ffmpeg / libavcodec / amrnbdec.c @ 72415b2a
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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 libavcodec/amrnbdec.c

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* AMR narrowband decoder

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*

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* This decoder uses floats for simplicity and so is not 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 "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 = SAMPLE_FMT_FLT; 
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// p>excitation always points to the same position in p>excitation_buf

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p>excitation = &p>excitation_buf[PITCH_DELAY_MAX + LP_FILTER_ORDER + 1];

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for (i = 0; i < LP_FILTER_ORDER; i++) { 
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p>prev_lsp_sub4[i] = lsp_sub4_init[i] * 1000 / (float)(1 << 15); 
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p>lsf_avg[i] = p>lsf_q[3][i] = lsp_avg_init[i] / (float)(1 << 15); 
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} 
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for (i = 0; i < 4; i++) 
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p>prediction_error[i] = MIN_ENERGY; 
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return 0; 
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} 
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/**

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* Unpack an RFC4867 speech frame into the AMR frame mode and parameters.

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*

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* The order of speech bits is specified by 3GPP TS 26.101.

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*

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* @param p the context

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* @param buf pointer to the input buffer

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* @param buf_size size of the input buffer

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*

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* @return the frame mode

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*/

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static enum Mode unpack_bitstream(AMRContext *p, const uint8_t *buf, 
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int buf_size)

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{ 
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GetBitContext gb; 
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enum Mode mode;

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init_get_bits(&gb, buf, buf_size * 8);

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// Decode the first octet.

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skip_bits(&gb, 1); // padding bit 
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mode = get_bits(&gb, 4); // frame type 
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p>bad_frame_indicator = !get_bits1(&gb); // quality bit

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skip_bits(&gb, 2); // two padding bits 
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if (mode <= MODE_DTX) {

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uint16_t *data = (uint16_t *)&p>frame; 
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const uint8_t *order = amr_unpacking_bitmaps_per_mode[mode];

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int field_size;

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memset(&p>frame, 0, sizeof(AMRNBFrame)); 
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buf++; 
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while ((field_size = *order++)) {

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int field = 0; 
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int field_offset = *order++;

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while (field_size) {

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int bit = *order++;

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field <<= 1;

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field = buf[bit >> 3] >> (bit & 7) & 1; 
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} 
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data[field_offset] = field; 
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} 
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} 
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return mode;

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} 
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/// @defgroup amr_lpc_decoding AMR pitch LPC coefficient decoding functions

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/// @{

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/**

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* Convert an lsf vector into an lsp vector.

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*

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* @param lsf input lsf vector

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* @param lsp output lsp vector

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*/

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static void lsf2lsp(const float *lsf, double *lsp) 
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{ 
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int i;

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for (i = 0; i < LP_FILTER_ORDER; i++) 
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lsp[i] = cos(2.0 * M_PI * lsf[i]); 
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} 
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/**

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* Interpolate the LSF vector (used for fixed gain smoothing).

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* The interpolation is done over all four subframes even in MODE_12k2.

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*

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* @param[in,out] lsf_q LSFs in [0,1] for each subframe

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* @param[in] lsf_new New LSFs in [0,1] for subframe 4

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*/

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static void interpolate_lsf(float lsf_q[4][LP_FILTER_ORDER], float *lsf_new) 
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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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lsf2lsp(lsf_q, lsp); 
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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)];

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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]];

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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); 
358 
memcpy(p>prev_lsf_r, lsf_r, LP_FILTER_ORDER * sizeof(*lsf_r));

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

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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] +

366 
(p>lsp[3][j]  p>prev_lsp_sub4[j]) * 0.25 * i; 
367 
} 
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/// @}

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/// @defgroup amr_pitch_vector_decoding AMR pitch vector decoding functions

373 
/// @{

374  
375 
/**

376 
* Like ff_decode_pitch_lag(), but with 1/6 resolution

377 
*/

378 
static void decode_pitch_lag_1_6(int *lag_int, int *lag_frac, int pitch_index, 
379 
const int prev_lag_int, const int subframe) 
380 
{ 
381 
if (subframe == 0  subframe == 2) { 
382 
if (pitch_index < 463) { 
383 
*lag_int = (pitch_index + 107) * 10923 >> 16; 
384 
*lag_frac = pitch_index  *lag_int * 6 + 105; 
385 
} else {

