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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 Libav.
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 *
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 * Libav 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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 * Libav 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 Libav; 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
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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 "amr.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 = AV_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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        ff_amr_bit_reorder((uint16_t *) &p->frame, sizeof(AMRNBFrame), buf + 1,
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                           amr_unpacking_bitmaps_per_mode[mode]);
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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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 * 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)
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{
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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);
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}
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/**
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 * 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)
243
{
244
    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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252
    if (sign) {
253
        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(*lsf_r));
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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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268
    ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER);
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}
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/**
272
 * Decode a set of 5 split-matrix quantized lsf indexes into 2 lsp vectors.
273
 *
274
 * @param p                 pointer to the AMRContext
275
 */
276
static void lsf2lsp_5(AMRContext *p)
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{
278
    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];
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    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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/**
301
 * Decode a set of 3 split-matrix quantized lsf indexes into an lsp vector.
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 *
303
 * @param p                 pointer to the AMRContext
304
 */
305
static void lsf2lsp_3(AMRContext *p)
306
{
307
    const uint16_t *lsf_param = p->frame.lsf;
308
    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]];
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    memcpy(lsf_r, lsf_quantizer, 3 * sizeof(*lsf_r));
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    lsf_quantizer = lsf_3_2[lsf_param[1] << (p->cur_frame_mode <= MODE_5k15)];
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    memcpy(lsf_r + 3, lsf_quantizer, 3 * sizeof(*lsf_r));
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    lsf_quantizer = (p->cur_frame_mode <= MODE_5k15 ? lsf_3_3_MODE_5k15 : lsf_3_3)[lsf_param[2]];
320
    memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r));
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    // calculate mean-removed LSF vector and add mean
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    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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    // store data for computing the next frame's LSFs
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    interpolate_lsf(p->lsf_q, lsf_q);
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    memcpy(p->prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));
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    ff_acelp_lsf2lspd(p->lsp[3], lsf_q, LP_FILTER_ORDER);
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    // interpolate LSP vectors at subframes 1, 2 and 3
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    for (i = 1; i <= 3; i++)
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        for(j = 0; j < LP_FILTER_ORDER; j++)
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            p->lsp[i-1][j] = p->prev_lsp_sub4[j] +
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                (p->lsp[3][j] - p->prev_lsp_sub4[j]) * 0.25 * i;
339
}
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/// @}
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/// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions
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/// @{
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347
/**
348
 * Like ff_decode_pitch_lag(), but with 1/6 resolution
349
 */
350
static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index,
351
                                 const int prev_lag_int, const int subframe)
352
{
353
    if (subframe == 0 || subframe == 2) {
354
        if (pitch_index < 463) {
355
            *lag_int  = (pitch_index + 107) * 10923 >> 16;
356
            *lag_frac = pitch_index - *lag_int * 6 + 105;
357
        } else {
358
            *lag_int  = pitch_index - 368;
359
            *lag_frac = 0;
360
        }
361
    } else {
362
        *lag_int  = ((pitch_index + 5) * 10923 >> 16) - 1;
363
        *lag_frac = pitch_index - *lag_int * 6 - 3;
364
        *lag_int += av_clip(prev_lag_int - 5, PITCH_LAG_MIN_MODE_12k2,
365
                            PITCH_DELAY_MAX - 9);
366
    }
367
}
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369
static void decode_pitch_vector(AMRContext *p,
370
                                const AMRNBSubframe *amr_subframe,
371
                                const int subframe)
372
{
373
    int pitch_lag_int, pitch_lag_frac;
374
    enum Mode mode = p->cur_frame_mode;
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376
    if (p->cur_frame_mode == MODE_12k2) {
377
        decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac,
378
                             amr_subframe->p_lag, p->pitch_lag_int,
379
                             subframe);
380
    } else
381
        ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac,
382
                            amr_subframe->p_lag,
383
                            p->pitch_lag_int, subframe,
384
                            mode != MODE_4k75 && mode != MODE_5k15,
385
                            mode <= MODE_6k7 ? 4 : (mode == MODE_7k95 ? 5 : 6));
386

