ffmpeg / libavcodec / amrwbdec.c @ b2ed95ec
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/*


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* AMR wideband decoder

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* Copyright (c) 2010 Marcelo Galvao Povoa

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*

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* This file is part of FFmpeg.

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*

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* FFmpeg is free software; you can redistribute it and/or

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* modify it under the terms of the GNU Lesser General Public

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* License as published by the Free Software Foundation; either

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* version 2.1 of the License, or (at your option) any later version.

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*

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* FFmpeg is distributed in the hope that it will be useful,

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* but WITHOUT ANY WARRANTY; without even the implied warranty of

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* MERCHANTABILITY or FITNESS FOR A particular PURPOSE. See the GNU

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* Lesser General Public License for more details.

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*

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* You should have received a copy of the GNU Lesser General Public

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* License along with FFmpeg; if not, write to the Free Software

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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 021101301 USA

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*/

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/**

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* @file

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* AMR wideband decoder

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*/

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#include "libavutil/lfg.h" 
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#include "avcodec.h" 
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#include "get_bits.h" 
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#include "lsp.h" 
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#include "celp_math.h" 
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#include "celp_filters.h" 
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#include "acelp_filters.h" 
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#include "acelp_vectors.h" 
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#include "acelp_pitch_delay.h" 
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#define AMR_USE_16BIT_TABLES

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#include "amr.h" 
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#include "amrwbdata.h" 
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typedef struct { 
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AMRWBFrame frame; ///< AMRWB parameters decoded from bitstream

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enum Mode fr_cur_mode; ///< mode index of current frame 
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uint8_t fr_quality; ///< frame quality index (FQI)

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float isf_cur[LP_ORDER]; ///< working ISF vector from current frame 
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float isf_q_past[LP_ORDER]; ///< quantized ISF vector of the previous frame 
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float isf_past_final[LP_ORDER]; ///< final processed ISF vector of the previous frame 
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double isp[4][LP_ORDER]; ///< ISP vectors from current frame 
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double isp_sub4_past[LP_ORDER]; ///< ISP vector for the 4th subframe of the previous frame 
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float lp_coef[4][LP_ORDER]; ///< Linear Prediction Coefficients from ISP vector 
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uint8_t base_pitch_lag; ///< integer part of pitch lag for the next relative subframe

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uint8_t pitch_lag_int; ///< integer part of pitch lag of the previous subframe

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float excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 2 + AMRWB_SFR_SIZE]; ///< current excitation and all necessary excitation history 
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float *excitation; ///< points to current excitation in excitation_buf[] 
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float pitch_vector[AMRWB_SFR_SIZE]; ///< adaptive codebook (pitch) vector for current subframe 
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float fixed_vector[AMRWB_SFR_SIZE]; ///< algebraic codebook (fixed) vector for current subframe 
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float prediction_error[4]; ///< quantified prediction errors {20log10(^gamma_gc)} for previous four subframes 
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float pitch_gain[6]; ///< quantified pitch gains for the current and previous five subframes 
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float fixed_gain[2]; ///< quantified fixed gains for the current and previous subframes 
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float tilt_coef; ///< {beta_1} related to the voicing of the previous subframe 
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float prev_sparse_fixed_gain; ///< previous fixed gain; used by antisparseness to determine "onset" 
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uint8_t prev_ir_filter_nr; ///< previous impulse response filter "impNr": 0  strong, 1  medium, 2  none

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float prev_tr_gain; ///< previous initial gain used by noise enhancer for threshold 
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float samples_az[LP_ORDER + AMRWB_SFR_SIZE]; ///< lowband samples and memory from synthesis at 12.8kHz 
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float samples_up[UPS_MEM_SIZE + AMRWB_SFR_SIZE]; ///< lowband samples and memory processed for upsampling 
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float samples_hb[LP_ORDER_16k + AMRWB_SFR_SIZE_16k]; ///< highband samples and memory from synthesis at 16kHz 
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float hpf_31_mem[2], hpf_400_mem[2]; ///< previous values in the high pass filters 
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float demph_mem[1]; ///< previous value in the deemphasis filter 
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float bpf_6_7_mem[HB_FIR_SIZE]; ///< previous values in the highband band pass filter 
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float lpf_7_mem[HB_FIR_SIZE]; ///< previous values in the highband low pass filter 
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AVLFG prng; ///< random number generator for white noise excitation

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uint8_t first_frame; ///< flag active during decoding of the first frame

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} AMRWBContext; 
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static av_cold int amrwb_decode_init(AVCodecContext *avctx) 
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{ 
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AMRWBContext *ctx = avctx>priv_data; 
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int i;

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avctx>sample_fmt = SAMPLE_FMT_FLT; 
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av_lfg_init(&ctx>prng, 1);

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ctx>excitation = &ctx>excitation_buf[AMRWB_P_DELAY_MAX + LP_ORDER + 1];

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ctx>first_frame = 1;

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for (i = 0; i < LP_ORDER; i++) 
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ctx>isf_past_final[i] = isf_init[i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 4; i++) 
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ctx>prediction_error[i] = MIN_ENERGY; 
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return 0; 
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} 
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/**

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* Decode the frame header in the "MIME/storage" format. This format

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* is simpler and does not carry the auxiliary information of the frame

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*

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* @param[in] ctx The Context

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* @param[in] buf Pointer to the input buffer

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*

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* @return The decoded header length in bytes

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*/

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static int decode_mime_header(AMRWBContext *ctx, const uint8_t *buf) 
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{ 
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GetBitContext gb; 
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init_get_bits(&gb, buf, 8);

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/* Decode frame header (1st octet) */

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skip_bits(&gb, 1); // padding bit 
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ctx>fr_cur_mode = get_bits(&gb, 4);

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ctx>fr_quality = get_bits1(&gb); 
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skip_bits(&gb, 2); // padding bits 
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return 1; 
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} 
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/**

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* Decodes quantized ISF vectors using 36bit indexes (6K60 mode only)

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*

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* @param[in] ind Array of 5 indexes

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* @param[out] isf_q Buffer for isf_q[LP_ORDER]

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*

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*/

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static void decode_isf_indices_36b(uint16_t *ind, float *isf_q) 
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{ 
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int i;

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for (i = 0; i < 9; i++) 
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isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 7; i++) 
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isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 5; i++) 
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isf_q[i] += dico21_isf_36b[ind[2]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 4; i++) 
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isf_q[i + 5] += dico22_isf_36b[ind[3]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 7; i++) 
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isf_q[i + 9] += dico23_isf_36b[ind[4]][i] * (1.0f / (1 << 15)); 
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} 
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/**

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* Decodes quantized ISF vectors using 46bit indexes (except 6K60 mode)

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*

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* @param[in] ind Array of 7 indexes

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* @param[out] isf_q Buffer for isf_q[LP_ORDER]

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*

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*/

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static void decode_isf_indices_46b(uint16_t *ind, float *isf_q) 
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{ 
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int i;