386 
*lag_int = pitch_index  368;

387 
*lag_frac = 0;

388 
} 
389 
} else {

390 
*lag_int = ((pitch_index + 5) * 10923 >> 16)  1; 
391 
*lag_frac = pitch_index  *lag_int * 6  3; 
392 
*lag_int += av_clip(prev_lag_int  5, PITCH_LAG_MIN_MODE_12k2,

393 
PITCH_DELAY_MAX  9);

394 
} 
395 
} 
396  
397 
static void decode_pitch_vector(AMRContext *p, 
398 
const AMRNBSubframe *amr_subframe,

399 
const int subframe) 
400 
{ 
401 
int pitch_lag_int, pitch_lag_frac;

402 
enum Mode mode = p>cur_frame_mode;

403  
404 
if (p>cur_frame_mode == MODE_12k2) {

405 
decode_pitch_lag_1_6(&pitch_lag_int, &pitch_lag_frac, 
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amr_subframe>p_lag, p>pitch_lag_int, 
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subframe); 
408 
} else

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ff_decode_pitch_lag(&pitch_lag_int, &pitch_lag_frac, 
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amr_subframe>p_lag, 
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p>pitch_lag_int, subframe, 
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mode != MODE_4k75 && mode != MODE_5k15, 
413 
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

416  
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pitch_lag_frac <<= (p>cur_frame_mode != MODE_12k2); 
418  
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pitch_lag_int += pitch_lag_frac > 0;

420  
421 
/* Calculate the pitch vector by interpolating the past excitation at the

422 
pitch lag using a b60 hamming windowed sinc function. */

423 
ff_acelp_interpolatef(p>excitation, p>excitation + 1  pitch_lag_int,

424 
ff_b60_sinc, 6,

425 
pitch_lag_frac + 6  6*(pitch_lag_frac > 0), 
426 
10, AMR_SUBFRAME_SIZE);

427  
428 
memcpy(p>pitch_vector, p>excitation, AMR_SUBFRAME_SIZE * sizeof(float)); 
429 
} 
430  
431 
/// @}

432  
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434 
/// @defgroup amr_algebraic_code_book AMR algebraic code book (fixed) vector decoding functions

435 
/// @{

436  
437 
/**

438 
* Decode a 10bit algebraic codebook index from a 10.2 kbit/s frame.

439 
*/

440 
static void decode_10bit_pulse(int code, int pulse_position[8], 
441 
int i1, int i2, int i3) 
442 
{ 
443 
// coded using 7+3 bits with the 3 LSBs being, individually, the LSB of 1 of

444 
// the 3 pulses and the upper 7 bits being coded in base 5

445 
const uint8_t *positions = base_five_table[code >> 3]; 
446 
pulse_position[i1] = (positions[2] << 1) + ( code & 1); 
447 
pulse_position[i2] = (positions[1] << 1) + ((code >> 1) & 1); 
448 
pulse_position[i3] = (positions[0] << 1) + ((code >> 2) & 1); 
449 
} 
450  
451 
/**

452 
* Decode the algebraic codebook index to pulse positions and signs and

453 
* construct the algebraic codebook vector for MODE_10k2.

454 
*

455 
* @param fixed_index positions of the eight pulses

456 
* @param fixed_sparse pointer to the algebraic codebook vector

457 
*/

458 
static void decode_8_pulses_31bits(const int16_t *fixed_index, 
459 
AMRFixed *fixed_sparse) 
460 
{ 
461 
int pulse_position[8]; 
462 
int i, temp;

463  
464 
decode_10bit_pulse(fixed_index[4], pulse_position, 0, 4, 1); 
465 
decode_10bit_pulse(fixed_index[5], pulse_position, 2, 6, 5); 
466  
467 
// coded using 5+2 bits with the 2 LSBs being, individually, the LSB of 1 of

