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/*
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 * AMR narrowband decoder
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 * Copyright (c) 2006-2007 Robert Swain
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 * Copyright (c) 2009 Colin McQuillan
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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 02110-1301 USA
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 */
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/**
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 * @file libavcodec/amrnbdec.c
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 * AMR narrowband decoder
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 *
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 * This decoder uses floats for simplicity and so is not bit-exact. One
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 * difference is that differences in phase can accumulate. The test sequences
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 * in 3GPP TS 26.074 can still be useful.
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 *
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 * - Comparing this file's output to the output of the ref decoder gives a
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 *   PSNR of 30 to 80. Plotting the output samples shows a difference in
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 *   phase in some areas.
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 *
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 * - Comparing both decoders against their input, this decoder gives a similar
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 *   PSNR. If the test sequence homing frames are removed (this decoder does
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 *   not detect them), the PSNR is at least as good as the reference on 140
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 *   out of 169 tests.
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 */
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#include <string.h>
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#include <math.h>
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#include "avcodec.h"
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#include "get_bits.h"
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#include "libavutil/common.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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#include "lsp.h"
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#include "amrnbdata.h"
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#define AMR_BLOCK_SIZE              160   ///< samples per frame
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#define AMR_SAMPLE_BOUND        32768.0   ///< threshold for synthesis overflow
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/**
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 * Scale from constructed speech to [-1,1]
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 *
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 * AMR is designed to produce 16-bit PCM samples (3GPP TS 26.090 4.2) but
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 * upscales by two (section 6.2.2).
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 *
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 * Fundamentally, this scale is determined by energy_mean through
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 * the fixed vector contribution to the excitation vector.
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 */
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#define AMR_SAMPLE_SCALE  (2.0 / 32768.0)
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/** Prediction factor for 12.2kbit/s mode */
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#define PRED_FAC_MODE_12k2             0.65
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#define LSF_R_FAC          (8000.0 / 32768.0) ///< LSF residual tables to Hertz
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#define MIN_LSF_SPACING    (50.0488 / 8000.0) ///< Ensures stability of LPC filter
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#define PITCH_LAG_MIN_MODE_12k2          18   ///< Lower bound on decoded lag search in 12.2kbit/s mode
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/** Initial energy in dB. Also used for bad frames (unimplemented). */
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#define MIN_ENERGY -14.0
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/** Maximum sharpening factor
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 *
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 * The specification says 0.8, which should be 13107, but the reference C code
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 * uses 13017 instead. (Amusingly the same applies to SHARP_MAX in g729dec.c.)
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 */
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#define SHARP_MAX 0.79449462890625
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/** Number of impulse response coefficients used for tilt factor */
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#define AMR_TILT_RESPONSE   22
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/** Tilt factor = 1st reflection coefficient * gamma_t */
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#define AMR_TILT_GAMMA_T   0.8
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/** Adaptive gain control factor used in post-filter */
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#define AMR_AGC_ALPHA      0.9
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typedef struct AMRContext {
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    AMRNBFrame                        frame; ///< decoded AMR parameters (lsf coefficients, codebook indexes, etc)
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    uint8_t             bad_frame_indicator; ///< bad frame ? 1 : 0
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    enum Mode                cur_frame_mode;
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    int16_t     prev_lsf_r[LP_FILTER_ORDER]; ///< residual LSF vector from previous subframe
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    double          lsp[4][LP_FILTER_ORDER]; ///< lsp vectors from current frame
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    double   prev_lsp_sub4[LP_FILTER_ORDER]; ///< lsp vector for the 4th subframe of the previous frame
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    float         lsf_q[4][LP_FILTER_ORDER]; ///< Interpolated LSF vector for fixed gain smoothing
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    float          lsf_avg[LP_FILTER_ORDER]; ///< vector of averaged lsf vector
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    float           lpc[4][LP_FILTER_ORDER]; ///< lpc coefficient vectors for 4 subframes
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    uint8_t                   pitch_lag_int; ///< integer part of pitch lag from current subframe
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    float excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1 + AMR_SUBFRAME_SIZE]; ///< current excitation and all necessary excitation history
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    float                       *excitation; ///< pointer to the current excitation vector in excitation_buf
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    float   pitch_vector[AMR_SUBFRAME_SIZE]; ///< adaptive code book (pitch) vector
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    float   fixed_vector[AMR_SUBFRAME_SIZE]; ///< algebraic codebook (fixed) vector (must be kept zero between frames)
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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[5]; ///< quantified pitch gains for the current and previous four subframes
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    float                     fixed_gain[5]; ///< quantified fixed gains for the current and previous four subframes