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for (i = 0; i < 9; i++) 
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isf_q[i] = dico1_isf[ind[0]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 7; i++) 
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isf_q[i + 9] = dico2_isf[ind[1]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 3; i++) 
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isf_q[i] += dico21_isf[ind[2]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 3; i++) 
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isf_q[i + 3] += dico22_isf[ind[3]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 3; i++) 
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isf_q[i + 6] += dico23_isf[ind[4]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 3; i++) 
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isf_q[i + 9] += dico24_isf[ind[5]][i] * (1.0f / (1 << 15)); 
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for (i = 0; i < 4; i++) 
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isf_q[i + 12] += dico25_isf[ind[6]][i] * (1.0f / (1 << 15)); 
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} 
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/**

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* Apply mean and past ISF values using the prediction factor

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* Updates past ISF vector

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*

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* @param[in,out] isf_q Current quantized ISF

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* @param[in,out] isf_past Past quantized ISF

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*

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*/

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static void isf_add_mean_and_past(float *isf_q, float *isf_past) 
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{ 
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int i;

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float tmp;

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for (i = 0; i < LP_ORDER; i++) { 
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tmp = isf_q[i]; 
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isf_q[i] += isf_mean[i] * (1.0f / (1 << 15)); 
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isf_q[i] += PRED_FACTOR * isf_past[i]; 
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isf_past[i] = tmp; 
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} 
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} 
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/**

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* Interpolate the fourth ISP vector from current and past frames

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* to obtain a ISP vector for each subframe

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*

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* @param[in,out] isp_q ISPs for each subframe

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* @param[in] isp4_past Past ISP for subframe 4

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*/

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static void interpolate_isp(double isp_q[4][LP_ORDER], const double *isp4_past) 
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{ 
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int i, k;

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for (k = 0; k < 3; k++) { 
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float c = isfp_inter[k];

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for (i = 0; i < LP_ORDER; i++) 
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isp_q[k][i] = (1.0  c) * isp4_past[i] + c * isp_q[3][i]; 
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} 
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} 
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/**

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* Decode an adaptive codebook index into pitch lag (except 6k60, 8k85 modes)

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* Calculate integer lag and fractional lag always using 1/4 resolution

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* In 1st and 3rd subframes the index is relative to last subframe integer lag

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*

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* @param[out] lag_int Decoded integer pitch lag

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* @param[out] lag_frac Decoded fractional pitch lag

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* @param[in] pitch_index Adaptive codebook pitch index

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* @param[in,out] base_lag_int Base integer lag used in relative subframes

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* @param[in] subframe Current subframe index (0 to 3)

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*/

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static void decode_pitch_lag_high(int *lag_int, int *lag_frac, int pitch_index, 
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uint8_t *base_lag_int, int subframe)

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{ 
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if (subframe == 0  subframe == 2) { 
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if (pitch_index < 376) { 
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*lag_int = (pitch_index + 137) >> 2; 
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*lag_frac = pitch_index  (*lag_int << 2) + 136; 
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} else if (pitch_index < 440) { 
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*lag_int = (pitch_index + 257  376) >> 1; 
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*lag_frac = (pitch_index  (*lag_int << 1) + 256  376) << 1; 
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/* the actual resolution is 1/2 but expressed as 1/4 */

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} else {

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*lag_int = pitch_index  280;

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*lag_frac = 0;

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} 
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/* minimum lag for next subframe */

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*base_lag_int = av_clip(*lag_int  8  (*lag_frac < 0), 
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AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX  15);

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// XXX: the spec states clearly that *base_lag_int should be

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// the nearest integer to *lag_int (minus 8), but the ref code

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// actually always uses its floor, I'm following the latter

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} else {

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*lag_int = (pitch_index + 1) >> 2; 
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*lag_frac = pitch_index  (*lag_int << 2);

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*lag_int += *base_lag_int; 
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} 
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} 
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/**

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* Decode a adaptive codebook index into pitch lag for 8k85 and 6k60 modes

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* Description is analogous to decode_pitch_lag_high, but in 6k60 relative

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* index is used for all subframes except the first

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*/

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static void decode_pitch_lag_low(int *lag_int, int *lag_frac, int pitch_index, 
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uint8_t *base_lag_int, int subframe, enum Mode mode) 
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{ 
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if (subframe == 0  (subframe == 2 && mode != MODE_6k60)) { 
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if (pitch_index < 116) { 
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*lag_int = (pitch_index + 69) >> 1; 
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*lag_frac = (pitch_index  (*lag_int << 1) + 68) << 1; 
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} else {

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*lag_int = pitch_index  24;

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*lag_frac = 0;

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} 
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// XXX: same problem as before

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*base_lag_int = av_clip(*lag_int  8  (*lag_frac < 0), 
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AMRWB_P_DELAY_MIN, AMRWB_P_DELAY_MAX  15);

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} else {

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*lag_int = (pitch_index + 1) >> 1; 
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*lag_frac = (pitch_index  (*lag_int << 1)) << 1; 
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*lag_int += *base_lag_int; 
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} 
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} 
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/**

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* Find the pitch vector by interpolating the past excitation at the

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* pitch delay, which is obtained in this function

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*

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* @param[in,out] ctx The context

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* @param[in] amr_subframe Current subframe data

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* @param[in] subframe Current subframe index (0 to 3)

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*/

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static void decode_pitch_vector(AMRWBContext *ctx, 
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const AMRWBSubFrame *amr_subframe,

305 
const int subframe) 
306 
{ 
307 
int pitch_lag_int, pitch_lag_frac;

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int i;

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float *exc = ctx>excitation;

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enum Mode mode = ctx>fr_cur_mode;

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if (mode <= MODE_8k85) {

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decode_pitch_lag_low(&pitch_lag_int, &pitch_lag_frac, amr_subframe>adap, 
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&ctx>base_pitch_lag, subframe, mode); 
315 
} else

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decode_pitch_lag_high(&pitch_lag_int, &pitch_lag_frac, amr_subframe>adap, 
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&ctx>base_pitch_lag, subframe); 
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ctx>pitch_lag_int = pitch_lag_int; 
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pitch_lag_int += pitch_lag_frac > 0;

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/* Calculate the pitch vector by interpolating the past excitation at the

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pitch lag using a hamming windowed sinc function */

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ff_acelp_interpolatef(exc, exc + 1  pitch_lag_int,

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ac_inter, 4,

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pitch_lag_frac + (pitch_lag_frac > 0 ? 0 : 4), 
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LP_ORDER, AMRWB_SFR_SIZE + 1);

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/* Check which pitch signal path should be used

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* 6k60 and 8k85 modes have the ltp flag set to 0 */

331 
if (amr_subframe>ltp) {

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memcpy(ctx>pitch_vector, exc, AMRWB_SFR_SIZE * sizeof(float)); 
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} else {

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for (i = 0; i < AMRWB_SFR_SIZE; i++) 
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ctx>pitch_vector[i] = 0.18 * exc[i  1] + 0.64 * exc[i] + 
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0.18 * exc[i + 1]; 
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memcpy(exc, ctx>pitch_vector, AMRWB_SFR_SIZE * sizeof(float)); 
338 
} 
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} 
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/** Get x bits in the index interval [lsb,lsb+len1] inclusive */

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#define BIT_STR(x,lsb,len) (((x) >> (lsb)) & ((1 << (len))  1)) 
343  
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/** Get the bit at specified position */