468 
// the 2 pulses and the upper 5 bits being coded in base 5

469 
temp = ((fixed_index[6] >> 2) * 25 + 12) >> 5; 
470 
pulse_position[3] = temp % 5; 
471 
pulse_position[7] = temp / 5; 
472 
if (pulse_position[7] & 1) 
473 
pulse_position[3] = 4  pulse_position[3]; 
474 
pulse_position[3] = (pulse_position[3] << 1) + ( fixed_index[6] & 1); 
475 
pulse_position[7] = (pulse_position[7] << 1) + ((fixed_index[6] >> 1) & 1); 
476  
477 
fixed_sparse>n = 8;

478 
for (i = 0; i < 4; i++) { 
479 
const int pos1 = (pulse_position[i] << 2) + i; 
480 
const int pos2 = (pulse_position[i + 4] << 2) + i; 
481 
const float sign = fixed_index[i] ? 1.0 : 1.0; 
482 
fixed_sparse>x[i ] = pos1; 
483 
fixed_sparse>x[i + 4] = pos2;

484 
fixed_sparse>y[i ] = sign; 
485 
fixed_sparse>y[i + 4] = pos2 < pos1 ? sign : sign;

486 
} 
487 
} 
488  
489 
/**

490 
* Decode the algebraic codebook index to pulse positions and signs,

491 
* then construct the algebraic codebook vector.

492 
*

493 
* nb of pulses  bits encoding pulses

494 
* For MODE_4k75 or MODE_5k15, 2  13, 46, 7

495 
* MODE_5k9, 2  1, 24, 56, 79

496 
* MODE_6k7, 3  13, 4, 57, 8, 911

497 
* MODE_7k4 or MODE_7k95, 4  13, 46, 79, 10, 1113

498 
*

499 
* @param fixed_sparse pointer to the algebraic codebook vector

500 
* @param pulses algebraic codebook indexes

501 
* @param mode mode of the current frame

502 
* @param subframe current subframe number

503 
*/

504 
static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses, 
505 
const enum Mode mode, const int subframe) 
506 
{ 
507 
assert(MODE_4k75 <= mode && mode <= MODE_12k2); 
508  
509 
if (mode == MODE_12k2) {

510 
ff_decode_10_pulses_35bits(pulses, fixed_sparse, gray_decode, 5, 3); 
511 
} else if (mode == MODE_10k2) { 
512 
decode_8_pulses_31bits(pulses, fixed_sparse); 
513 
} else {

514 
int *pulse_position = fixed_sparse>x;

515 
int i, pulse_subset;

516 
const int fixed_index = pulses[0]; 
517  
518 
if (mode <= MODE_5k15) {

519 
pulse_subset = ((fixed_index >> 3) & 8) + (subframe << 1); 
520 
pulse_position[0] = ( fixed_index & 7) * 5 + track_position[pulse_subset]; 
521 
pulse_position[1] = ((fixed_index >> 3) & 7) * 5 + track_position[pulse_subset + 1]; 
522 
fixed_sparse>n = 2;

523 
} else if (mode == MODE_5k9) { 
524 
pulse_subset = ((fixed_index & 1) << 1) + 1; 
525 
pulse_position[0] = ((fixed_index >> 1) & 7) * 5 + pulse_subset; 
526 
pulse_subset = (fixed_index >> 4) & 3; 
527 
pulse_position[1] = ((fixed_index >> 6) & 7) * 5 + pulse_subset + (pulse_subset == 3 ? 1 : 0); 
528 
fixed_sparse>n = pulse_position[0] == pulse_position[1] ? 1 : 2; 
529 
} else if (mode == MODE_6k7) { 
530 
pulse_position[0] = (fixed_index & 7) * 5; 
531 
pulse_subset = (fixed_index >> 2) & 2; 
532 
pulse_position[1] = ((fixed_index >> 4) & 7) * 5 + pulse_subset + 1; 
533 
pulse_subset = (fixed_index >> 6) & 2; 
534 
pulse_position[2] = ((fixed_index >> 8) & 7) * 5 + pulse_subset + 2; 
535 
fixed_sparse>n = 3;