    
387
    p->pitch_lag_int = pitch_lag_int; // store previous lag in a uint8_t
388

    
389
    pitch_lag_frac <<= (p->cur_frame_mode != MODE_12k2);
390

    
391
    pitch_lag_int += pitch_lag_frac > 0;
392

    
393
    /* Calculate the pitch vector by interpolating the past excitation at the
394
       pitch lag using a b60 hamming windowed sinc function.   */
395
    ff_acelp_interpolatef(p->excitation, p->excitation + 1 - pitch_lag_int,
396
                          ff_b60_sinc, 6,
397
                          pitch_lag_frac + 6 - 6*(pitch_lag_frac > 0),
398
                          10, AMR_SUBFRAME_SIZE);
399

    
400
    memcpy(p->pitch_vector, p->excitation, AMR_SUBFRAME_SIZE * sizeof(float));
401
}
402

    
403
/// @}
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405

    
406
/// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions
407
/// @{
408

    
409
/**
410
 * Decode a 10-bit algebraic codebook index from a 10.2 kbit/s frame.
411
 */
412
static void decode_10bit_pulse(int code, int pulse_position[8],
413
                               int i1, int i2, int i3)
414
{
415
    // coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of
416
    // the 3 pulses and the upper 7 bits being coded in base 5
417
    const uint8_t *positions = base_five_table[code >> 3];
418
    pulse_position[i1] = (positions[2] << 1) + ( code       & 1);
419
    pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1);
420
    pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1);
421
}
422

    
423
/**
424
 * Decode the algebraic codebook index to pulse positions and signs and
425
 * construct the algebraic codebook vector for MODE_10k2.
426
 *
427
 * @param fixed_index          positions of the eight pulses
428
 * @param fixed_sparse         pointer to the algebraic codebook vector
429
 */
430
static void decode_8_pulses_31bits(const int16_t *fixed_index,
431
                                   AMRFixed *fixed_sparse)
432
{
433
    int pulse_position[8];
434
    int i, temp;
435

    
436
    decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1);
437
    decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5);
438

    
439
    // coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of
440
    // the 2 pulses and the upper 5 bits being coded in base 5
441
    temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5;
442
    pulse_position[3] = temp % 5;
443
    pulse_position[7] = temp / 5;
444
    if (pulse_position[7] & 1)
445
        pulse_position[3] = 4 - pulse_position[3];
446
    pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6]       & 1);
447
    pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1);
448

    
449
    fixed_sparse->n = 8;
450
    for (i = 0; i < 4; i++) {
451
        const int pos1   = (pulse_position[i]     << 2) + i;
452
        const int pos2   = (pulse_position[i + 4] << 2) + i;
453
        const float sign = fixed_index[i] ? -1.0 : 1.0;
454
        fixed_sparse->x[i    ] = pos1;
455
        fixed_sparse->x[i + 4] = pos2;
456
        fixed_sparse->y[i    ] = sign;
457
        fixed_sparse->y[i + 4] = pos2 < pos1 ? -sign : sign;
458
    }
459
}
460

    
461
/**
462
 * Decode the algebraic codebook index to pulse positions and signs,
463
 * then construct the algebraic codebook vector.
464
 *
465
 *                              nb of pulses | bits encoding pulses
466
 * For MODE_4k75 or MODE_5k15,             2 | 1-3, 4-6, 7
467
 *                  MODE_5k9,              2 | 1,   2-4, 5-6, 7-9
468
 *                  MODE_6k7,              3 | 1-3, 4,   5-7, 8,  9-11
469
 *      MODE_7k4 or MODE_7k95,             4 | 1-3, 4-6, 7-9, 10, 11-13
470
 *
471
 * @param fixed_sparse pointer to the algebraic codebook vector
472
 * @param pulses       algebraic codebook indexes
473
 * @param mode         mode of the current frame
474
 * @param subframe     current subframe number
475
 */
476
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
477
                                const enum Mode mode, const int subframe)
478
{
479
    assert(MODE_4k75 <= mode && mode <= MODE_12k2);
480