345 
#define BIT_POS(x, p) (((x) >> (p)) & 1) 
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/**

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* The next six functions decode_[i]p_track decode exactly i pulses

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* positions and amplitudes (1 or 1) in a subframe track using

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* an encoded pulse indexing (TS 26.190 section 5.8.2)

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*

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* The results are given in out[], in which a negative number means

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* amplitude 1 and vice versa (i.e., ampl(x) = x / abs(x) )

354 
*

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* @param[out] out Output buffer (writes i elements)

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* @param[in] code Pulse index (no. of bits varies, see below)

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* @param[in] m (log2) Number of potential positions

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* @param[in] off Offset for decoded positions

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*/

360 
static inline void decode_1p_track(int *out, int code, int m, int off) 
361 
{ 
362 
int pos = BIT_STR(code, 0, m) + off; ///code: m+1 bits 
363  
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out[0] = BIT_POS(code, m) ? pos : pos;

365 
} 
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static inline void decode_2p_track(int *out, int code, int m, int off) ///code: 2m+1 bits 
368 
{ 
369 
int pos0 = BIT_STR(code, m, m) + off;

370 
int pos1 = BIT_STR(code, 0, m) + off; 
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out[0] = BIT_POS(code, 2*m) ? pos0 : pos0; 
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out[1] = BIT_POS(code, 2*m) ? pos1 : pos1; 
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out[1] = pos0 > pos1 ? out[1] : out[1]; 
375 
} 
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377 
static void decode_3p_track(int *out, int code, int m, int off) ///code: 3m+1 bits 
378 
{ 
379 
int half_2p = BIT_POS(code, 2*m  1) << (m  1); 
380  
381 
decode_2p_track(out, BIT_STR(code, 0, 2*m  1), 
382 
m  1, off + half_2p);

383 
decode_1p_track(out + 2, BIT_STR(code, 2*m, m + 1), m, off); 
384 
} 
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386 
static void decode_4p_track(int *out, int code, int m, int off) ///code: 4m bits 
387 
{ 
388 
int half_4p, subhalf_2p;

389 
int b_offset = 1 << (m  1); 
390  
391 
switch (BIT_STR(code, 4*m  2, 2)) { /* case ID (2 bits) */ 
392 
case 0: /* 0 pulses in A, 4 pulses in B or vice versa */ 
393 
half_4p = BIT_POS(code, 4*m  3) << (m  1); // which has 4 pulses 
394 
subhalf_2p = BIT_POS(code, 2*m  3) << (m  2); 
395  
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decode_2p_track(out, BIT_STR(code, 0, 2*m  3), 
397 
m  2, off + half_4p + subhalf_2p);

398 
decode_2p_track(out + 2, BIT_STR(code, 2*m  2, 2*m  1), 
399 
m  1, off + half_4p);

400 
break;

401 
case 1: /* 1 pulse in A, 3 pulses in B */ 
402 
decode_1p_track(out, BIT_STR(code, 3*m  2, m), 
403 
m  1, off);

404 
decode_3p_track(out + 1, BIT_STR(code, 0, 3*m  2), 
405 
m  1, off + b_offset);

406 
break;

407 
case 2: /* 2 pulses in each half */ 
408 
decode_2p_track(out, BIT_STR(code, 2*m  1, 2*m  1), 
409 
m  1, off);

410 
decode_2p_track(out + 2, BIT_STR(code, 0, 2*m  1), 
411 
m  1, off + b_offset);

412 
break;

413 
case 3: /* 3 pulses in A, 1 pulse in B */ 
414 
decode_3p_track(out, BIT_STR(code, m, 3*m  2), 
415 
m  1, off);

416 
decode_1p_track(out + 3, BIT_STR(code, 0, m), 
417 
m  1, off + b_offset);

418 
break;

419 
} 
420 
} 
421  
422 
static void decode_5p_track(int *out, int code, int m, int off) ///code: 5m bits 
423 
{ 
424 
int half_3p = BIT_POS(code, 5*m  1) << (m  1); 
425  
426 
decode_3p_track(out, BIT_STR(code, 2*m + 1, 3*m  2), 
427 
m  1, off + half_3p);

428  
429 
decode_2p_track(out + 3, BIT_STR(code, 0, 2*m + 1), m, off); 
430 
} 
431  
432 
static void decode_6p_track(int *out, int code, int m, int off) ///code: 6m2 bits 
433 
{ 
434 
int b_offset = 1 << (m  1); 
435 
/* which half has more pulses in cases 0 to 2 */

436 
int half_more = BIT_POS(code, 6*m  5) << (m  1); 
437 
int half_other = b_offset  half_more;

438  
439 
switch (BIT_STR(code, 6*m  4, 2)) { /* case ID (2 bits) */ 
440 
case 0: /* 0 pulses in A, 6 pulses in B or vice versa */ 
441 
decode_1p_track(out, BIT_STR(code, 0, m),

442 
m  1, off + half_more);

443 
decode_5p_track(out + 1, BIT_STR(code, m, 5*m  5), 
444 
m  1, off + half_more);

445 
break;

446 
case 1: /* 1 pulse in A, 5 pulses in B or vice versa */ 
447 
decode_1p_track(out, BIT_STR(code, 0, m),

448 
m  1, off + half_other);

449 
decode_5p_track(out + 1, BIT_STR(code, m, 5*m  5), 
450 
m  1, off + half_more);

451 
break;

452 
case 2: /* 2 pulses in A, 4 pulses in B or vice versa */ 
453 
decode_2p_track(out, BIT_STR(code, 0, 2*m  1), 
454 
m  1, off + half_other);

455 
decode_4p_track(out + 2, BIT_STR(code, 2*m  1, 4*m  4), 
456 
m  1, off + half_more);

457 
break;

458 
case 3: /* 3 pulses in A, 3 pulses in B */ 
459 
decode_3p_track(out, BIT_STR(code, 3*m  2, 3*m  2), 
460 
m  1, off);

461 
decode_3p_track(out + 3, BIT_STR(code, 0, 3*m  2), 
462 
m  1, off + b_offset);

463 
break;

464 
} 
465 
} 
466  
467 
/**

468 
* Decode the algebraic codebook index to pulse positions and signs,

469 
* then construct the algebraic codebook vector

470 
*

471 
* @param[out] fixed_vector Buffer for the fixed codebook excitation

472 
* @param[in] pulse_hi MSBs part of the pulse index array (higher modes only)

473 
* @param[in] pulse_lo LSBs part of the pulse index array

474 
* @param[in] mode Mode of the current frame

475 
*/

476 
static void decode_fixed_vector(float *fixed_vector, const uint16_t *pulse_hi, 
477 
const uint16_t *pulse_lo, const enum Mode mode) 
478 
{ 
479 
/* sig_pos stores for each track the decoded pulse position indexes

480 
* (1based) multiplied by its corresponding amplitude (+1 or 1) */

481 
int sig_pos[4][6]; 
482 
int spacing = (mode == MODE_6k60) ? 2 : 4; 
483 
int i, j;

484  
485 
switch (mode) {

486 
case MODE_6k60:

487 
for (i = 0; i < 2; i++) 
488 
decode_1p_track(sig_pos[i], pulse_lo[i], 5, 1); 
489 
break;

