // factor of 4
00776     if (overflow)
00777         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
00778             p->pitch_vector[i] *= 0.25;
00779 
00780     ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
00781                             p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
00782 
00783     // emphasize pitch vector contribution
00784     if (p->pitch_gain[4] > 0.5 && !overflow) {
00785         float energy = ff_dot_productf(excitation, excitation,
00786                                        AMR_SUBFRAME_SIZE);
00787         float pitch_factor =
00788             p->pitch_gain[4] *
00789             (p->cur_frame_mode == MODE_12k2 ?
00790                 0.25 * FFMIN(p->pitch_gain[4], 1.0) :
00791                 0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
00792 
00793         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
00794             excitation[i] += pitch_factor * p->pitch_vector[i];
00795 
00796         ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
00797                                                 AMR_SUBFRAME_SIZE);
00798     }
00799 
00800     ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
00801                                  LP_FILTER_ORDER);
00802 
00803     // detect overflow
00804     for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
00805         if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
00806             return 1;
00807         }
00808 
00809     return 0;
00810 }
00811 
00813 
00814 
00817 
00823 static void update_state(AMRContext *p)
00824 {
00825     memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
00826 
00827     memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
00828             (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
00829 
00830     memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
00831     memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
00832 
00833     memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
00834             LP_FILTER_ORDER * sizeof(float));
00835 }
00836 
00838 
00839 
00842 
00849 static float tilt_factor(float *lpc_n, float *lpc_d)
00850 {
00851     float rh0, rh1; // autocorrelation at lag 0 and 1
00852 
00853     // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
00854     float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
00855     float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
00856 
00857     hf[0] = 1.0;
00858     memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
00859     ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
00860                                  LP_FILTER_ORDER);
00861 
00862     rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
00863     rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
00864 
00865     // The spec only specifies this check for 12.2 and 10.2 kbit/s
00866     // modes. But in the ref source the tilt is always non-negative.
00867     return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
00868 }
00869 
00878 static void postfilter(AMRContext *p, float *lpc, float *buf_out)
00879 {
00880     int i;
00881     float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
00882 
00883     float speech_gain       = ff_dot_productf(samples, samples,
00884                                               AMR_SUBFRAME_SIZE);
00885 
00886     float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
00887     const float *gamma_n, *gamma_d;                       // Formant filter factor table
00888     float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
00889 
00890     if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
00891         gamma_n = ff_pow_0_7;
00892         gamma_d = ff_pow_0_75;
00893     } else {
00894         gamma_n = ff_pow_0_55;
00895         gamma_d = ff_pow_0_7;
00896     }
00897 
00898     for (i = 0; i < LP_FILTER_ORDER; i++) {
00899          lpc_n[i] = lpc[i] * gamma_n[i];
00900          lpc_d[i] = lpc[i] * gamma_d[i];
00901     }
00902 
00903     memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
00904     ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
00905                                  AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
00906     memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
00907            sizeof(float) * LP_FILTER_ORDER);
00908 
00909     ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
00910                                       pole_out + LP_FILTER_ORDER,
00911                                       AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
00912 
00913     ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
00914                          AMR_SUBFRAME_SIZE);
00915 
00916     ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
00917                              AMR_AGC_ALPHA, &p->postfilter_agc);
00918 }
00919 
00921 
00922 static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
00923                               AVPacket *avpkt)
00924 {
00925 
00926     AMRContext *p = avctx->priv_data;        // pointer to private data
00927     const uint8_t *buf = avpkt->data;
00928     int buf_size       = avpkt->size;
00929     float *buf_out = data;                   // pointer to the output data buffer
00930     int i, subframe;
00931     float fixed_gain_factor;
00932     AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
00933     float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
00934     float synth_fixed_gain;                  // the fixed gain that synthesis should use
00935     const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
00936 
00937     p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
00938     if (p->cur_frame_mode == MODE_DTX) {
00939         av_log_missing_feature(avctx, "dtx mode", 1);
00940         return -1;
00941     }
00942 
00943     if (p->cur_frame_mode == MODE_12k2) {
00944         lsf2lsp_5(p);
00945     } else
00946         lsf2lsp_3(p);
00947 
00948     for (i = 0; i < 4; i++)
00949         ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
00950 
00951     for (subframe = 0; subframe < 4; subframe++) {
00952         const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
00953 
00954         decode_pitch_vector(p, amr_subframe, subframe);
00955 
00956         decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
00957                             p->cur_frame_mode, subframe);
00958 
00959         // The fixed gain (section 6.1.3) depends on the fixed vector
00960         // (section 6.1.2), but the fixed vector calculation uses
00961         // pitch sharpening based on the on the pitch gain (section 6.1.3).