    
481
    if (mode == MODE_12k2) {
482
        ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3);
483
    } else if (mode == MODE_10k2) {
484
        decode_8_pulses_31bits(pulses, fixed_sparse);
485
    } else {
486
        int *pulse_position = fixed_sparse->x;
487
        int i, pulse_subset;
488
        const int fixed_index = pulses[0];
489

    
490
        if (mode <= MODE_5k15) {
491
            pulse_subset      = ((fixed_index >> 3) & 8)     + (subframe << 1);
492
            pulse_position[0] = ( fixed_index       & 7) * 5 + track_position[pulse_subset];
493
            pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1];
494
            fixed_sparse->n = 2;
495
        } else if (mode == MODE_5k9) {
496
            pulse_subset      = ((fixed_index & 1) << 1) + 1;
497
            pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset;
498
            pulse_subset      = (fixed_index  >> 4) & 3;
499
            pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0);
500
            fixed_sparse->n = pulse_position[0] == pulse_position[1] ? 1 : 2;
501
        } else if (mode == MODE_6k7) {
502
            pulse_position[0] = (fixed_index        & 7) * 5;
503
            pulse_subset      = (fixed_index  >> 2) & 2;
504
            pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1;
505
            pulse_subset      = (fixed_index  >> 6) & 2;
506
            pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2;
507
            fixed_sparse->n = 3;
508
        } else { // mode <= MODE_7k95
509
            pulse_position[0] = gray_decode[ fixed_index        & 7];
510
            pulse_position[1] = gray_decode[(fixed_index >> 3)  & 7] + 1;
511
            pulse_position[2] = gray_decode[(fixed_index >> 6)  & 7] + 2;
512
            pulse_subset      = (fixed_index >> 9) & 1;
513
            pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3;
514
            fixed_sparse->n = 4;
515
        }
516
        for (i = 0; i < fixed_sparse->n; i++)
517
            fixed_sparse->y[i] = (pulses[1] >> i) & 1 ? 1.0 : -1.0;
518
    }
519
}
520

    
521
/**
522
 * Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)
523
 *
524
 * @param p the context
525
 * @param subframe unpacked amr subframe
526
 * @param mode mode of the current frame
527
 * @param fixed_sparse sparse respresentation of the fixed vector
528
 */
529
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,
530
                             AMRFixed *fixed_sparse)
531
{
532
    // The spec suggests the current pitch gain is always used, but in other
533
    // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)
534
    // so the codebook gain cannot depend on the quantized pitch gain.
535
    if (mode == MODE_12k2)
536
        p->beta = FFMIN(p->pitch_gain[4], 1.0);
537

    
538
    fixed_sparse->pitch_lag  = p->pitch_lag_int;
539
    fixed_sparse->pitch_fac  = p->beta;
540

    
541
    // Save pitch sharpening factor for the next subframe
542
    // MODE_4k75 only updates on the 2nd and 4th subframes - this follows from
543
    // the fact that the gains for two subframes are jointly quantized.
544
    if (mode != MODE_4k75 || subframe & 1)
545
        p->beta = av_clipf(p->pitch_gain[4], 0.0, SHARP_MAX);
546
}
547
/// @}
548

    
549

    
550
/// @defgroup amr_gain_decoding AMR gain decoding functions
551
/// @{
552

    
553
/**
554
 * fixed gain smoothing
555
 * Note that where the spec specifies the "spectrum in the q domain"
556
 * in section 6.1.4, in fact frequencies should be used.
557
 *
558
 * @param p the context
559
 * @param lsf LSFs for the current subframe, in the range [0,1]
560
 * @param lsf_avg averaged LSFs
561
 * @param mode mode of the current frame
562
 *
563
 * @return fixed gain smoothed
564
 */
565
static float fixed_gain_smooth(AMRContext *p , const float *lsf,
566
                               const float *lsf_avg, const enum Mode mode)
567
{
568
    float diff = 0.0;
569
    int i;
570