490 
case MODE_8k85:

491 
for (i = 0; i < 4; i++) 
492 
decode_1p_track(sig_pos[i], pulse_lo[i], 4, 1); 
493 
break;

494 
case MODE_12k65:

495 
for (i = 0; i < 4; i++) 
496 
decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1); 
497 
break;

498 
case MODE_14k25:

499 
for (i = 0; i < 2; i++) 
500 
decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1); 
501 
for (i = 2; i < 4; i++) 
502 
decode_2p_track(sig_pos[i], pulse_lo[i], 4, 1); 
503 
break;

504 
case MODE_15k85:

505 
for (i = 0; i < 4; i++) 
506 
decode_3p_track(sig_pos[i], pulse_lo[i], 4, 1); 
507 
break;

508 
case MODE_18k25:

509 
for (i = 0; i < 4; i++) 
510 
decode_4p_track(sig_pos[i], (int) pulse_lo[i] +

511 
((int) pulse_hi[i] << 14), 4, 1); 
512 
break;

513 
case MODE_19k85:

514 
for (i = 0; i < 2; i++) 
515 
decode_5p_track(sig_pos[i], (int) pulse_lo[i] +

516 
((int) pulse_hi[i] << 10), 4, 1); 
517 
for (i = 2; i < 4; i++) 
518 
decode_4p_track(sig_pos[i], (int) pulse_lo[i] +

519 
((int) pulse_hi[i] << 14), 4, 1); 
520 
break;

521 
case MODE_23k05:

522 
case MODE_23k85:

523 
for (i = 0; i < 4; i++) 
524 
decode_6p_track(sig_pos[i], (int) pulse_lo[i] +

525 
((int) pulse_hi[i] << 11), 4, 1); 
526 
break;

527 
} 
528  
529 
memset(fixed_vector, 0, sizeof(float) * AMRWB_SFR_SIZE); 
530  
531 
for (i = 0; i < 4; i++) 
532 
for (j = 0; j < pulses_nb_per_mode_tr[mode][i]; j++) { 
533 
int pos = (FFABS(sig_pos[i][j])  1) * spacing + i; 
534  
535 
fixed_vector[pos] += sig_pos[i][j] < 0 ? 1.0 : 1.0; 
536 
} 
537 
} 
538  
539 
/**

540 
* Decode pitch gain and fixed gain correction factor

541 
*

542 
* @param[in] vq_gain Vectorquantized index for gains

543 
* @param[in] mode Mode of the current frame

544 
* @param[out] fixed_gain_factor Decoded fixed gain correction factor

545 
* @param[out] pitch_gain Decoded pitch gain

546 
*/

547 
static void decode_gains(const uint8_t vq_gain, const enum Mode mode, 
548 
float *fixed_gain_factor, float *pitch_gain) 
549 
{ 
550 
const int16_t *gains = (mode <= MODE_8k85 ? qua_gain_6b[vq_gain] :

551 
qua_gain_7b[vq_gain]); 
552  
553 
*pitch_gain = gains[0] * (1.0f / (1 << 14)); 
554 
*fixed_gain_factor = gains[1] * (1.0f / (1 << 11)); 
555 
} 
556  
557 
/**

558 
* Apply pitch sharpening filters to the fixed codebook vector

559 
*

560 
* @param[in] ctx The context

561 
* @param[in,out] fixed_vector Fixed codebook excitation

562 
*/

563 
// XXX: Spec states this procedure should be applied when the pitch

564 
// lag is less than 64, but this checking seems absent in reference and AMRNB

565 
static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector) 
566 
{ 
567 
int i;

568  
569 
/* Tilt part */

570 
for (i = AMRWB_SFR_SIZE  1; i != 0; i) 
571 
fixed_vector[i] = fixed_vector[i  1] * ctx>tilt_coef;

572  
573 
/* Periodicity enhancement part */

574 
for (i = ctx>pitch_lag_int; i < AMRWB_SFR_SIZE; i++)

575 
fixed_vector[i] += fixed_vector[i  ctx>pitch_lag_int] * 0.85; 
576 
} 
577  
578 
/**

579 
* Calculate the voicing factor (1.0 = unvoiced to 1.0 = voiced)

580 
*

581 
* @param[in] p_vector, f_vector Pitch and fixed excitation vectors

582 
* @param[in] p_gain, f_gain Pitch and fixed gains

583 
*/

584 
// XXX: There is something wrong with the precision here! The magnitudes

585 
// of the energies are not correct. Please check the reference code carefully

586 
static float voice_factor(float *p_vector, float p_gain, 
587 
float *f_vector, float f_gain) 
588 
{ 
589 
double p_ener = (double) ff_dot_productf(p_vector, p_vector, 
590 
AMRWB_SFR_SIZE) * p_gain * p_gain; 
591 
double f_ener = (double) ff_dot_productf(f_vector, f_vector, 
592 
AMRWB_SFR_SIZE) * f_gain * f_gain; 
593  
594 
return (p_ener  f_ener) / (p_ener + f_ener);

595 
} 
596  
597 
/**

598 
* Reduce fixed vector sparseness by smoothing with one of three IR filters

599 
* Also known as "adaptive phase dispersion"

600 
*

601 
* @param[in] ctx The context

602 
* @param[in,out] fixed_vector Unfiltered fixed vector

603 
* @param[out] buf Space for modified vector if necessary

604 
*

605 
* @return The potentially overwritten filtered fixed vector address

606 
*/

607 
static float *anti_sparseness(AMRWBContext *ctx, 
608 
float *fixed_vector, float *buf) 
609 
{ 
610 
int ir_filter_nr;

611  
612 
if (ctx>fr_cur_mode > MODE_8k85) // no filtering in higher modes 
613 
return fixed_vector;

614  
615 
if (ctx>pitch_gain[0] < 0.6) { 
616 
ir_filter_nr = 0; // strong filtering 
617 
} else if (ctx>pitch_gain[0] < 0.9) { 
618 
ir_filter_nr = 1; // medium filtering 
619 
} else

620 
ir_filter_nr = 2; // no filtering 
621  
622 
/* detect 'onset' */

623 
if (ctx>fixed_gain[0] > 3.0 * ctx>fixed_gain[1]) { 
624 
if (ir_filter_nr < 2) 
625 
ir_filter_nr++; 
626 
} else {

627 
int i, count = 0; 
628  
629 
for (i = 0; i < 6; i++) 
630 
if (ctx>pitch_gain[i] < 0.6) 
631 
count++; 
632  
633 
if (count > 2) 
634 
ir_filter_nr = 0;

635  
636 
if (ir_filter_nr > ctx>prev_ir_filter_nr + 1) 
637 
ir_filter_nr; 
638 
} 
639  
640 
/* update ir filter strength history */

641 
ctx>prev_ir_filter_nr = ir_filter_nr; 
642  
643 
ir_filter_nr += (ctx>fr_cur_mode == MODE_8k85); 
644  
645 
if (ir_filter_nr < 2) { 
646 
int i;

647 
const float *coef = ir_filters_lookup[ir_filter_nr]; 
648  
649 
/* Circular convolution code in the reference

650 
* decoder was modified to avoid using one

651 
* extra array. The filtered vector is given by:

652 
*

653 
* c2(n) = sum(i,0,len1){ c(i) * coef( (n  i + len) % len ) }

654 
*/

655  
656 
memset(buf, 0, sizeof(float) * AMRWB_SFR_SIZE); 
657 
for (i = 0; i < AMRWB_SFR_SIZE; i++) 
658 
if (fixed_vector[i])

659 
ff_celp_circ_addf(buf, buf, coef, i, fixed_vector[i], 
660 
AMRWB_SFR_SIZE); 
661 
fixed_vector = buf; 
662 
} 
663  
664 
return fixed_vector;

665 
} 
666  
667 
/**

668 
* Calculate a stability factor {teta} based on distance between

669 
* current and past isf. A value of 1 shows maximum signal stability

670 
*/

671 
static float stability_factor(const float *isf, const float *isf_past) 
672 
{ 
673 
int i;

674 
float acc = 0.0; 
675  
676 
for (i = 0; i < LP_ORDER  1; i++) 
677 
acc += (isf[i]  isf_past[i]) * (isf[i]  isf_past[i]); 
678  
679 
// XXX: This part is not so clear from the reference code

680 
// the result is more accurate changing the "/ 256" to "* 512"

681 
return FFMAX(0.0, 1.25  acc * 0.8 * 512); 
682 
} 
683  
684 
/**

685 
* Apply a nonlinear fixed gain smoothing in order to reduce

686 
* fluctuation in the energy of excitation

687 
*

688 
* @param[in] fixed_gain Unsmoothed fixed gain

689 
* @param[in,out] prev_tr_gain Previous threshold gain (updated)

690 
* @param[in] voice_fac Frame voicing factor

691 
* @param[in] stab_fac Frame stability factor

692 
*

693 
* @return The smoothed gain

694 
*/

695 
static float noise_enhancer(float fixed_gain, float *prev_tr_gain, 
696 
float voice_fac, float stab_fac) 
697 
{ 
698 
float sm_fac = 0.5 * (1  voice_fac) * stab_fac; 
699 
float g0;

700  
701 
// XXX: the following fixedpoint constants used to in(de)crement

702 
// gain by 1.5dB were taken from the reference code, maybe it could

703 
// be simpler

704 
if (fixed_gain < *prev_tr_gain) {

705 
g0 = FFMIN(*prev_tr_gain, fixed_gain + fixed_gain * 
706 
(6226 * (1.0f / (1 << 15)))); // +1.5 dB 
707 
} else

708 
g0 = FFMAX(*prev_tr_gain, fixed_gain * 
709 
(27536 * (1.0f / (1 << 15)))); // 1.5 dB 
710  
711 
*prev_tr_gain = g0; // update next frame threshold

712  
713 
return sm_fac * g0 + (1  sm_fac) * fixed_gain; 
714 
} 
715  
716 
/**

717 
* Filter the fixed_vector to emphasize the higher frequencies

718 
*

719 
* @param[in,out] fixed_vector Fixed codebook vector

720 
* @param[in] voice_fac Frame voicing factor

721 
*/

722 
static void pitch_enhancer(float *fixed_vector, float voice_fac) 
723 
{ 
724 
int i;

725 
float cpe = 0.125 * (1 + voice_fac); 
726 
float last = fixed_vector[0]; // holds c(i  1) 
727  
728 
fixed_vector[0] = cpe * fixed_vector[1]; 
729  
730 
for (i = 1; i < AMRWB_SFR_SIZE  1; i++) { 
731 
float cur = fixed_vector[i];

732  
733 
fixed_vector[i] = cpe * (last + fixed_vector[i + 1]);

734 
last = cur; 
735 
} 
736  
737 
fixed_vector[AMRWB_SFR_SIZE  1] = cpe * last;

738 
} 
739  
740 
/**

741 
* Conduct 16th order linear predictive coding synthesis from excitation

742 
*

743 
* @param[in] ctx Pointer to the AMRWBContext

744 
* @param[in] lpc Pointer to the LPC coefficients

745 
* @param[out] excitation Buffer for synthesis final excitation

746 
* @param[in] fixed_gain Fixed codebook gain for synthesis

747 
* @param[in] fixed_vector Algebraic codebook vector

748 
* @param[in,out] samples Pointer to the output samples and memory

749 
*/

750 
static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation, 
751 
float fixed_gain, const float *fixed_vector, 
752 
float *samples)

753 
{ 
754 
ff_weighted_vector_sumf(excitation, ctx>pitch_vector, fixed_vector, 
755 
ctx>pitch_gain[0], fixed_gain, AMRWB_SFR_SIZE);

756  
757 
/* emphasize pitch vector contribution in low bitrate modes */

758 
if (ctx>pitch_gain[0] > 0.5 && ctx>fr_cur_mode <= MODE_8k85) { 
759 
int i;

760 
float energy = ff_dot_productf(excitation, excitation,

761 
AMRWB_SFR_SIZE); 
762  
763 
// XXX: Weird part in both ref code and spec. A unknown parameter

764 
// {beta} seems to be identical to the current pitch gain

765 
float pitch_factor = 0.25 * ctx>pitch_gain[0] * ctx>pitch_gain[0]; 
766  
767 
for (i = 0; i < AMRWB_SFR_SIZE; i++) 
768 
excitation[i] += pitch_factor * ctx>pitch_vector[i]; 
769  
770 
ff_scale_vector_to_given_sum_of_squares(excitation, excitation, 
771 
energy, AMRWB_SFR_SIZE); 
772 
} 
773  
774 
ff_celp_lp_synthesis_filterf(samples, lpc, excitation, 
775 
AMRWB_SFR_SIZE, LP_ORDER); 
776 
} 
777  
778 
/**

779 
* Apply to synthesis a deemphasis filter of the form:

780 
* H(z) = 1 / (1  m * z^1)

781 
*

782 
* @param[out] out Output buffer

783 
* @param[in] in Input samples array with in[1]

784 
* @param[in] m Filter coefficient

785 
* @param[in,out] mem State from last filtering

786 
*/

787 
static void de_emphasis(float *out, float *in, float m, float mem[1]) 
788 
{ 
789 
int i;

790  
791 
out[0] = in[0] + m * mem[0]; 
792  
793 
for (i = 1; i < AMRWB_SFR_SIZE; i++) 
794 
out[i] = in[i] + out[i  1] * m;

795  
796 
mem[0] = out[AMRWB_SFR_SIZE  1]; 
797 
} 
798  
799 
/**

800 
* Upsample a signal by 5/4 ratio (from 12.8kHz to 16kHz) using

801 
* a FIR interpolation filter. Uses past data from before *in address

802 
*

803 
* @param[out] out Buffer for interpolated signal

804 
* @param[in] in Current signal data (length 0.8*o_size)

805 
* @param[in] o_size Output signal length

806 
*/

807 
static void upsample_5_4(float *out, const float *in, int o_size) 
808 
{ 
809 
const float *in0 = in  UPS_FIR_SIZE + 1; 
810 
int i, j, k;

811 
int int_part = 0, frac_part; 
812  
813 
i = 0;

814 
for (j = 0; j < o_size / 5; j++) { 
815 
out[i] = in[int_part]; 
816 
frac_part = 4;

