ffmpeg / libavcodec / amrnbdec.c @ 5d6e4c16
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/*


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* AMR narrowband decoder

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* Copyright (c) 20062007 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 021101301 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 bitexact. 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 16bit 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 postfilter */

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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 antisparseness 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 highpass 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 splitmatrix 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) 
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{ 
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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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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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ff_acelp_lsf2lspd(lsp, lsf_q, LP_FILTER_ORDER); 
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} 
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/**

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* Decode a set of 5 splitmatrix quantized lsf indexes into 2 lsp vectors.

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*

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* @param p pointer to the AMRContext

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*/

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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]; 
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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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/**

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* Decode a set of 3 splitmatrix quantized lsf indexes into an lsp vector.

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*

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* @param p pointer to the AMRContext

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*/

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static void lsf2lsp_3(AMRContext *p) 
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{ 
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const uint16_t *lsf_param = p>frame.lsf;

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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)];

317 
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]];

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memcpy(lsf_r + 6, lsf_quantizer, 4 * sizeof(*lsf_r)); 
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// calculate meanremoved 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[i1][j] = p>prev_lsp_sub4[j] +

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(p>lsp[3][j]  p>prev_lsp_sub4[j]) * 0.25 * i; 
339 
} 
340  
341 
/// @}

342  
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/// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions

345 
/// @{

346  
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) { 
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*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;

375  
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)); 
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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  
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/* 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 
/// @}

404  
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406 
/// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions

407 
/// @{

408  
409 
/**

410 
* Decode a 10bit 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  13, 46, 7

467 
* MODE_5k9, 2  1, 24, 56, 79

468 
* MODE_6k7, 3  13, 4, 57, 8, 911

469 
* MODE_7k4 or MODE_7k95, 4  13, 46, 79, 10, 1113

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 40subframe

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 preprocessing 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,len1){ 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 16bit 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 nonnegative.

867 
return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0; 
868 
} 
869  
870 
/**

871 
* Perform adaptive postfiltering 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 antisparseness processing 
933 
float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from antisparseness 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(n1) 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 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 MultiRate NarrowBand"),

1047 
.sample_fmts = (enum AVSampleFormat[]){AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE},

1048 
}; 