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    float                              beta; ///< previous pitch_gain, bounded by [0.0,SHARP_MAX]
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    uint8_t                      diff_count; ///< the number of subframes for which diff has been above 0.65
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    uint8_t                      hang_count; ///< the number of subframes since a hangover period started
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    float            prev_sparse_fixed_gain; ///< previous fixed gain; used by anti-sparseness processing 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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    uint8_t                 ir_filter_onset; ///< flag for impulse response filter strength
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    float                postfilter_mem[10]; ///< previous intermediate values in the formant filter
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    float                          tilt_mem; ///< previous input to tilt compensation filter
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    float                    postfilter_agc; ///< previous factor used for adaptive gain control
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    float                  high_pass_mem[2]; ///< previous intermediate values in the high-pass filter
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    float samples_in[LP_FILTER_ORDER + AMR_SUBFRAME_SIZE]; ///< floating point samples
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} AMRContext;
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/** Double version of ff_weighted_vector_sumf() */
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static void weighted_vector_sumd(double *out, const double *in_a,
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                                 const double *in_b, double weight_coeff_a,
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                                 double weight_coeff_b, int length)
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{
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    int i;
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    for (i = 0; i < length; i++)
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        out[i] = weight_coeff_a * in_a[i]
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               + weight_coeff_b * in_b[i];
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}
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static av_cold int amrnb_decode_init(AVCodecContext *avctx)
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{
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    AMRContext *p = avctx->priv_data;
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    int i;
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    avctx->sample_fmt = SAMPLE_FMT_FLT;
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    // p->excitation always points to the same position in p->excitation_buf
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    p->excitation = &p->excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];
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    for (i = 0; i < LP_FILTER_ORDER; i++) {
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        p->prev_lsp_sub4[i] =    lsp_sub4_init[i] * 1000 / (float)(1 << 15);
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        p->lsf_avg[i] = p->lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15);
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    }
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    for (i = 0; i < 4; i++)
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        p->prediction_error[i] = MIN_ENERGY;
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    return 0;
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}
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/**
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 * Unpack an RFC4867 speech frame into the AMR frame mode and parameters.
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 *
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 * The order of speech bits is specified by 3GPP TS 26.101.
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 *
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 * @param p the context
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 * @param buf               pointer to the input buffer
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 * @param buf_size          size of the input buffer
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 *
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 * @return the frame mode
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 */
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static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf,
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                                  int buf_size)
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{
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    GetBitContext gb;
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    enum Mode mode;
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    init_get_bits(&gb, buf, buf_size * 8);
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    // Decode the first octet.
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    skip_bits(&gb, 1);                        // padding bit
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    mode = get_bits(&gb, 4);                  // frame type
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    p->bad_frame_indicator = !get_bits1(&gb); // quality bit
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    skip_bits(&gb, 2);                        // two padding bits
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    if (mode <= MODE_DTX) {
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        uint16_t *data = (uint16_t *)&p->frame;
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        const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];
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        int field_size;
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        memset(&p->frame, 0, sizeof(AMRNBFrame));
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        buf++;
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        while ((field_size = *order++)) {
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            int field = 0;
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            int field_offset = *order++;
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            while (field_size--) {
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               int bit = *order++;
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               field <<= 1;
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               field |= buf[bit >> 3] >> (bit & 7) & 1;
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            }
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            data[field_offset] = field;
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        }
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    }
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    return mode;
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}
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/// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions
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/// @{
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/**
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 * Convert an lsf vector into an lsp vector.
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 *
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 * @param lsf               input lsf vector
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 * @param lsp               output lsp vector
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 */
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static void lsf2lsp(const float *lsf, double *lsp)
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{
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    int i;