536 
} else { // mode <= MODE_7k95 
537 
pulse_position[0] = gray_decode[ fixed_index & 7]; 
538 
pulse_position[1] = gray_decode[(fixed_index >> 3) & 7] + 1; 
539 
pulse_position[2] = gray_decode[(fixed_index >> 6) & 7] + 2; 
540 
pulse_subset = (fixed_index >> 9) & 1; 
541 
pulse_position[3] = gray_decode[(fixed_index >> 10) & 7] + pulse_subset + 3; 
542 
fixed_sparse>n = 4;

543 
} 
544 
for (i = 0; i < fixed_sparse>n; i++) 
545 
fixed_sparse>y[i] = (pulses[1] >> i) & 1 ? 1.0 : 1.0; 
546 
} 
547 
} 
548  
549 
/**

550 
* Apply pitch lag to obtain the sharpened fixed vector (section 6.1.2)

551 
*

552 
* @param p the context

553 
* @param subframe unpacked amr subframe

554 
* @param mode mode of the current frame

555 
* @param fixed_sparse sparse respresentation of the fixed vector

556 
*/

557 
static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode, 
558 
AMRFixed *fixed_sparse) 
559 
{ 
560 
// The spec suggests the current pitch gain is always used, but in other

561 
// modes the pitch and codebook gains are joinly quantized (sec 5.8.2)

562 
// so the codebook gain cannot depend on the quantized pitch gain.

563 
if (mode == MODE_12k2)

564 
p>beta = FFMIN(p>pitch_gain[4], 1.0); 
565  
566 
fixed_sparse>pitch_lag = p>pitch_lag_int; 
567 
fixed_sparse>pitch_fac = p>beta; 
568  
569 
// Save pitch sharpening factor for the next subframe

570 
// MODE_4k75 only updates on the 2nd and 4th subframes  this follows from

571 
// the fact that the gains for two subframes are jointly quantized.

572 
if (mode != MODE_4k75  subframe & 1) 
573 
p>beta = av_clipf(p>pitch_gain[4], 0.0, SHARP_MAX); 
574 
} 
575 
/// @}

576  
577  
578 
/// @defgroup amr_gain_decoding AMR gain decoding functions

579 
/// @{

580  
581 
/**

582 
* fixed gain smoothing

583 
* Note that where the spec specifies the "spectrum in the q domain"

584 
* in section 6.1.4, in fact frequencies should be used.

585 
*

586 
* @param p the context

587 
* @param lsf LSFs for the current subframe, in the range [0,1]

588 
* @param lsf_avg averaged LSFs

589 
* @param mode mode of the current frame

590 
*

591 
* @return fixed gain smoothed

592 
*/

593 
static float fixed_gain_smooth(AMRContext *p , const float *lsf, 
594 
const float *lsf_avg, const enum Mode mode) 
595 
{ 
596 
float diff = 0.0; 
597 
int i;

598  
599 
for (i = 0; i < LP_FILTER_ORDER; i++) 
600 
diff += fabs(lsf_avg[i]  lsf[i]) / lsf_avg[i]; 
601  
602 
// If diff is large for ten subframes, disable smoothing for a 40subframe

603 
// hangover period.

604 
p>diff_count++; 
605 
if (diff <= 0.65) 
606 
p>diff_count = 0;

607  
608 
if (p>diff_count > 10) { 
609 
p>hang_count = 0;

610 
p>diff_count; // don't let diff_count overflow

611 
} 
612  
613 
if (p>hang_count < 40) { 
614 
p>hang_count++; 
615 
} else if (mode < MODE_7k4  mode == MODE_10k2) { 
616 
const float smoothing_factor = av_clipf(4.0 * diff  1.6, 0.0, 1.0); 
617 
const float fixed_gain_mean = (p>fixed_gain[0] + p>fixed_gain[1] + 
618 
p>fixed_gain[2] + p>fixed_gain[3] + 
619 
p>fixed_gain[4]) * 0.2; 
620 
return smoothing_factor * p>fixed_gain[4] + 
621 
(1.0  smoothing_factor) * fixed_gain_mean; 
622 
} 
623 
return p>fixed_gain[4]; 
624 
} 
625  
626 
/**

627 
* Decode pitch gain and fixed gain factor (part of section 6.1.3).