817 
i++; 
818  
819 
for (k = 1; k < 5; k++) { 
820 
out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4  frac_part],

821 
UPS_MEM_SIZE); 
822 
int_part++; 
823 
frac_part; 
824 
i++; 
825 
} 
826 
} 
827 
} 
828  
829 
/**

830 
* Calculate the highband gain based on encoded index (23k85 mode) or

831 
* on the lowband speech signal and the Voice Activity Detection flag

832 
*

833 
* @param[in] ctx The context

834 
* @param[in] synth LB speech synthesis at 12.8k

835 
* @param[in] hb_idx Gain index for mode 23k85 only

836 
* @param[in] vad VAD flag for the frame

837 
*/

838 
static float find_hb_gain(AMRWBContext *ctx, const float *synth, 
839 
uint16_t hb_idx, uint8_t vad) 
840 
{ 
841 
int wsp = (vad > 0); 
842 
float tilt;

843  
844 
if (ctx>fr_cur_mode == MODE_23k85)

845 
return qua_hb_gain[hb_idx] * (1.0f / (1 << 14)); 
846  
847 
tilt = ff_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE  1) / 
848 
ff_dot_productf(synth, synth, AMRWB_SFR_SIZE); 
849  
850 
/* return gain bounded by [0.1, 1.0] */

851 
return av_clipf((1.0  FFMAX(0.0, tilt)) * (1.25  0.25 * wsp), 0.1, 1.0); 
852 
} 
853  
854 
/**

855 
* Generate the highband excitation with the same energy from the lower

856 
* one and scaled by the given gain

857 
*

858 
* @param[in] ctx The context

859 
* @param[out] hb_exc Buffer for the excitation

860 
* @param[in] synth_exc Lowband excitation used for synthesis

861 
* @param[in] hb_gain Wanted excitation gain

862 
*/

863 
static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc, 
864 
const float *synth_exc, float hb_gain) 
865 
{ 
866 
int i;

867 
float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);

868  
869 
/* Generate a whitenoise excitation */

870 
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) 
871 
hb_exc[i] = 32768.0  (uint16_t) av_lfg_get(&ctx>prng); 
872  
873 
ff_scale_vector_to_given_sum_of_squares(hb_exc, hb_exc, 
874 
energy * hb_gain * hb_gain, 
875 
AMRWB_SFR_SIZE_16k); 
876 
} 
877  
878 
/**

879 
* Calculate the autocorrelation for the ISF difference vector

880 
*/

881 
static float auto_correlation(float *diff_isf, float mean, int lag) 
882 
{ 
883 
int i;

884 
float sum = 0.0; 
885  
886 
for (i = 7; i < LP_ORDER  2; i++) { 
887 
float prod = (diff_isf[i]  mean) * (diff_isf[i  lag]  mean);

888 
sum += prod * prod; 
889 
} 
890 
return sum;

891 
} 
892  
893 
/**

894 
* Extrapolate a ISF vector to the 16kHz range (20th order LP)

895 
* used at mode 6k60 LP filter for the high frequency band

896 
*

897 
* @param[out] out Buffer for extrapolated isf

898 
* @param[in] isf Input isf vector

899 
*/

900 
static void extrapolate_isf(float out[LP_ORDER_16k], float isf[LP_ORDER]) 
901 
{ 
902 
float diff_isf[LP_ORDER  2], diff_mean; 
903 
float *diff_hi = diff_isf  LP_ORDER + 1; // diff array for extrapolated indexes 
904 
float corr_lag[3]; 
905 
float est, scale;

906 
int i, i_max_corr;

907  
908 
memcpy(out, isf, (LP_ORDER  1) * sizeof(float)); 
909 
out[LP_ORDER_16k  1] = isf[LP_ORDER  1]; 
910  
911 
/* Calculate the difference vector */

912 
for (i = 0; i < LP_ORDER  2; i++) 
913 
diff_isf[i] = isf[i + 1]  isf[i];

914  
915 
diff_mean = 0.0; 
916 
for (i = 2; i < LP_ORDER  2; i++) 
917 
diff_mean += diff_isf[i] * (1.0f / (LP_ORDER  4)); 
918  
919 
/* Find which is the maximum autocorrelation */

920 
i_max_corr = 0;

921 
for (i = 0; i < 3; i++) { 
922 
corr_lag[i] = auto_correlation(diff_isf, diff_mean, i + 2);

923  
924 
if (corr_lag[i] > corr_lag[i_max_corr])

925 
i_max_corr = i; 
926 
} 
927 
i_max_corr++; 
928  
929 
for (i = LP_ORDER  1; i < LP_ORDER_16k  1; i++) 
930 
out[i] = isf[i  1] + isf[i  1  i_max_corr] 
931 
 isf[i  2  i_max_corr];

932  
933 
/* Calculate an estimate for ISF(18) and scale ISF based on the error */

934 
est = 7965 + (out[2]  out[3]  out[4]) / 6.0; 
935 
scale = 0.5 * (FFMIN(est, 7600)  out[LP_ORDER  2]) / 
936 
(out[LP_ORDER_16k  2]  out[LP_ORDER  2]); 
937  
938 
for (i = LP_ORDER  1; i < LP_ORDER_16k  1; i++) 
939 
diff_hi[i] = scale * (out[i]  out[i  1]);

940  
941 
/* Stability insurance */

942 
for (i = LP_ORDER; i < LP_ORDER_16k  1; i++) 
943 
if (diff_hi[i] + diff_hi[i  1] < 5.0) { 
944 
if (diff_hi[i] > diff_hi[i  1]) { 
945 
diff_hi[i  1] = 5.0  diff_hi[i]; 
946 
} else

947 
diff_hi[i] = 5.0  diff_hi[i  1]; 
948 
} 
949  
950 
for (i = LP_ORDER  1; i < LP_ORDER_16k  1; i++) 
951 
out[i] = out[i  1] + diff_hi[i] * (1.0f / (1 << 15)); 
952  
953 
/* Scale the ISF vector for 16000 Hz */

954 
for (i = 0; i < LP_ORDER_16k  1; i++) 
955 
out[i] *= 0.8; 
956 
} 
957  
958 
/**

959 
* Spectral expand the LP coefficients using the equation:

960 
* y[i] = x[i] * (gamma ** i)

961 
*

962 
* @param[out] out Output buffer (may use input array)

963 
* @param[in] lpc LP coefficients array

964 
* @param[in] gamma Weighting factor

965 
* @param[in] size LP array size

966 
*/

967 
static void lpc_weighting(float *out, const float *lpc, float gamma, int size) 
968 
{ 
969 
int i;

970 
float fac = gamma;

971  
972 
for (i = 0; i < size; i++) { 
973 
out[i] = lpc[i] * fac; 
974 
fac *= gamma; 
975 
} 
976 
} 
977  
978 
/**

979 
* Conduct 20th order linear predictive coding synthesis for the high

980 
* frequency band excitation at 16kHz

981 
*

982 
* @param[in] ctx The context

983 
* @param[in] subframe Current subframe index (0 to 3)