    
571
    for (i = 0; i < LP_FILTER_ORDER; i++)
572
        diff += fabs(lsf_avg[i] - lsf[i]) / lsf_avg[i];
573

    
574
    // If diff is large for ten subframes, disable smoothing for a 40-subframe
575
    // hangover period.
576
    p->diff_count++;
577
    if (diff <= 0.65)
578
        p->diff_count = 0;
579

    
580
    if (p->diff_count > 10) {
581
        p->hang_count = 0;
582
        p->diff_count--; // don't let diff_count overflow
583
    }
584

    
585
    if (p->hang_count < 40) {
586
        p->hang_count++;
587
    } else if (mode < MODE_7k4 || mode == MODE_10k2) {
588
        const float smoothing_factor = av_clipf(4.0 * diff - 1.6, 0.0, 1.0);
589
        const float fixed_gain_mean = (p->fixed_gain[0] + p->fixed_gain[1] +
590
                                       p->fixed_gain[2] + p->fixed_gain[3] +
591
                                       p->fixed_gain[4]) * 0.2;
592
        return smoothing_factor * p->fixed_gain[4] +
593
               (1.0 - smoothing_factor) * fixed_gain_mean;
594
    }
595
    return p->fixed_gain[4];
596
}
597

    
598
/**
599
 * Decode pitch gain and fixed gain factor (part of section 6.1.3).
600
 *
601
 * @param p the context
602
 * @param amr_subframe unpacked amr subframe
603
 * @param mode mode of the current frame
604
 * @param subframe current subframe number
605
 * @param fixed_gain_factor decoded gain correction factor
606
 */
607
static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe,
608
                         const enum Mode mode, const int subframe,
609
                         float *fixed_gain_factor)
610
{
611
    if (mode == MODE_12k2 || mode == MODE_7k95) {
612
        p->pitch_gain[4]   = qua_gain_pit [amr_subframe->p_gain    ]
613
            * (1.0 / 16384.0);
614
        *fixed_gain_factor = qua_gain_code[amr_subframe->fixed_gain]
615
            * (1.0 /  2048.0);
616
    } else {
617
        const uint16_t *gains;
618

    
619
        if (mode >= MODE_6k7) {
620
            gains = gains_high[amr_subframe->p_gain];
621
        } else if (mode >= MODE_5k15) {
622
            gains = gains_low [amr_subframe->p_gain];
623
        } else {
624
            // gain index is only coded in subframes 0,2 for MODE_4k75
625
            gains = gains_MODE_4k75[(p->frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)];
626
        }
627

    
628
        p->pitch_gain[4]   = gains[0] * (1.0 / 16384.0);
629
        *fixed_gain_factor = gains[1] * (1.0 /  4096.0);
630
    }
631
}
632

    
633
/// @}
634

    
635

    
636
/// @defgroup amr_pre_processing AMR pre-processing functions
637
/// @{
638

    
639
/**
640
 * Circularly convolve a sparse fixed vector with a phase dispersion impulse
641
 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
642
 *
643
 * @param out vector with filter applied
644
 * @param in source vector
645
 * @param filter phase filter coefficients
646
 *
647
 *  out[n] = sum(i,0,len-1){ in[i] * filter[(len + n - i)%len] }
648
 */
649
static void apply_ir_filter(float *out, const AMRFixed *in,
650
                            const float *filter)
651
{
652
    float filter1[AMR_SUBFRAME_SIZE],     //!< filters at pitch lag*1 and *2
653
          filter2[AMR_SUBFRAME_SIZE];
654
    int   lag = in->pitch_lag;
655
    float fac = in->pitch_fac;
656
    int i;
657

    
658
    if (lag < AMR_SUBFRAME_SIZE) {
659
        ff_celp_circ_addf(filter1, filter, filter, lag, fac,
660
                          AMR_SUBFRAME_SIZE);
661

    
662
        if (lag < AMR_SUBFRAME_SIZE >> 1)
663
            ff_celp_circ_addf(filter2, filter, filter1, lag, fac,
664
                              AMR_SUBFRAME_SIZE);
665
    }
666