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    for (i = 0; i < LP_FILTER_ORDER; i++)
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        lsp[i] = cos(2.0 * M_PI * lsf[i]);
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}
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/**
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 * Interpolate the LSF vector (used for fixed gain smoothing).
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 * The interpolation is done over all four subframes even in MODE_12k2.
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 *
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 * @param[in,out] lsf_q     LSFs in [0,1] for each subframe
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 * @param[in]     lsf_new   New LSFs in [0,1] for subframe 4
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 */
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static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new)
246
{
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    int i;
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    for (i = 0; i < 4; i++)
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        ff_weighted_vector_sumf(lsf_q[i], lsf_q[3], lsf_new,
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                                0.25 * (3 - i), 0.25 * (i + 1),
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                                LP_FILTER_ORDER);
253
}
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/**
256
 * Decode a set of 5 split-matrix quantized lsf indexes into an lsp vector.
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 *
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 * @param p the context
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 * @param lsp output LSP vector
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 * @param lsf_no_r LSF vector without the residual vector added
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 * @param lsf_quantizer pointers to LSF dictionary tables
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 * @param quantizer_offset offset in tables
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 * @param sign for the 3 dictionary table
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 * @param update store data for computing the next frame's LSFs
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 */
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static void lsf2lsp_for_mode12k2(AMRContext *p, double lsp[LP_FILTER_ORDER],
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                                 const float lsf_no_r[LP_FILTER_ORDER],
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                                 const int16_t *lsf_quantizer[5],
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                                 const int quantizer_offset,
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                                 const int sign, const int update)
271
{
272
    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
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    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
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    int i;
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    for (i = 0; i < LP_FILTER_ORDER >> 1; i++)
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        memcpy(&lsf_r[i << 1], &lsf_quantizer[i][quantizer_offset],
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               2 * sizeof(*lsf_r));
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280
    if (sign) {
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        lsf_r[4] *= -1;
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        lsf_r[5] *= -1;
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    }
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    if (update)
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        memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(float));
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    for (i = 0; i < LP_FILTER_ORDER; i++)
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        lsf_q[i] = lsf_r[i] * (LSF_R_FAC / 8000.0) + lsf_no_r[i] * (1.0 / 8000.0);
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    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
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    if (update)
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        interpolate_lsf(p->lsf_q, lsf_q);
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    lsf2lsp(lsf_q, lsp);
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}
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/**
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 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
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 *
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 * @param p                 pointer to the AMRContext
303
 */
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static void lsf2lsp_5(AMRContext *p)
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{
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    const uint16_t *lsf_param = p->frame.lsf;
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    float lsf_no_r[LP_FILTER_ORDER]; // LSFs without the residual vector
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    const int16_t *lsf_quantizer[5];
309
    int i;
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    lsf_quantizer[0] = lsf_5_1[lsf_param[0]];
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    lsf_quantizer[1] = lsf_5_2[lsf_param[1]];
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    lsf_quantizer[2] = lsf_5_3[lsf_param[2] >> 1];
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    lsf_quantizer[3] = lsf_5_4[lsf_param[3]];
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    lsf_quantizer[4] = lsf_5_5[lsf_param[4]];
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    for (i = 0; i < LP_FILTER_ORDER; i++)
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        lsf_no_r[i] = p->prev_lsf_r[i] * LSF_R_FAC * PRED_FAC_MODE_12k2 + lsf_5_mean[i];
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    lsf2lsp_for_mode12k2(p, p->lsp[1], lsf_no_r, lsf_quantizer, 0, lsf_param[2] & 1, 0);
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    lsf2lsp_for_mode12k2(p, p->lsp[3], lsf_no_r, lsf_quantizer, 2, lsf_param[2] & 1, 1);
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    // interpolate LSP vectors at subframes 1 and 3
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    weighted_vector_sumd(p->lsp[0], p->prev_lsp_sub4, p->lsp[1], 0.5, 0.5, LP_FILTER_ORDER);
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    weighted_vector_sumd(p->lsp[2], p->lsp[1]       , p->lsp[3], 0.5, 0.5, LP_FILTER_ORDER);
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}
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/**
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 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
330
 *
331
 * @param p                 pointer to the AMRContext
332
 */
333
static void lsf2lsp_3(AMRContext *p)
334
{
335
    const uint16_t *lsf_param = p->frame.lsf;
336
    int16_t lsf_r[LP_FILTER_ORDER]; // residual LSF vector
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    float lsf_q[LP_FILTER_ORDER]; // quantified LSF vector
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    const int16_t *lsf_quantizer;
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    int i, j;
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    lsf_quantizer = (p->cur_frame_mode == MODE_7k95 ? lsf_3_1_MODE_7k95 : lsf_3_1)[lsf_param[0]];
342
    memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
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344
    lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
345
    memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
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347
    lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
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    memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
349