984 
* @param[in,out] samples Pointer to the output speech samples

985 
* @param[in] exc Generated whitenoise scaled excitation

986 
* @param[in] isf Current frame isf vector

987 
* @param[in] isf_past Past frame final isf vector

988 
*/

989 
static void hb_synthesis(AMRWBContext *ctx, int subframe, float *samples, 
990 
const float *exc, const float *isf, const float *isf_past) 
991 
{ 
992 
float hb_lpc[LP_ORDER_16k];

993 
enum Mode mode = ctx>fr_cur_mode;

994  
995 
if (mode == MODE_6k60) {

996 
float e_isf[LP_ORDER_16k]; // ISF vector for extrapolation 
997 
double e_isp[LP_ORDER_16k];

998  
999 
ff_weighted_vector_sumf(e_isf, isf_past, isf, isfp_inter[subframe], 
1000 
1.0  isfp_inter[subframe], LP_ORDER); 
1001  
1002 
extrapolate_isf(e_isf, e_isf); 
1003  
1004 
e_isf[LP_ORDER_16k  1] *= 2.0; 
1005 
ff_acelp_lsf2lspd(e_isp, e_isf, LP_ORDER_16k); 
1006 
ff_amrwb_lsp2lpc(e_isp, hb_lpc, LP_ORDER_16k); 
1007  
1008 
lpc_weighting(hb_lpc, hb_lpc, 0.9, LP_ORDER_16k); 
1009 
} else {

1010 
lpc_weighting(hb_lpc, ctx>lp_coef[subframe], 0.6, LP_ORDER); 
1011 
} 
1012  
1013 
ff_celp_lp_synthesis_filterf(samples, hb_lpc, exc, AMRWB_SFR_SIZE_16k, 
1014 
(mode == MODE_6k60) ? LP_ORDER_16k : LP_ORDER); 
1015 
} 
1016  
1017 
/**

1018 
* Apply to highband samples a 15th order filter

1019 
* The filter characteristic depends on the given coefficients

1020 
*

1021 
* @param[out] out Buffer for filtered output

1022 
* @param[in] fir_coef Filter coefficients

1023 
* @param[in,out] mem State from last filtering (updated)

1024 
* @param[in] in Input speech data (highband)

1025 
*

1026 
* @remark It is safe to pass the same array in in and out parameters

1027 
*/

1028 
static void hb_fir_filter(float *out, const float fir_coef[HB_FIR_SIZE + 1], 
1029 
float mem[HB_FIR_SIZE], const float *in) 
1030 
{ 
1031 
int i, j;

1032 
float data[AMRWB_SFR_SIZE_16k + HB_FIR_SIZE]; // past and current samples 
1033  
1034 
memcpy(data, mem, HB_FIR_SIZE * sizeof(float)); 
1035 
memcpy(data + HB_FIR_SIZE, in, AMRWB_SFR_SIZE_16k * sizeof(float)); 
1036  
1037 
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) { 
1038 
out[i] = 0.0; 
1039 
for (j = 0; j <= HB_FIR_SIZE; j++) 
1040 
out[i] += data[i + j] * fir_coef[j]; 
1041 
} 
1042  
1043 
memcpy(mem, data + AMRWB_SFR_SIZE_16k, HB_FIR_SIZE * sizeof(float)); 
1044 
} 
1045  
1046 
/**

1047 
* Update context state before the next subframe

1048 
*/

1049 
static void update_sub_state(AMRWBContext *ctx) 
1050 
{ 
1051 
memmove(&ctx>excitation_buf[0], &ctx>excitation_buf[AMRWB_SFR_SIZE],

1052 
(AMRWB_P_DELAY_MAX + LP_ORDER + 1) * sizeof(float)); 
1053  
1054 
memmove(&ctx>pitch_gain[1], &ctx>pitch_gain[0], 5 * sizeof(float)); 
1055 
memmove(&ctx>fixed_gain[1], &ctx>fixed_gain[0], 1 * sizeof(float)); 
1056  
1057 
memmove(&ctx>samples_az[0], &ctx>samples_az[AMRWB_SFR_SIZE],

1058 
LP_ORDER * sizeof(float)); 
1059 
memmove(&ctx>samples_up[0], &ctx>samples_up[AMRWB_SFR_SIZE],

1060 
UPS_MEM_SIZE * sizeof(float)); 
1061 
memmove(&ctx>samples_hb[0], &ctx>samples_hb[AMRWB_SFR_SIZE_16k],

1062 
LP_ORDER_16k * sizeof(float)); 
1063 
} 
1064  
1065 
static int amrwb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, 
1066 
AVPacket *avpkt) 
1067 
{ 
1068 
AMRWBContext *ctx = avctx>priv_data; 
1069 
AMRWBFrame *cf = &ctx>frame; 
1070 
const uint8_t *buf = avpkt>data;

1071 
int buf_size = avpkt>size;

1072 
int expected_fr_size, header_size;

1073 
float *buf_out = data;

1074 
float spare_vector[AMRWB_SFR_SIZE]; // extra stack space to hold result from antisparseness processing 
1075 
float fixed_gain_factor; // fixed gain correction factor (gamma) 
1076 
float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use 
1077 
float synth_fixed_gain; // the fixed gain that synthesis should use 
1078 
float voice_fac, stab_fac; // parameters used for gain smoothing 
1079 
float synth_exc[AMRWB_SFR_SIZE]; // postprocessed excitation for synthesis 
1080 
float hb_exc[AMRWB_SFR_SIZE_16k]; // excitation for the high frequency band 
1081 
float hb_samples[AMRWB_SFR_SIZE_16k]; // filtered highband samples from synthesis 
1082 
float hb_gain;

1083 
int sub, i;

1084  
1085 
header_size = decode_mime_header(ctx, buf); 
1086 
expected_fr_size = ((cf_sizes_wb[ctx>fr_cur_mode] + 7) >> 3) + 1; 
1087  
1088 
if (buf_size < expected_fr_size) {

1089 
av_log(avctx, AV_LOG_ERROR, 
1090 
"Frame too small (%d bytes). Truncated file?\n", buf_size);

1091 
*data_size = 0;

1092 
return buf_size;

1093 
} 
1094  
1095 
if (!ctx>fr_quality  ctx>fr_cur_mode > MODE_SID)

1096 
av_log(avctx, AV_LOG_ERROR, "Encountered a bad or corrupted frame\n");

1097  
1098 
if (ctx>fr_cur_mode == MODE_SID) /* Comfort noise frame */ 
1099 
av_log_missing_feature(avctx, "SID mode", 1); 
1100  
1101 
if (ctx>fr_cur_mode >= MODE_SID)

1102 
return 1; 
1103  
1104 
ff_amr_bit_reorder((uint16_t *) &ctx>frame, sizeof(AMRWBFrame),

1105 
buf + header_size, amr_bit_orderings_by_mode[ctx>fr_cur_mode]); 
1106  
1107 
/* Decode the quantized ISF vector */

1108 
if (ctx>fr_cur_mode == MODE_6k60) {

1109 
decode_isf_indices_36b(cf>isp_id, ctx>isf_cur); 
1110 
} else {

1111 
decode_isf_indices_46b(cf>isp_id, ctx>isf_cur); 
1112 
} 
1113  
1114 
isf_add_mean_and_past(ctx>isf_cur, ctx>isf_q_past); 
1115 
ff_set_min_dist_lsf(ctx>isf_cur, MIN_ISF_SPACING, LP_ORDER  1);