00962         // So the correct order is: pitch gain, pitch sharpening, fixed gain.
00963         decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
00964                      &fixed_gain_factor);
00965 
00966         pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
00967 
00968         ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
00969                             AMR_SUBFRAME_SIZE);
00970 
00971         p->fixed_gain[4] =
00972             ff_amr_set_fixed_gain(fixed_gain_factor,
00973                        ff_dot_productf(p->fixed_vector, p->fixed_vector,
00974                                        AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
00975                        p->prediction_error,
00976                        energy_mean[p->cur_frame_mode], energy_pred_fac);
00977 
00978         // The excitation feedback is calculated without any processing such
00979         // as fixed gain smoothing. This isn't mentioned in the specification.
00980         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
00981             p->excitation[i] *= p->pitch_gain[4];
00982         ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
00983                             AMR_SUBFRAME_SIZE);
00984 
00985         // In the ref decoder, excitation is stored with no fractional bits.
00986         // This step prevents buzz in silent periods. The ref encoder can
00987         // emit long sequences with pitch factor greater than one. This
00988         // creates unwanted feedback if the excitation vector is nonzero.
00989         // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
00990         for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
00991             p->excitation[i] = truncf(p->excitation[i]);
00992 
00993         // Smooth fixed gain.
00994         // The specification is ambiguous, but in the reference source, the
00995         // smoothed value is NOT fed back into later fixed gain smoothing.
00996         synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
00997                                              p->lsf_avg, p->cur_frame_mode);
00998 
00999         synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
01000                                              synth_fixed_gain, spare_vector);
01001 
01002         if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
01003                       synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
01004             // overflow detected -> rerun synthesis scaling pitch vector down
01005             // by a factor of 4, skipping pitch vector contribution emphasis
01006             // and adaptive gain control
01007             synthesis(p, p->lpc[subframe], synth_fixed_gain,
01008                       synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
01009 
01010         postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
01011 
01012         // update buffers and history
01013         ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
01014         update_state(p);
01015     }
01016 
01017     ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
01018                                              highpass_poles,
01019                                              highpass_gain * AMR_SAMPLE_SCALE,
01020                                              p->high_pass_mem, AMR_BLOCK_SIZE);
01021 
01022     /* Update averaged lsf vector (used for fixed gain smoothing).
01023      *
01024      * Note that lsf_avg should not incorporate the current frame's LSFs
01025      * for fixed_gain_smooth.
01026      * The specification has an incorrect formula: the reference decoder uses
01027      * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
01028     ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
01029                             0.84, 0.16, LP_FILTER_ORDER);
01030 
01031     /* report how many samples we got */
01032     *data_size = AMR_BLOCK_SIZE * sizeof(float);
01033 
01034     /* return the amount of bytes consumed if everything was OK */
01035     return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
01036 }
01037 
01038 
01039 AVCodec ff_amrnb_decoder = {
01040     .name           = "amrnb",
01041     .type           = AVMEDIA_TYPE_AUDIO,
01042     .id             = CODEC_ID_AMR_NB,
01043     .priv_data_size = sizeof(AMRContext),
01044     .init           = amrnb_decode_init,
01045     .decode         = amrnb_decode_frame,
01046     .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
01047     .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
01048 };

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