    
350
    // calculate mean-removed LSF vector and add mean
351
    for (i = 0; i < LP_FILTER_ORDER; i++)
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        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);
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    ff_set_min_dist_lsf(lsf_q, MIN_LSF_SPACING, LP_FILTER_ORDER);
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356
    // store data for computing the next frame's LSFs
357
    interpolate_lsf(p->lsf_q, lsf_q);
358
    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
359

    
360
    lsf2lsp(lsf_q, p->lsp[3]);
361

    
362
    // interpolate LSP vectors at subframes 1, 2 and 3
363
    for (i = 1; i <= 3; i++)
364
        for(j = 0; j < LP_FILTER_ORDER; j++)
365
            p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
366
                (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
367
}
368

    
369
/// @}
370

    
371

    
372
/// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
373
/// @{
374

    
375
/**
376
 * Like ff_decode_pitch_lag(), but with 1/6 resolution
377
 */
378
static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
379
                                 const int prev_lag_int, const int subframe)
380
{
381
    if (subframe == 0 || subframe == 2) {
382
        if (pitch_index < 463) {
383
            *lag_int  = (pitch_index + 107) * 10923 >> 16;
384
            *lag_frac = pitch_index - *lag_int * 6 + 105;
385
        } else {
386
            *lag_int  = pitch_index - 368;
387
            *lag_frac = 0;
388
        }
389
    } else {
390
        *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
391
        *lag_frac = pitch_index - *lag_int * 6 - 3;
392
        *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
393
                            PITCH_DELAY_MAX - 9);
394
    }
395
}
396

    
397
static void decode_pitch_vector(AMRContext *p,
398
                                const AMRNBSubframe *amr_subframe,
399
                                const int subframe)
400
{
401
    int pitch_lag_int, pitch_lag_frac;
402
    enum Mode mode = p->cur_frame_mode;
403

    
404
    if (p->cur_frame_mode == MODE_12k2) {
405
        decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
406
                             amr_subframe->p_lag, p->pitch_lag_int,
407
                             subframe);
408
    } else
409
        ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
410
                            amr_subframe->p_lag,
411
                            p->pitch_lag_int, subframe,
412
                            mode != MODE_4k75 && mode != MODE_5k15,
413
                            mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
414

    
415
    p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
416

    
417
    pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
418

    
419
    pitch_lag_int += pitch_lag_frac > 0;
420

    
421
    /* Calculate the pitch vector by interpolating the past excitation at the
422
       pitch lag using a b60 hamming windowed sinc function.   */
423
    ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
424
                          ff_b60_sinc, 6,
425
                          pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
426
                          10, AMR_SUBFRAME_SIZE);
427

    
428
    memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
429
}
430

    
431
/// @}
432

    
433

    
434
/// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
435
/// @{
436

    
437
/**
438
 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
439
 */
440
static void decode_10bit_pulse(int code, int pulse_position[8],
441
                               int i1, int i2, int i3)
442
{
443
    // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
444
    // the 3 pulses and the upper 7 bits being coded in base 5
445
    const uint8_t *positions = base_five_table[code >> 3];
446
    pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
447
    pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
448
    pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
449
}
450

    
451
/**
452
 * Decode the algebraic codebook index to pulse positions and signs and
453
 * construct the algebraic codebook vector for MODE_10k2.
454
 *
455
 * @param fixed_index          positions of the eight pulses
456
 * @param fixed_sparse         pointer to the algebraic codebook vector
457
 */
458
static void decode_8_pulses_31bits(const int16_t *fixed_index,
459
                                   AMRFixed *fixed_sparse)
460
{
461
    int pulse_position[8];
462
    int i, temp;
463

    
464
    decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
465
    decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
466

    
467
    // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
468
    // the 2 pulses and the upper 5 bits being coded in base 5
469
    temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
470
    pulse_position[3] = temp % 5;
471
    pulse_position[7] = temp / 5;
472
    if (pulse_position[7] & 1)
473
        pulse_position[3] = 4 - pulse_position[3];
474
    pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
475
    pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
476

    
477
    fixed_sparse->n = 8;
478
    for (i = 0; i < 4; i++) {
479
        const int pos1   = (pulse_position[i]     << 2) + i;
480
        const int pos2   = (pulse_position[i + 4] << 2) + i;
481
        const float sign = fixed_index[i] ? -1.0 : 1.0;
482
        fixed_sparse->x[i    ] = pos1;
483
        fixed_sparse->x[i + 4] = pos2;
484
        fixed_sparse->y[i    ] = sign;
485
        fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
486
    }
487
}
488

    
489
/**
490
 * Decode the algebraic codebook index to pulse positions and signs,
491
 * then construct the algebraic codebook vector.
492
 *
493
 *                              nb of pulses | bits encoding pulses
494
 * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
495
 *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
496
 *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
497
 *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
498
 *
499
 * @param fixed_sparse pointer to the algebraic codebook vector
500
 * @param pulses       algebraic codebook indexes
501
 * @param mode         mode of the current frame
502
 * @param subframe     current subframe number
503
 */
504
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
505
                                const enum Mode mode, const int subframe)
506
{
507
    assert(MODE_4k75 <= mode && mode <= MODE_12k2);
508

    
509
    if (mode == MODE_12k2) {
510
        ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
511
    } else if (mode == MODE_10k2) {
512
        decode_8_pulses_31bits(pulses, fixed_sparse);
513
    } else {
514
        int *pulse_position = fixed_sparse->x;
515
        int i, pulse_subset;
516
        const int fixed_index = pulses[0];
517