    
667
    memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE);
668
    for (i = 0; i < in->n; i++) {
669
        int   x = in->x[i];
670
        float y = in->y[i];
671
        const float *filterp;
672

    
673
        if (x >= AMR_SUBFRAME_SIZE - lag) {
674
            filterp = filter;
675
        } else if (x >= AMR_SUBFRAME_SIZE - (lag << 1)) {
676
            filterp = filter1;
677
        } else
678
            filterp = filter2;
679

    
680
        ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE);
681
    }
682
}
683

    
684
/**
685
 * Reduce fixed vector sparseness by smoothing with one of three IR filters.
686
 * Also know as "adaptive phase dispersion".
687
 *
688
 * This implements 3GPP TS 26.090 section 6.1(5).
689
 *
690
 * @param p the context
691
 * @param fixed_sparse algebraic codebook vector
692
 * @param fixed_vector unfiltered fixed vector
693
 * @param fixed_gain smoothed gain
694
 * @param out space for modified vector if necessary
695
 */
696
static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse,
697
                                    const float *fixed_vector,
698
                                    float fixed_gain, float *out)
699
{
700
    int ir_filter_nr;
701

    
702
    if (p->pitch_gain[4] < 0.6) {
703
        ir_filter_nr = 0;      // strong filtering
704
    } else if (p->pitch_gain[4] < 0.9) {
705
        ir_filter_nr = 1;      // medium filtering
706
    } else
707
        ir_filter_nr = 2;      // no filtering
708

    
709
    // detect 'onset'
710
    if (fixed_gain > 2.0 * p->prev_sparse_fixed_gain) {
711
        p->ir_filter_onset = 2;
712
    } else if (p->ir_filter_onset)
713
        p->ir_filter_onset--;
714

    
715
    if (!p->ir_filter_onset) {
716
        int i, count = 0;
717

    
718
        for (i = 0; i < 5; i++)
719
            if (p->pitch_gain[i] < 0.6)
720
                count++;
721
        if (count > 2)
722
            ir_filter_nr = 0;
723

    
724
        if (ir_filter_nr > p->prev_ir_filter_nr + 1)
725
            ir_filter_nr--;
726
    } else if (ir_filter_nr < 2)
727
        ir_filter_nr++;
728

    
729
    // Disable filtering for very low level of fixed_gain.
730
    // Note this step is not specified in the technical description but is in
731
    // the reference source in the function Ph_disp.
732
    if (fixed_gain < 5.0)
733
        ir_filter_nr = 2;
734

    
735
    if (p->cur_frame_mode != MODE_7k4 && p->cur_frame_mode < MODE_10k2
736
         && ir_filter_nr < 2) {
737
        apply_ir_filter(out, fixed_sparse,
738
                        (p->cur_frame_mode == MODE_7k95 ?
739
                             ir_filters_lookup_MODE_7k95 :
740
                             ir_filters_lookup)[ir_filter_nr]);
741
        fixed_vector = out;
742
    }
743

    
744
    // update ir filter strength history
745
    p->prev_ir_filter_nr       = ir_filter_nr;
746
    p->prev_sparse_fixed_gain  = fixed_gain;
747

    
748
    return fixed_vector;
749
}
750

    
751
/// @}
752

    
753

    
754
/// @defgroup amr_synthesis AMR synthesis functions
755
/// @{
756

    
757
/**
758
 * Conduct 10th order linear predictive coding synthesis.
759
 *
760
 * @param p             pointer to the AMRContext
761
 * @param lpc           pointer to the LPC coefficients
762
 * @param fixed_gain    fixed codebook gain for synthesis
763
 * @param fixed_vector  algebraic codebook vector
764
 * @param samples       pointer to the output speech samples
765
 * @param overflow      16-bit overflow flag
766
 */
767
static int synthesis(AMRContext *p, float *lpc,
768
                     float fixed_gain, const float *fixed_vector,
769
                     float *samples, uint8_t overflow)
770
{
771
    int i;
772
    float excitation[AMR_SUBFRAME_SIZE];
773

