~ppsspp/ppsspp/ffmpeg-upstream

1 by Sérgio Benjamim
FFmpeg 2.7.1 source for ppsspp.
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/*
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 * gain code, gain pitch and pitch delay decoding
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 *
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 * Copyright (c) 2008 Vladimir Voroshilov
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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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#ifndef AVCODEC_ACELP_PITCH_DELAY_H
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#define AVCODEC_ACELP_PITCH_DELAY_H
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#include <stdint.h>
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#include "audiodsp.h"
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#define PITCH_DELAY_MIN             20
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#define PITCH_DELAY_MAX             143
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/**
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 * @brief Decode pitch delay of the first subframe encoded by 8 bits with 1/3
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 *        resolution.
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 * @param ac_index adaptive codebook index (8 bits)
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 *
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 * @return pitch delay in 1/3 units
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 *
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 * Pitch delay is coded:
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 *    with 1/3 resolution, 19  < pitch_delay <  85
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 *    integers only,       85 <= pitch_delay <= 143
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 */
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int ff_acelp_decode_8bit_to_1st_delay3(int ac_index);
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/**
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 * @brief Decode pitch delay of the second subframe encoded by 5 or 6 bits
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 *        with 1/3 precision.
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 * @param ac_index adaptive codebook index (5 or 6 bits)
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 * @param pitch_delay_min lower bound (integer) of pitch delay interval
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 *                      for second subframe
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 *
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 * @return pitch delay in 1/3 units
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 *
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 * Pitch delay is coded:
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 *    with 1/3 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
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 *
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 * @remark The routine is used in G.729 @@8k, AMR @@10.2k, AMR @@7.95k,
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 *         AMR @@7.4k for the second subframe.
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 */
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int ff_acelp_decode_5_6_bit_to_2nd_delay3(
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        int ac_index,
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        int pitch_delay_min);
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/**
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 * @brief Decode pitch delay with 1/3 precision.
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 * @param ac_index adaptive codebook index (4 bits)
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 * @param pitch_delay_min lower bound (integer) of pitch delay interval for
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 *                      second subframe
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 *
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 * @return pitch delay in 1/3 units
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 *
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 * Pitch delay is coded:
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 *    integers only,          -6  < pitch_delay - int(prev_pitch_delay) <= -2
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 *    with 1/3 resolution,    -2  < pitch_delay - int(prev_pitch_delay) <  1
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 *    integers only,           1 <= pitch_delay - int(prev_pitch_delay) <  5
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 *
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 * @remark The routine is used in G.729 @@6.4k, AMR @@6.7k, AMR @@5.9k,
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 *         AMR @@5.15k, AMR @@4.75k for the second subframe.
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 */
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int ff_acelp_decode_4bit_to_2nd_delay3(
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        int ac_index,
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        int pitch_delay_min);
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/**
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 * @brief Decode pitch delay of the first subframe encoded by 9 bits
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 *        with 1/6 precision.
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 * @param ac_index adaptive codebook index (9 bits)
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 *
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 * @return pitch delay in 1/6 units
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 *
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 * Pitch delay is coded:
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 *    with 1/6 resolution,  17  < pitch_delay <  95
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 *    integers only,        95 <= pitch_delay <= 143
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 *
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 * @remark The routine is used in AMR @@12.2k for the first and third subframes.
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 */
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int ff_acelp_decode_9bit_to_1st_delay6(int ac_index);
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/**
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 * @brief Decode pitch delay of the second subframe encoded by 6 bits
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 *        with 1/6 precision.
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 * @param ac_index adaptive codebook index (6 bits)
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 * @param pitch_delay_min lower bound (integer) of pitch delay interval for
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 *                      second subframe
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 *
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 * @return pitch delay in 1/6 units
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 *
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 * Pitch delay is coded:
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 *    with 1/6 resolution, -6 < pitch_delay - int(prev_pitch_delay) < 5
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 *
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 * @remark The routine is used in AMR @@12.2k for the second and fourth subframes.
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 */
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int ff_acelp_decode_6bit_to_2nd_delay6(
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        int ac_index,
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        int pitch_delay_min);
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/**
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 * @brief Update past quantized energies
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 * @param[in,out]  quant_energy  past quantized energies (5.10)
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 * @param gain_corr_factor gain correction factor
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 * @param log2_ma_pred_order log2() of MA prediction order
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 * @param erasure frame erasure flag
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 *
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 * If frame erasure flag is not equal to zero, memory is updated with
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 * averaged energy, attenuated by 4dB:
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 *     max(avg(quant_energy[i])-4, -14), i=0,ma_pred_order
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 *
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 * In normal mode memory is updated with
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 *     Er - Ep = 20 * log10(gain_corr_factor)
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 *
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 * @remark The routine is used in G.729 and AMR (all modes).
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 */
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void ff_acelp_update_past_gain(
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        int16_t* quant_energy,
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        int gain_corr_factor,
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        int log2_ma_pred_order,
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        int erasure);
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/**
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 * @brief Decode the adaptive codebook gain and add
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 *        correction (4.1.5 and 3.9.1 of G.729).