    
518
        if (mode <= MODE_5k15) {
519
            pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
520
            pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
521
            pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
522
            fixed_sparse->n = 2;
523
        } else if (mode == MODE_5k9) {
524
            pulse_subset      = ((fixed_index & 1) << 1) + 1;
525
            pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
526
            pulse_subset      = (fixed_index  >> 4) & 3;
527
            pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
528
            fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
529
        } else if (mode == MODE_6k7) {
530
            pulse_position[0] = (fixed_index        & 7) * 5;
531
            pulse_subset      = (fixed_index  >> 2) & 2;
532
            pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
533
            pulse_subset      = (fixed_index  >> 6) & 2;
534
            pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
535
            fixed_sparse->n = 3;
536
        } else { // mode <= MODE_7k95
537
            pulse_position[0] = gray_decode[ fixed_index        & 7];
538
            pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
539
            pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
540
            pulse_subset      = (fixed_index >> 9) & 1;
541
            pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
542
            fixed_sparse->n = 4;
543
        }
544
        for (i = 0; i < fixed_sparse->n; i++)
545
            fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
546
    }
547
}
548

    
549
/**
550
 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
551
 *
552
 * @param p the context
553
 * @param subframe unpacked amr subframe
554
 * @param mode mode of the current frame
555
 * @param fixed_sparse sparse respresentation of the fixed vector
556
 */
557
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
558
                             AMRFixed *fixed_sparse)
559
{
560
    // The spec suggests the current pitch gain is always used, but in other
561
    // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
562
    // so the codebook gain cannot depend on the quantized pitch gain.
563
    if (mode == MODE_12k2)
564
        p->beta = FFMIN(p->pitch_gain[4], 1.0);
565

    
566
    fixed_sparse->pitch_lag  = p->pitch_lag_int;
567
    fixed_sparse->pitch_fac  = p->beta;
568

    
569
    // Save pitch sharpening factor for the next subframe
570
    // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
571
    // the fact that the gains for two subframes are jointly quantized.
572
    if (mode != MODE_4k75 || subframe & 1)
573
        p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
574
}
575
/// @}
576

    
577

    
578
/// @defgroup amr_gain_decoding AMR gain decoding functions
579
/// @{
580

    
581
/**
582
 * fixed gain smoothing
583
 * Note that where the spec specifies the "spectrum in the q domain"
584
 * in section 6.1.4, in fact frequencies should be used.
585
 *
586
 * @param p the context
587
 * @param lsf LSFs for the current subframe, in the range [0,1]
588
 * @param lsf_avg averaged LSFs
589
 * @param mode mode of the current frame
590
 *
591
 * @return fixed gain smoothed
592
 */
593
static float fixed_gain_smooth(AMRContext *p , const float *lsf,
594
                               const float *lsf_avg, const enum Mode mode)
595
{
596
    float diff = 0.0;
597
    int i;
598

    
599
    for (i = 0; i < LP_FILTER_ORDER; i++)
600
        diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
601

    
602
    // If diff is large for ten subframes, disable smoothing for a 40-subframe
603
    // hangover period.
604
    p->diff_count++;
605
    if (diff <= 0.65)
606
        p->diff_count = 0;
607

    
608
    if (p->diff_count > 10) {
609
        p->hang_count = 0;
610
        p->diff_count--; // don't let diff_count overflow
611
    }
612

    
613
    if (p->hang_count < 40) {
614
        p->hang_count++;
615
    } else if (mode < MODE_7k4 || mode == MODE_10k2) {
616
        const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
617
        const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
618
                                       p->fixed_gain[2] + p->fixed_gain[3] +
619
                                       p->fixed_gain[4]) * 0.2;
620
        return smoothing_factor * p->fixed_gain[4] +
621
               (1.0 - smoothing_factor) * fixed_gain_mean;
622
    }
623
    return p->fixed_gain[4];
624
}
625

    
626
/**
627
 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
628
 *
629
 * @param p the context
630
 * @param amr_subframe unpacked amr subframe
631
 * @param mode mode of the current frame
632
 * @param subframe current subframe number
633
 * @param fixed_gain_factor decoded gain correction factor
634
 */
635
static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
636
                         const enum Mode mode, const int subframe,
637
                         float *fixed_gain_factor)
638
{
639
    if (mode == MODE_12k2 || mode == MODE_7k95) {
640
        p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
641
            * (1.0 / 16384.0);
642
        *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
643
            * (1.0 /  2048.0);
644
    } else {
645
        const uint16_t *gains;
646

    
647
        if (mode >= MODE_6k7) {
648
            gains = gains_high[amr_subframe->p_gain];
649
        } else if (mode >= MODE_5k15) {
650
            gains = gains_low [amr_subframe->p_gain];
651
        } else {
652
            // gain index is only coded in subframes 0,2 for MODE_4k75
653
            gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
654
        }
655