    
774
    // if an overflow has been detected, the pitch vector is scaled down by a
775
    // factor of 4
776
    if (overflow)
777
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
778
            p->pitch_vector[i] *= 0.25;
779

    
780
    ff_weighted_vector_sumf(excitation, p->pitch_vector, fixed_vector,
781
                            p->pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);
782

    
783
    // emphasize pitch vector contribution
784
    if (p->pitch_gain[4] > 0.5 && !overflow) {
785
        float energy = ff_dot_productf(excitation, excitation,
786
                                       AMR_SUBFRAME_SIZE);
787
        float pitch_factor =
788
            p->pitch_gain[4] *
789
            (p->cur_frame_mode == MODE_12k2 ?
790
                0.25 * FFMIN(p->pitch_gain[4], 1.0) :
791
                0.5  * FFMIN(p->pitch_gain[4], SHARP_MAX));
792

    
793
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
794
            excitation[i] += pitch_factor * p->pitch_vector[i];
795

    
796
        ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy,
797
                                                AMR_SUBFRAME_SIZE);
798
    }
799

    
800
    ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE,
801
                                 LP_FILTER_ORDER);
802

    
803
    // detect overflow
804
    for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
805
        if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
806
            return 1;
807
        }
808

    
809
    return 0;
810
}
811

    
812
/// @}
813

    
814

    
815
/// @defgroup amr_update AMR update functions
816
/// @{
817

    
818
/**
819
 * Update buffers and history at the end of decoding a subframe.
820
 *
821
 * @param p             pointer to the AMRContext
822
 */
823
static void update_state(AMRContext *p)
824
{
825
    memcpy(p->prev_lsp_sub4, p->lsp[3], LP_FILTER_ORDER * sizeof(p->lsp[3][0]));
826

    
827
    memmove(&p->excitation_buf[0], &p->excitation_buf[AMR_SUBFRAME_SIZE],
828
            (PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float));
829

    
830
    memmove(&p->pitch_gain[0], &p->pitch_gain[1], 4 * sizeof(float));
831
    memmove(&p->fixed_gain[0], &p->fixed_gain[1], 4 * sizeof(float));
832

    
833
    memmove(&p->samples_in[0], &p->samples_in[AMR_SUBFRAME_SIZE],
834
            LP_FILTER_ORDER * sizeof(float));
835
}
836

    
837
/// @}
838

    
839

    
840
/// @defgroup amr_postproc AMR Post processing functions
841
/// @{
842

    
843
/**
844
 * Get the tilt factor of a formant filter from its transfer function
845
 *
846
 * @param lpc_n LP_FILTER_ORDER coefficients of the numerator
847
 * @param lpc_d LP_FILTER_ORDER coefficients of the denominator
848
 */
849
static float tilt_factor(float *lpc_n, float *lpc_d)
850
{
851
    float rh0, rh1; // autocorrelation at lag 0 and 1
852

    
853
    // LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf
854
    float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 };
855
    float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response
856

    
857
    hf[0] = 1.0;
858
    memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER);
859
    ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
860
                                 LP_FILTER_ORDER);
861

    
862
    rh0 = ff_dot_productf(hf, hf,     AMR_TILT_RESPONSE);
863
    rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
864

    
865
    // The spec only specifies this check for 12.2 and 10.2 kbit/s
866
    // modes. But in the ref source the tilt is always non-negative.
867
    return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0;
868
}
869

    
870
/**
871
 * Perform adaptive post-filtering to enhance the quality of the speech.
872
 * See section 6.2.1.
873
 *
874
 * @param p             pointer to the AMRContext
875
 * @param lpc           interpolated LP coefficients for this subframe
876
 * @param buf_out       output of the filter
877
 */
878
static void postfilter(AMRContext *p, float *lpc, float *buf_out)
879
{
880
    int i;
881
    float *samples          = p->samples_in + LP_FILTER_ORDER; // Start of input
882

    
883
    float speech_gain       = ff_dot_productf(samples, samples,
884
                                              AMR_SUBFRAME_SIZE);
885