1116  
1117 
stab_fac = stability_factor(ctx>isf_cur, ctx>isf_past_final); 
1118  
1119 
ctx>isf_cur[LP_ORDER  1] *= 2.0; 
1120 
ff_acelp_lsf2lspd(ctx>isp[3], ctx>isf_cur, LP_ORDER);

1121  
1122 
/* Generate a ISP vector for each subframe */

1123 
if (ctx>first_frame) {

1124 
ctx>first_frame = 0;

1125 
memcpy(ctx>isp_sub4_past, ctx>isp[3], LP_ORDER * sizeof(double)); 
1126 
} 
1127 
interpolate_isp(ctx>isp, ctx>isp_sub4_past); 
1128  
1129 
for (sub = 0; sub < 4; sub++) 
1130 
ff_amrwb_lsp2lpc(ctx>isp[sub], ctx>lp_coef[sub], LP_ORDER); 
1131  
1132 
for (sub = 0; sub < 4; sub++) { 
1133 
const AMRWBSubFrame *cur_subframe = &cf>subframe[sub];

1134 
float *sub_buf = buf_out + sub * AMRWB_SFR_SIZE_16k;

1135  
1136 
/* Decode adaptive codebook (pitch vector) */

1137 
decode_pitch_vector(ctx, cur_subframe, sub); 
1138 
/* Decode innovative codebook (fixed vector) */

1139 
decode_fixed_vector(ctx>fixed_vector, cur_subframe>pul_ih, 
1140 
cur_subframe>pul_il, ctx>fr_cur_mode); 
1141  
1142 
pitch_sharpening(ctx, ctx>fixed_vector); 
1143  
1144 
decode_gains(cur_subframe>vq_gain, ctx>fr_cur_mode, 
1145 
&fixed_gain_factor, &ctx>pitch_gain[0]);

1146  
1147 
ctx>fixed_gain[0] =

1148 
ff_amr_set_fixed_gain(fixed_gain_factor, 
1149 
ff_dot_productf(ctx>fixed_vector, ctx>fixed_vector, 
1150 
AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE, 
1151 
ctx>prediction_error, 
1152 
ENERGY_MEAN, energy_pred_fac); 
1153  
1154 
/* Calculate voice factor and store tilt for next subframe */

1155 
voice_fac = voice_factor(ctx>pitch_vector, ctx>pitch_gain[0],

1156 
ctx>fixed_vector, ctx>fixed_gain[0]);

1157 
ctx>tilt_coef = voice_fac * 0.25 + 0.25; 
1158  
1159 
/* Construct current excitation */

1160 
for (i = 0; i < AMRWB_SFR_SIZE; i++) { 
1161 
ctx>excitation[i] *= ctx>pitch_gain[0];

1162 
ctx>excitation[i] += ctx>fixed_gain[0] * ctx>fixed_vector[i];

1163 
ctx>excitation[i] = truncf(ctx>excitation[i]); 
1164 
} 
1165  
1166 
/* Postprocessing of excitation elements */

1167 
synth_fixed_gain = noise_enhancer(ctx>fixed_gain[0], &ctx>prev_tr_gain,

1168 
voice_fac, stab_fac); 
1169  
1170 
synth_fixed_vector = anti_sparseness(ctx, ctx>fixed_vector, 
1171 
spare_vector); 
1172  
1173 
pitch_enhancer(synth_fixed_vector, voice_fac); 
1174  
1175 
synthesis(ctx, ctx>lp_coef[sub], synth_exc, synth_fixed_gain, 
1176 
synth_fixed_vector, &ctx>samples_az[LP_ORDER]); 
1177  
1178 
/* Synthesis speech postprocessing */

1179 
de_emphasis(&ctx>samples_up[UPS_MEM_SIZE], 
1180 
&ctx>samples_az[LP_ORDER], PREEMPH_FAC, ctx>demph_mem); 
1181  
1182 
ff_acelp_apply_order_2_transfer_function(&ctx>samples_up[UPS_MEM_SIZE], 
1183 
&ctx>samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_31_poles, 
1184 
hpf_31_gain, ctx>hpf_31_mem, AMRWB_SFR_SIZE); 
1185  
1186 
upsample_5_4(sub_buf, &ctx>samples_up[UPS_FIR_SIZE], 
1187 
AMRWB_SFR_SIZE_16k); 
1188  
1189 
/* High frequency band (6.4  7.0 kHz) generation part */

1190 
ff_acelp_apply_order_2_transfer_function(hb_samples, 
1191 
&ctx>samples_up[UPS_MEM_SIZE], hpf_zeros, hpf_400_poles, 
1192 
hpf_400_gain, ctx>hpf_400_mem, AMRWB_SFR_SIZE); 
1193  
1194 
hb_gain = find_hb_gain(ctx, hb_samples, 
1195 
cur_subframe>hb_gain, cf>vad); 
1196  
1197 
scaled_hb_excitation(ctx, hb_exc, synth_exc, hb_gain); 
1198  
1199 
hb_synthesis(ctx, sub, &ctx>samples_hb[LP_ORDER_16k], 
1200 
hb_exc, ctx>isf_cur, ctx>isf_past_final); 
1201  
1202 
/* Highband postprocessing filters */

1203 
hb_fir_filter(hb_samples, bpf_6_7_coef, ctx>bpf_6_7_mem, 
1204 
&ctx>samples_hb[LP_ORDER_16k]); 
1205  
1206 
if (ctx>fr_cur_mode == MODE_23k85)

1207 
hb_fir_filter(hb_samples, lpf_7_coef, ctx>lpf_7_mem, 
1208 
hb_samples); 
1209  
1210 
/* Add the low and high frequency bands */

1211 
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++) 
1212 
sub_buf[i] = (sub_buf[i] + hb_samples[i]) * (1.0f / (1 << 15)); 
1213  
1214 
/* Update buffers and history */

1215 
update_sub_state(ctx); 
1216 
} 
1217  
1218 
/* update state for next frame */

1219 
memcpy(ctx>isp_sub4_past, ctx>isp[3], LP_ORDER * sizeof(ctx>isp[3][0])); 
1220 
memcpy(ctx>isf_past_final, ctx>isf_cur, LP_ORDER * sizeof(float)); 
1221  
1222 
/* report how many samples we got */

1223 
*data_size = 4 * AMRWB_SFR_SIZE_16k * sizeof(float); 
1224  
1225 
return expected_fr_size;

1226 
} 
1227  
1228 
AVCodec ff_amrwb_decoder = { 
1229 
.name = "amrwb",

1230 
.type = AVMEDIA_TYPE_AUDIO, 
1231 
.id = CODEC_ID_AMR_WB, 
1232 
.priv_data_size = sizeof(AMRWBContext),

1233 
.init = amrwb_decode_init, 
1234 
.decode = amrwb_decode_frame, 
1235 
.long_name = NULL_IF_CONFIG_SMALL("Adaptive MultiRate WideBand"),

1236 
.sample_fmts = (enum AVSampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},

1237 
}; 