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 * @param adsp initialized audio DSP context
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 * @param gain_corr_factor gain correction factor (2.13)
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 * @param fc_v fixed-codebook vector (2.13)
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 * @param mr_energy mean innovation energy and fixed-point correction (7.13)
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 * @param[in,out]  quant_energy  past quantized energies (5.10)
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 * @param subframe_size length of subframe
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 *
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 * @return quantized fixed-codebook gain (14.1)
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 *
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 * The routine implements equations 69, 66 and 71 of the G.729 specification (3.9.1)
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 *
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 *    Em   - mean innovation energy (dB, constant, depends on decoding algorithm)
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 *    Ep   - mean-removed predicted energy (dB)
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 *    Er   - mean-removed innovation energy (dB)
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 *    Ei   - mean energy of the fixed-codebook contribution (dB)
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 *    N    - subframe_size
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 *    M    - MA (Moving Average) prediction order
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 *    gc   - fixed-codebook gain
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 *    gc_p - predicted fixed-codebook gain
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 *
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 *    Fixed codebook gain is computed using predicted gain gc_p and
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 *    correction factor gain_corr_factor as shown below:
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 *
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 *        gc = gc_p * gain_corr_factor
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 *
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 *    The predicted fixed codebook gain gc_p is found by predicting
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 *    the energy of the fixed-codebook contribution from the energy
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 *    of previous fixed-codebook contributions.
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 *
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 *        mean = 1/N * sum(i,0,N){ fc_v[i] * fc_v[i] }
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 *
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 *        Ei = 10log(mean)
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 *
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 *        Er = 10log(1/N * gc^2 * mean) - Em = 20log(gc) + Ei - Em
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 *
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 *    Replacing Er with Ep and gc with gc_p we will receive:
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 *
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 *        Ep = 10log(1/N * gc_p^2 * mean) - Em = 20log(gc_p) + Ei - Em
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 *
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 *    and from above:
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 *
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 *        gc_p = 10^((Ep - Ei + Em) / 20)
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 *
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 *    Ep is predicted using past energies and prediction coefficients:
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 *
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 *        Ep = sum(i,0,M){ ma_prediction_coeff[i] * quant_energy[i] }
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 *
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 *    gc_p in fixed-point arithmetic is calculated as following:
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 *
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 *        mean = 1/N * sum(i,0,N){ (fc_v[i] / 2^13) * (fc_v[i] / 2^13) } =
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 *        = 1/N * sum(i,0,N) { fc_v[i] * fc_v[i] } / 2^26
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 *
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 *        Ei = 10log(mean) = -10log(N) - 10log(2^26) +
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 *        + 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
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 *
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 *        Ep - Ei + Em = Ep + Em + 10log(N) + 10log(2^26) -
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 *        - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] }) =
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 *        = Ep + mr_energy - 10log(sum(i,0,N) { fc_v[i] * fc_v[i] })
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 *
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 *        gc_p = 10 ^ ((Ep - Ei + Em) / 20) =
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 *        = 2 ^ (3.3219 * (Ep - Ei + Em) / 20) = 2 ^ (0.166 * (Ep - Ei + Em))
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 *
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 *    where
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 *
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 *        mr_energy = Em + 10log(N) + 10log(2^26)
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 *
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 * @remark The routine is used in G.729 and AMR (all modes).
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 */
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int16_t ff_acelp_decode_gain_code(
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    AudioDSPContext *adsp,
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    int gain_corr_factor,
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    const int16_t* fc_v,
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    int mr_energy,
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    const int16_t* quant_energy,
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    const int16_t* ma_prediction_coeff,
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    int subframe_size,
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    int max_pred_order);
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/**
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 * Calculate fixed gain (part of section 6.1.3 of AMR spec)
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 *
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 * @param fixed_gain_factor gain correction factor
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 * @param fixed_mean_energy mean decoded algebraic codebook vector energy
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 * @param prediction_error vector of the quantified predictor errors of
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 *        the four previous subframes. It is updated by this function.
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 * @param energy_mean desired mean innovation energy
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 * @param pred_table table of four moving average coefficients
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 */
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float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
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                            float *prediction_error, float energy_mean,
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                            const float *pred_table);
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/**
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 * Decode the adaptive codebook index to the integer and fractional parts
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 * of the pitch lag for one subframe at 1/3 fractional precision.
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 *
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 * The choice of pitch lag is described in 3GPP TS 26.090 section 5.6.1.
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 *
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 * @param lag_int             integer part of pitch lag of the current subframe
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 * @param lag_frac            fractional part of pitch lag of the current subframe
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 * @param pitch_index         parsed adaptive codebook (pitch) index
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 * @param prev_lag_int        integer part of pitch lag for the previous subframe
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 * @param subframe            current subframe number
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 * @param third_as_first      treat the third frame the same way as the first
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 */
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void ff_decode_pitch_lag(int *lag_int, int *lag_frac, int pitch_index,
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                         const int prev_lag_int, const int subframe,
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                         int third_as_first, int resolution);
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#endif /* AVCODEC_ACELP_PITCH_DELAY_H */