628 
*

629 
* @param p the context

630 
* @param amr_subframe unpacked amr subframe

631 
* @param mode mode of the current frame

632 
* @param subframe current subframe number

633 
* @param fixed_gain_factor decoded gain correction factor

634 
*/

635 
static void decode_gains(AMRContext *p, const AMRNBSubframe *amr_subframe, 
636 
const enum Mode mode, const int subframe, 
637 
float *fixed_gain_factor)

638 
{ 
639 
if (mode == MODE_12k2  mode == MODE_7k95) {

640 
p>pitch_gain[4] = qua_gain_pit [amr_subframe>p_gain ]

641 
* (1.0 / 16384.0); 
642 
*fixed_gain_factor = qua_gain_code[amr_subframe>fixed_gain] 
643 
* (1.0 / 2048.0); 
644 
} else {

645 
const uint16_t *gains;

646  
647 
if (mode >= MODE_6k7) {

648 
gains = gains_high[amr_subframe>p_gain]; 
649 
} else if (mode >= MODE_5k15) { 
650 
gains = gains_low [amr_subframe>p_gain]; 
651 
} else {

652 
// gain index is only coded in subframes 0,2 for MODE_4k75

653 
gains = gains_MODE_4k75[(p>frame.subframe[subframe & 2].p_gain << 1) + (subframe & 1)]; 
654 
} 
655  
656 
p>pitch_gain[4] = gains[0] * (1.0 / 16384.0); 
657 
*fixed_gain_factor = gains[1] * (1.0 / 4096.0); 
658 
} 
659 
} 
660  
661 
/// @}

662  
663  
664 
/// @defgroup amr_pre_processing AMR preprocessing functions

665 
/// @{

666  
667 
/**

668 
* Circularly convolve a sparse fixed vector with a phase dispersion impulse

669 
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).

670 
*

671 
* @param out vector with filter applied

672 
* @param in source vector

673 
* @param filter phase filter coefficients

674 
*

675 
* out[n] = sum(i,0,len1){ in[i] * filter[(len + n  i)%len] }

676 
*/

677 
static void apply_ir_filter(float *out, const AMRFixed *in, 
678 
const float *filter) 
679 
{ 
680 
float filter1[AMR_SUBFRAME_SIZE], //!< filters at pitch lag*1 and *2 
681 
filter2[AMR_SUBFRAME_SIZE]; 
682 
int lag = in>pitch_lag;

683 
float fac = in>pitch_fac;

684 
int i;

685  
686 
if (lag < AMR_SUBFRAME_SIZE) {

687 
ff_celp_circ_addf(filter1, filter, filter, lag, fac, 
688 
AMR_SUBFRAME_SIZE); 
689  
690 
if (lag < AMR_SUBFRAME_SIZE >> 1) 
691 
ff_celp_circ_addf(filter2, filter, filter1, lag, fac, 
692 
AMR_SUBFRAME_SIZE); 
693 
} 
694  
695 
memset(out, 0, sizeof(float) * AMR_SUBFRAME_SIZE); 
696 
for (i = 0; i < in>n; i++) { 
697 
int x = in>x[i];

698 
float y = in>y[i];

699 
const float *filterp; 
700  
701 
if (x >= AMR_SUBFRAME_SIZE  lag) {

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

706 
filterp = filter2; 
707  
708 
ff_celp_circ_addf(out, out, filterp, x, y, AMR_SUBFRAME_SIZE); 
709 
} 
710 
} 
711  
712 
/**

713 
* Reduce fixed vector sparseness by smoothing with one of three IR filters.

714 
* Also know as "adaptive phase dispersion".

715 
*

716 
* This implements 3GPP TS 26.090 section 6.1(5).