    
656
        p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
657
        *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
658
    }
659
}
660

    
661
/// @}
662

    
663

    
664
/// @defgroup amr_pre_processing AMR pre-processing functions
665
/// @{
666

    
667
/**
668
 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
669
 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
670
 *
671
 * @param out vector with filter applied
672
 * @param in source vector
673
 * @param filter phase filter coefficients
674
 *
675
 *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
676
 */
677
static void apply_ir_filter(float *out, const AMRFixed *in,
678
                            const float *filter)
679
{
680
    float filter1[AMR_SUBFRAME_SIZE],     //!< filters at pitch lag*1 and *2
681
          filter2[AMR_SUBFRAME_SIZE];
682
    int   lag = in->pitch_lag;
683
    float fac = in->pitch_fac;
684
    int i;
685

    
686
    if (lag < AMR_SUBFRAME_SIZE) {
687
        ff_celp_circ_addf(filter1, filter, filter, lag, fac,
688
                          AMR_SUBFRAME_SIZE);
689

    
690
        if (lag < AMR_SUBFRAME_SIZE >> 1)
691
            ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
692
                              AMR_SUBFRAME_SIZE);
693
    }
694

    
695
    memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
696
    for (i = 0; i < in->n; i++) {
697
        int   x = in->x[i];
698
        float y = in->y[i];
699
        const float *filterp;
700

    
701
        if (x >= AMR_SUBFRAME_SIZE - lag) {
702
            filterp = filter;
703
        } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
704
            filterp = filter1;
705
        } else
706
            filterp = filter2;
707

    
708
        ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
709
    }
710
}
711

    
712
/**
713
 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
714
 * Also know as "adaptive phase dispersion".
715
 *
716
 * This implements 3GPP TS 26.090 section 6.1(5).
717
 *
718
 * @param p the context
719
 * @param fixed_sparse algebraic codebook vector
720
 * @param fixed_vector unfiltered fixed vector
721
 * @param fixed_gain smoothed gain
722
 * @param out space for modified vector if necessary
723
 */
724
static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
725
                                    const float *fixed_vector,
726
                                    float fixed_gain, float *out)
727
{
728
    int ir_filter_nr;
729

    
730
    if (p->pitch_gain[4] < 0.6) {
731
        ir_filter_nr = 0;      // strong filtering
732
    } else if (p->pitch_gain[4] < 0.9) {
733
        ir_filter_nr = 1;      // medium filtering
734
    } else
735
        ir_filter_nr = 2;      // no filtering
736

    
737
    // detect 'onset'
738
    if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
739
        p->ir_filter_onset = 2;
740
    } else if (p->ir_filter_onset)
741
        p->ir_filter_onset--;
742

    
743
    if (!p->ir_filter_onset) {
744
        int i, count = 0;
745

    
746
        for (i = 0; i < 5; i++)
747
            if (p->pitch_gain[i] < 0.6)
748
                count++;
749
        if (count > 2)
750
            ir_filter_nr = 0;
751

    
752
        if (ir_filter_nr > p->prev_ir_filter_nr + 1)
753
            ir_filter_nr--;
754
    } else if (ir_filter_nr < 2)
755
        ir_filter_nr++;
756

    
757
    // Disable filtering for very low level of fixed_gain.
758
    // Note this step is not specified in the technical description but is in
759
    // the reference source in the function Ph_disp.
760
    if (fixed_gain < 5.0)
761
        ir_filter_nr = 2;
762

    
763
    if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
764
         && ir_filter_nr < 2) {
765
        apply_ir_filter(out, fixed_sparse,
766
                        (p->cur_frame_mode == MODE_7k95 ?
767
                             ir_filters_lookup_MODE_7k95 :
768
                             ir_filters_lookup)[ir_filter_nr]);
769
        fixed_vector = out;
770
    }
771

    
772
    // update ir filter strength history
773
    p->prev_ir_filter_nr       = ir_filter_nr;
774
    p->prev_sparse_fixed_gain  = fixed_gain;
775

    
776
    return fixed_vector;
777
}
778

    
779
/// @}
780

    
781

    
782
/// @defgroup amr_synthesis AMR synthesis functions
783
/// @{
784

    
785
/**
786
 * Conduct 10th order linear predictive coding synthesis.
787
 *
788
 * @param p             pointer to the AMRContext
789
 * @param lpc           pointer to the LPC coefficients
790
 * @param fixed_gain    fixed codebook gain for synthesis
791
 * @param fixed_vector  algebraic codebook vector
792
 * @param samples       pointer to the output speech samples
793
 * @param overflow      16-bit overflow flag
794
 */
795
static int synthesis(AMRContext *p, float *lpc,
796
                     float fixed_gain, const float *fixed_vector,
797
                     float *samples, uint8_t overflow)
798
{
799
    int i, overflow_temp = 0;
800
    float excitation[AMR_SUBFRAME_SIZE];
801