    
886
    float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER];  // Output of pole filter
887
    const float *gamma_n, *gamma_d;                       // Formant filter factor table
888
    float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients
889

    
890
    if (p->cur_frame_mode == MODE_12k2 || p->cur_frame_mode == MODE_10k2) {
891
        gamma_n = ff_pow_0_7;
892
        gamma_d = ff_pow_0_75;
893
    } else {
894
        gamma_n = ff_pow_0_55;
895
        gamma_d = ff_pow_0_7;
896
    }
897

    
898
    for (i = 0; i < LP_FILTER_ORDER; i++) {
899
         lpc_n[i] = lpc[i] * gamma_n[i];
900
         lpc_d[i] = lpc[i] * gamma_d[i];
901
    }
902

    
903
    memcpy(pole_out, p->postfilter_mem, sizeof(float) * LP_FILTER_ORDER);
904
    ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples,
905
                                 AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
906
    memcpy(p->postfilter_mem, pole_out + AMR_SUBFRAME_SIZE,
907
           sizeof(float) * LP_FILTER_ORDER);
908

    
909
    ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n,
910
                                      pole_out + LP_FILTER_ORDER,
911
                                      AMR_SUBFRAME_SIZE, LP_FILTER_ORDER);
912

    
913
    ff_tilt_compensation(&p->tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out,
914
                         AMR_SUBFRAME_SIZE);
915

    
916
    ff_adaptive_gain_control(buf_out, buf_out, speech_gain, AMR_SUBFRAME_SIZE,
917
                             AMR_AGC_ALPHA, &p->postfilter_agc);
918
}
919

    
920
/// @}
921

    
922
static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
923
                              AVPacket *avpkt)
924
{
925

    
926
    AMRContext *p = avctx->priv_data;        // pointer to private data
927
    const uint8_t *buf = avpkt->data;
928
    int buf_size       = avpkt->size;
929
    float *buf_out = data;                   // pointer to the output data buffer
930
    int i, subframe;
931
    float fixed_gain_factor;
932
    AMRFixed fixed_sparse = {0};             // fixed vector up to anti-sparseness processing
933
    float spare_vector[AMR_SUBFRAME_SIZE];   // extra stack space to hold result from anti-sparseness processing
934
    float synth_fixed_gain;                  // the fixed gain that synthesis should use
935
    const float *synth_fixed_vector;         // pointer to the fixed vector that synthesis should use
936

    
937
    p->cur_frame_mode = unpack_bitstream(p, buf, buf_size);
938
    if (p->cur_frame_mode == MODE_DTX) {
939
        av_log_missing_feature(avctx, "dtx mode", 1);
940
        return -1;
941
    }
942

    
943
    if (p->cur_frame_mode == MODE_12k2) {
944
        lsf2lsp_5(p);
945
    } else
946
        lsf2lsp_3(p);
947

    
948
    for (i = 0; i < 4; i++)
949
        ff_acelp_lspd2lpc(p->lsp[i], p->lpc[i], 5);
950

    
951
    for (subframe = 0; subframe < 4; subframe++) {
952
        const AMRNBSubframe *amr_subframe = &p->frame.subframe[subframe];
953

    
954
        decode_pitch_vector(p, amr_subframe, subframe);
955

    
956
        decode_fixed_sparse(&fixed_sparse, amr_subframe->pulses,
957
                            p->cur_frame_mode, subframe);
958

    
959
        // The fixed gain (section 6.1.3) depends on the fixed vector
960
        // (section 6.1.2), but the fixed vector calculation uses
961
        // pitch sharpening based on the on the pitch gain (section 6.1.3).
962
        // So the correct order is: pitch gain, pitch sharpening, fixed gain.
963
        decode_gains(p, amr_subframe, p->cur_frame_mode, subframe,
964
                     &fixed_gain_factor);
965

    
966
        pitch_sharpening(p, subframe, p->cur_frame_mode, &fixed_sparse);
967

    
968
        ff_set_fixed_vector(p->fixed_vector, &fixed_sparse, 1.0,
969
                            AMR_SUBFRAME_SIZE);
970