717 
*

718 
* @param p the context

719 
* @param fixed_sparse algebraic codebook vector

720 
* @param fixed_vector unfiltered fixed vector

721 
* @param fixed_gain smoothed gain

722 
* @param out space for modified vector if necessary

723 
*/

724 
static const float *anti_sparseness(AMRContext *p, AMRFixed *fixed_sparse, 
725 
const float *fixed_vector, 
726 
float fixed_gain, float *out) 
727 
{ 
728 
int ir_filter_nr;

729  
730 
if (p>pitch_gain[4] < 0.6) { 
731 
ir_filter_nr = 0; // strong filtering 
732 
} else if (p>pitch_gain[4] < 0.9) { 
733 
ir_filter_nr = 1; // medium filtering 
734 
} else

735 
ir_filter_nr = 2; // no filtering 
736  
737 
// detect 'onset'

738 
if (fixed_gain > 2.0 * p>prev_sparse_fixed_gain) { 
739 
p>ir_filter_onset = 2;

740 
} else if (p>ir_filter_onset) 
741 
p>ir_filter_onset; 
742  
743 
if (!p>ir_filter_onset) {

744 
int i, count = 0; 
745  
746 
for (i = 0; i < 5; i++) 
747 
if (p>pitch_gain[i] < 0.6) 
748 
count++; 
749 
if (count > 2) 
750 
ir_filter_nr = 0;

751  
752 
if (ir_filter_nr > p>prev_ir_filter_nr + 1) 
753 
ir_filter_nr; 
754 
} else if (ir_filter_nr < 2) 
755 
ir_filter_nr++; 
756  
757 
// Disable filtering for very low level of fixed_gain.

758 
// Note this step is not specified in the technical description but is in

759 
// the reference source in the function Ph_disp.

760 
if (fixed_gain < 5.0) 
761 
ir_filter_nr = 2;

762  
763 
if (p>cur_frame_mode != MODE_7k4 && p>cur_frame_mode < MODE_10k2

764 
&& ir_filter_nr < 2) {

765 
apply_ir_filter(out, fixed_sparse, 
766 
(p>cur_frame_mode == MODE_7k95 ? 
767 
ir_filters_lookup_MODE_7k95 : 
768 
ir_filters_lookup)[ir_filter_nr]); 
769 
fixed_vector = out; 
770 
} 
771  
772 
// update ir filter strength history

773 
p>prev_ir_filter_nr = ir_filter_nr; 
774 
p>prev_sparse_fixed_gain = fixed_gain; 
775  
776 
return fixed_vector;

777 
} 
778  
779 
/// @}

780  
781  
782 
/// @defgroup amr_synthesis AMR synthesis functions

783 
/// @{

784  
785 
/**

786 
* Conduct 10th order linear predictive coding synthesis.

787 
*

788 
* @param p pointer to the AMRContext

789 
* @param lpc pointer to the LPC coefficients

790 
* @param fixed_gain fixed codebook gain for synthesis

791 
* @param fixed_vector algebraic codebook vector

792 
* @param samples pointer to the output speech samples

793 
* @param overflow 16bit overflow flag

794 
*/

795 
static int synthesis(AMRContext *p, float *lpc, 
796 
float fixed_gain, const float *fixed_vector, 
797 
float *samples, uint8_t overflow)

798 
{ 
799 
int i, overflow_temp = 0; 
800 
float excitation[AMR_SUBFRAME_SIZE];

801  
802 
// if an overflow has been detected, the pitch vector is scaled down by a

803 
// factor of 4

804 
if (overflow)

805 
for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 
806 
p>pitch_vector[i] *= 0.25; 
807  
808 
ff_weighted_vector_sumf(excitation, p>pitch_vector, fixed_vector, 
809 
p>pitch_gain[4], fixed_gain, AMR_SUBFRAME_SIZE);

810  
811 
// emphasize pitch vector contribution

812 
if (p>pitch_gain[4] > 0.5 && !overflow) { 
813 
float energy = ff_dot_productf(excitation, excitation,

814 
AMR_SUBFRAME_SIZE); 
815 
float pitch_factor =

816 
p>pitch_gain[4] *

817 
(p>cur_frame_mode == MODE_12k2 ? 
818 
0.25 * FFMIN(p>pitch_gain[4], 1.0) : 
819 
0.5 * FFMIN(p>pitch_gain[4], SHARP_MAX)); 
820  
821 
for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 
822 
excitation[i] += pitch_factor * p>pitch_vector[i]; 
823  
824 
ff_scale_vector_to_given_sum_of_squares(excitation, excitation, energy, 
825 
AMR_SUBFRAME_SIZE); 
826 
} 
827  
828 
ff_celp_lp_synthesis_filterf(samples, lpc, excitation, AMR_SUBFRAME_SIZE, 
829 
LP_FILTER_ORDER); 
830  
831 
// detect overflow

832 
for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 
833 
if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {

834 
overflow_temp = 1;

835 
samples[i] = av_clipf(samples[i], AMR_SAMPLE_BOUND, 
836 
AMR_SAMPLE_BOUND); 
837 
} 
838  
839 
return overflow_temp;

840 
} 
841  
842 
/// @}

843  
844  
845 
/// @defgroup amr_update AMR update functions

846 
/// @{

847  
848 
/**

849 
* Update buffers and history at the end of decoding a subframe.