    
802
    // if an overflow has been detected, the pitch vector is scaled down by a
803
    // factor of 4
804
    if (overflow)
805
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
806
            p->pitch_vector[i] *= 0.25;
807

    
808
    ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
809
                            p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
810

    
811
    // emphasize pitch vector contribution
812
    if (p->pitch_gain[4] > 0.5 && !overflow) {
813
        float energy = ff_dot_productf(excitation, excitation,
814
                                       AMR_SUBFRAME_SIZE);
815
        float pitch_factor =
816
            p->pitch_gain[4] *
817
            (p->cur_frame_mode == MODE_12k2 ?
818
                0.25 * FFMIN(p->pitch_gain[4], 1.0) :
819
                0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
820

    
821
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
822
            excitation[i] += pitch_factor * p->pitch_vector[i];
823

    
824
        ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
825
                                                AMR_SUBFRAME_SIZE);
826
    }
827

    
828
    ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
829
                                 LP_FILTER_ORDER);
830

    
831
    // detect overflow
832
    for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
833
        if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
834
            overflow_temp = 1;
835
            samples[i] = av_clipf(samples[i], -AMR_SAMPLE_BOUND,
836
                                               AMR_SAMPLE_BOUND);
837
        }
838

    
839
    return overflow_temp;
840
}
841

    
842
/// @}
843

    
844

    
845
/// @defgroup amr_update AMR update functions
846
/// @{
847

    
848
/**
849
 * Update buffers and history at the end of decoding a subframe.
850
 *
851
 * @param p             pointer to the AMRContext
852
 */
853
static void update_state(AMRContext *p)
854
{
855
    memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
856

    
857
    memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
858
            (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
859

    
860
    memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
861
    memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
862

    
863
    memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
864
            LP_FILTER_ORDER * sizeof(float));
865
}
866

    
867
/// @}
868

    
869

    
870
/// @defgroup amr_postproc AMR Post processing functions
871
/// @{
872

    
873
/**
874
 * Get the tilt factor of a formant filter from its transfer function
875
 *
876
 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
877
 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
878
 */
879
static float tilt_factor(float *lpc_n, float *lpc_d)
880
{
881
    float rh0, rh1; // autocorrelation at lag 0 and 1
882

    
883
    // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
884
    float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
885
    float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
886

    
887
    hf[0] = 1.0;
888
    memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
889
    ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
890
                                 LP_FILTER_ORDER);
891

    
892
    rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
893
    rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
894

    
895
    // The spec only specifies this check for 12.2 and 10.2 kbit/s
896
    // modes. But in the ref source the tilt is always non-negative.
897
    return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
898
}
899

    
900
/**
901
 * Perform adaptive post-filtering to enhance the quality of the speech.
902
 * See section 6.2.1.
903
 *
904
 * @param p             pointer to the AMRContext
905
 * @param lpc           interpolated LP coefficients for this subframe
906
 * @param buf_out       output of the filter
907
 */
908
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
909
{
910
    int i;
911
    float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
912

    
913
    float speech_gain       = ff_dot_productf(samples, samples,
914
                                              AMR_SUBFRAME_SIZE);
915

    
916
    float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
917
    const float *gamma_n, *gamma_d;                       // Formant filter factor table
918
    float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
919

    
920
    if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
921
        gamma_n = ff_pow_0_7;
922
        gamma_d = ff_pow_0_75;
923
    } else {
924
        gamma_n = ff_pow_0_55;
925
        gamma_d = ff_pow_0_7;
926
    }
927

    
928
    for (i = 0; i < LP_FILTER_ORDER; i++) {
929
         lpc_n[i] = lpc[i] * gamma_n[i];
930
         lpc_d[i] = lpc[i] * gamma_d[i];
931
    }
932

    
933
    memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
934
    ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
935
                                 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
936
    memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
937
           sizeof(float) * LP_FILTER_ORDER);
938

    
939
    ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
940
                                      pole_out + LP_FILTER_ORDER,
941
                                      AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
942

    
943
    ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
944
                         AMR_SUBFRAME_SIZE);
945

    
946
    ff_adaptive_gain_control(buf_out, speech_gain, AMR_SUBFRAME_SIZE,
947
                             AMR_AGC_ALPHA, &p->postfilter_agc);
948
}
949

    
950
/// @}
951

    
952
static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
953
                              AVPacket *avpkt)
954
{
955