    
971
        p->fixed_gain[4] =
972
            ff_amr_set_fixed_gain(fixed_gain_factor,
973
                       ff_dot_productf(p->fixed_vector, p->fixed_vector,
974
                                       AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
975
                       p->prediction_error,
976
                       energy_mean[p->cur_frame_mode], energy_pred_fac);
977

    
978
        // The excitation feedback is calculated without any processing such
979
        // as fixed gain smoothing. This isn't mentioned in the specification.
980
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
981
            p->excitation[i] *= p->pitch_gain[4];
982
        ff_set_fixed_vector(p->excitation, &fixed_sparse, p->fixed_gain[4],
983
                            AMR_SUBFRAME_SIZE);
984

    
985
        // In the ref decoder, excitation is stored with no fractional bits.
986
        // This step prevents buzz in silent periods. The ref encoder can
987
        // emit long sequences with pitch factor greater than one. This
988
        // creates unwanted feedback if the excitation vector is nonzero.
989
        // (e.g. test sequence T19_795.COD in 3GPP TS 26.074)
990
        for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
991
            p->excitation[i] = truncf(p->excitation[i]);
992

    
993
        // Smooth fixed gain.
994
        // The specification is ambiguous, but in the reference source, the
995
        // smoothed value is NOT fed back into later fixed gain smoothing.
996
        synth_fixed_gain = fixed_gain_smooth(p, p->lsf_q[subframe],
997
                                             p->lsf_avg, p->cur_frame_mode);
998

    
999
        synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p->fixed_vector,
1000
                                             synth_fixed_gain, spare_vector);
1001

    
1002
        if (synthesis(p, p->lpc[subframe], synth_fixed_gain,
1003
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 0))
1004
            // overflow detected -> rerun synthesis scaling pitch vector down
1005
            // by a factor of 4, skipping pitch vector contribution emphasis
1006
            // and adaptive gain control
1007
            synthesis(p, p->lpc[subframe], synth_fixed_gain,
1008
                      synth_fixed_vector, &p->samples_in[LP_FILTER_ORDER], 1);
1009

    
1010
        postfilter(p, p->lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE);
1011

    
1012
        // update buffers and history
1013
        ff_clear_fixed_vector(p->fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE);
1014
        update_state(p);
1015
    }
1016

    
1017
    ff_acelp_apply_order_2_transfer_function(buf_out, buf_out, highpass_zeros,
1018
                                             highpass_poles,
1019
                                             highpass_gain * AMR_SAMPLE_SCALE,
1020
                                             p->high_pass_mem, AMR_BLOCK_SIZE);
1021

    
1022
    /* Update averaged lsf vector (used for fixed gain smoothing).
1023
     *
1024
     * Note that lsf_avg should not incorporate the current frame's LSFs
1025
     * for fixed_gain_smooth.
1026
     * The specification has an incorrect formula: the reference decoder uses
1027
     * qbar(n-1) rather than qbar(n) in section 6.1(4) equation 71. */
1028
    ff_weighted_vector_sumf(p->lsf_avg, p->lsf_avg, p->lsf_q[3],
1029
                            0.84, 0.16, LP_FILTER_ORDER);
1030

    
1031
    /* report how many samples we got */
1032
    *data_size = AMR_BLOCK_SIZE * sizeof(float);
1033

    
1034
    /* return the amount of bytes consumed if everything was OK */
1035
    return frame_sizes_nb[p->cur_frame_mode] + 1; // +7 for rounding and +8 for TOC
1036
}
1037

    
1038

    
1039
AVCodec ff_amrnb_decoder = {
1040
    .name           = "amrnb",
1041
    .type           = AVMEDIA_TYPE_AUDIO,
1042
    .id             = CODEC_ID_AMR_NB,
1043
    .priv_data_size = sizeof(AMRContext),
1044
    .init           = amrnb_decode_init,
1045
    .decode         = amrnb_decode_frame,
1046
    .long_name      = NULL_IF_CONFIG_SMALL("Adaptive Multi-Rate NarrowBand"),
1047
    .sample_fmts    = (enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},
1048
};