850 
*

851 
* @param p pointer to the AMRContext

852 
*/

853 
static void update_state(AMRContext *p) 
854 
{ 
855 
memcpy(p>prev_lsp_sub4, p>lsp[3], LP_FILTER_ORDER * sizeof(p>lsp[3][0])); 
856  
857 
memmove(&p>excitation_buf[0], &p>excitation_buf[AMR_SUBFRAME_SIZE],

858 
(PITCH_DELAY_MAX + LP_FILTER_ORDER + 1) * sizeof(float)); 
859  
860 
memmove(&p>pitch_gain[0], &p>pitch_gain[1], 4 * sizeof(float)); 
861 
memmove(&p>fixed_gain[0], &p>fixed_gain[1], 4 * sizeof(float)); 
862  
863 
memmove(&p>samples_in[0], &p>samples_in[AMR_SUBFRAME_SIZE],

864 
LP_FILTER_ORDER * sizeof(float)); 
865 
} 
866  
867 
/// @}

868  
869  
870 
/// @defgroup amr_postproc AMR Post processing functions

871 
/// @{

872  
873 
/**

874 
* Get the tilt factor of a formant filter from its transfer function

875 
*

876 
* @param lpc_n LP_FILTER_ORDER coefficients of the numerator

877 
* @param lpc_d LP_FILTER_ORDER coefficients of the denominator

878 
*/

879 
static float tilt_factor(float *lpc_n, float *lpc_d) 
880 
{ 
881 
float rh0, rh1; // autocorrelation at lag 0 and 1 
882  
883 
// LP_FILTER_ORDER prior zeros are needed for ff_celp_lp_synthesis_filterf

884 
float impulse_buffer[LP_FILTER_ORDER + AMR_TILT_RESPONSE] = { 0 }; 
885 
float *hf = impulse_buffer + LP_FILTER_ORDER; // start of impulse response 
886  
887 
hf[0] = 1.0; 
888 
memcpy(hf + 1, lpc_n, sizeof(float) * LP_FILTER_ORDER); 
889 
ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE, 
890 
LP_FILTER_ORDER); 
891  
892 
rh0 = ff_dot_productf(hf, hf, AMR_TILT_RESPONSE); 
893 
rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE  1); 
894  
895 
// The spec only specifies this check for 12.2 and 10.2 kbit/s

896 
// modes. But in the ref source the tilt is always nonnegative.

897 
return rh1 >= 0.0 ? rh1 / rh0 * AMR_TILT_GAMMA_T : 0.0; 
898 
} 
899  
900 
/**

901 
* Perform adaptive postfiltering to enhance the quality of the speech.

902 
* See section 6.2.1.

903 
*

904 
* @param p pointer to the AMRContext

905 
* @param lpc interpolated LP coefficients for this subframe

906 
* @param buf_out output of the filter

907 
*/

908 
static void postfilter(AMRContext *p, float *lpc, float *buf_out) 
909 
{ 
910 
int i;

911 
float *samples = p>samples_in + LP_FILTER_ORDER; // Start of input 
912  
913 
float speech_gain = ff_dot_productf(samples, samples,

914 
AMR_SUBFRAME_SIZE); 
915  
916 
float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter 
917 
const float *gamma_n, *gamma_d; // Formant filter factor table 
918 
float lpc_n[LP_FILTER_ORDER], lpc_d[LP_FILTER_ORDER]; // Transfer function coefficients 
919  
920 
if (p>cur_frame_mode == MODE_12k2  p>cur_frame_mode == MODE_10k2) {