    
956
    AMRContext *p = avctx->priv_data;        // pointer to private data
957
    const uint8_t *buf = avpkt->data;
958
    int buf_size       = avpkt->size;
959
    float *buf_out = data;                   // pointer to the output data buffer
960
    int i, subframe;
961
    float fixed_gain_factor;
962
    AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
963
    float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
964
    float synth_fixed_gain;                  // the fixed gain that synthesis should use
965
    const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
966

    
967
    p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
968
    if (p->cur_frame_mode == MODE_DTX) {
969
        av_log_missing_feature(avctx, "dtx mode", 1);
970
        return -1;
971
    }
972

    
973
    if (p->cur_frame_mode == MODE_12k2) {
974
        lsf2lsp_5(p);
975
    } else
976
        lsf2lsp_3(p);
977

    
978
    for (i = 0; i < 4; i++)
979
        ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
980

    
981
    for (subframe = 0; subframe < 4; subframe++) {
982
        const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
983

    
984
        decode_pitch_vector(p, amr_subframe, subframe);
985

    
986
        decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
987
                            p->cur_frame_mode, subframe);
988

    
989
        // The fixed gain (section 6.1.3) depends on the fixed vector
990
        // (section 6.1.2), but the fixed vector calculation uses
991
        // pitch sharpening based on the on the pitch gain (section 6.1.3).
992
        // So the correct order is: pitch gain, pitch sharpening, fixed gain.
993
        decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
994
                     &fixed_gain_factor);
995

    
996
        pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
997

    
998
        ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
999
                            AMR_SUBFRAME_SIZE);
1000

    
1001
        p->fixed_gain[4] =
1002
            ff_amr_set_fixed_gain(fixed_gain_factor,
1003
                       ff_dot_productf(p->fixed_vector, p->fixed_vector,
1004
                                       AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
1005
                       p->prediction_error,
1006
                       energy_mean[p->cur_frame_mode], energy_pred_fac);
1007

    
1008
        // The excitation feedback is calculated without any processing such
1009
        // as fixed gain smoothing. This isn't mentioned in the specification.
1010
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1011
            p->excitation[i] *= p->pitch_gain[4];
1012
        ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
1013
                            AMR_SUBFRAME_SIZE);
1014

    
1015
        // In the ref decoder, excitation is stored with no fractional bits.
1016
        // This step prevents buzz in silent periods. The ref encoder can
1017
        // emit long sequences with pitch factor greater than one. This
1018
        // creates unwanted feedback if the excitation vector is nonzero.
1019
        // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
1020
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
1021
            p->excitation[i] = truncf(p->excitation[i]);
1022

    
1023
        // Smooth fixed gain.
1024
        // The specification is ambiguous, but in the reference source, the
1025
        // smoothed value is NOT fed back into later fixed gain smoothing.
1026
        synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
1027
                                             p->lsf_avg, p->cur_frame_mode);
1028

    
1029
        synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1030
                                             synth_fixed_gain, spare_vector);
1031

    
1032
        if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1033
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1034
            // overflow detected -> rerun synthesis scaling pitch vector down
1035
            // by a factor of 4, skipping pitch vector contribution emphasis
1036
            // and adaptive gain control
1037
            synthesis(p, p->lpc[subframe], synth_fixed_gain,
1038
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1039

    
1040
        postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1041

    
1042
        // update buffers and history
1043
        ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1044
        update_state(p);
1045
    }
1046

    
1047
    ff_acelp_apply_order_2_transfer_function(buf_out, highpass_zeros,
1048
                                             highpass_poles, highpass_gain,
1049
                                             p->high_pass_mem, AMR_BLOCK_SIZE);
1050

    
1051
    for (i = 0; i < AMR_BLOCK_SIZE; i++)
1052
        buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE,
1053
                              -1.0, 32767.0 / 32768.0);
1054

    
1055
    /* Update averaged lsf vector (used for fixed gain smoothing).
1056
     *
1057
     * Note that lsf_avg should not incorporate the current frame's LSFs
1058
     * for fixed_gain_smooth.
1059
     * The specification has an incorrect formula: the reference decoder uses
1060
     * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1061
    ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1062
                            0.84, 0.16, LP_FILTER_ORDER);
1063

    
1064
    /* report how many samples we got */
1065
    *data_size = AMR_BLOCK_SIZE * sizeof(float);
1066

    
1067
    /* return the amount of bytes consumed if everything was OK */
1068
    return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1069
}
1070

    
1071

    
1072
AVCodec amrnb_decoder = {
1073
    .name           = "amrnb",
1074
    .type           = AVMEDIA_TYPE_AUDIO,
1075
    .id             = CODEC_ID_AMR_NB,
1076
    .priv_data_size = sizeof(AMRContext),
1077
    .init           = amrnb_decode_init,
1078
    .decode         = amrnb_decode_frame,
1079
    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
1080
    .sample_fmts    = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},
1081
};