921 
gamma_n = ff_pow_0_7; 
922 
gamma_d = ff_pow_0_75; 
923 
} else {

924 
gamma_n = ff_pow_0_55; 
925 
gamma_d = ff_pow_0_7; 
926 
} 
927  
928 
for (i = 0; i < LP_FILTER_ORDER; i++) { 
929 
lpc_n[i] = lpc[i] * gamma_n[i]; 
930 
lpc_d[i] = lpc[i] * gamma_d[i]; 
931 
} 
932  
933 
memcpy(pole_out, p>postfilter_mem, sizeof(float) * LP_FILTER_ORDER); 
934 
ff_celp_lp_synthesis_filterf(pole_out + LP_FILTER_ORDER, lpc_d, samples, 
935 
AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); 
936 
memcpy(p>postfilter_mem, pole_out + AMR_SUBFRAME_SIZE, 
937 
sizeof(float) * LP_FILTER_ORDER); 
938  
939 
ff_celp_lp_zero_synthesis_filterf(buf_out, lpc_n, 
940 
pole_out + LP_FILTER_ORDER, 
941 
AMR_SUBFRAME_SIZE, LP_FILTER_ORDER); 
942  
943 
ff_tilt_compensation(&p>tilt_mem, tilt_factor(lpc_n, lpc_d), buf_out, 
944 
AMR_SUBFRAME_SIZE); 
945  
946 
ff_adaptive_gain_control(buf_out, speech_gain, AMR_SUBFRAME_SIZE, 
947 
AMR_AGC_ALPHA, &p>postfilter_agc); 
948 
} 
949  
950 
/// @}

951  
952 
static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size, 
953 
AVPacket *avpkt) 
954 
{ 
955  
956 
AMRContext *p = avctx>priv_data; // pointer to private data

957 
const uint8_t *buf = avpkt>data;

958 
int buf_size = avpkt>size;

959 
float *buf_out = data; // pointer to the output data buffer 
960 
int i, subframe;

961 
float fixed_gain_factor;

962 
AMRFixed fixed_sparse = {0}; // fixed vector up to antisparseness processing 
963 
float spare_vector[AMR_SUBFRAME_SIZE]; // extra stack space to hold result from antisparseness processing 
964 
float synth_fixed_gain; // the fixed gain that synthesis should use 
965 
const float *synth_fixed_vector; // pointer to the fixed vector that synthesis should use 
966  
967 
p>cur_frame_mode = unpack_bitstream(p, buf, buf_size); 
968 
if (p>cur_frame_mode == MODE_DTX) {

969 
av_log_missing_feature(avctx, "dtx mode", 1); 
970 
return 1; 
971 
} 
972  
973 
if (p>cur_frame_mode == MODE_12k2) {

974 
lsf2lsp_5(p); 
975 
} else

976 
lsf2lsp_3(p); 
977  
978 
for (i = 0; i < 4; i++) 
979 
ff_acelp_lspd2lpc(p>lsp[i], p>lpc[i], 5);

980  
981 
for (subframe = 0; subframe < 4; subframe++) { 
982 
const AMRNBSubframe *amr_subframe = &p>frame.subframe[subframe];

983  
984 
decode_pitch_vector(p, amr_subframe, subframe); 
985  
986 
decode_fixed_sparse(&fixed_sparse, amr_subframe>pulses, 
987 
p>cur_frame_mode, subframe); 
988  
989 
// The fixed gain (section 6.1.3) depends on the fixed vector

990 
// (section 6.1.2), but the fixed vector calculation uses

991 
// pitch sharpening based on the on the pitch gain (section 6.1.3).

992 
// So the correct order is: pitch gain, pitch sharpening, fixed gain.

993 
decode_gains(p, amr_subframe, p>cur_frame_mode, subframe, 
994 
&fixed_gain_factor); 
995  
996 
pitch_sharpening(p, subframe, p>cur_frame_mode, &fixed_sparse); 
997  
998 
ff_set_fixed_vector(p>fixed_vector, &fixed_sparse, 1.0, 
999 
AMR_SUBFRAME_SIZE); 
1000  
1001 
p>fixed_gain[4] =

1002 
ff_amr_set_fixed_gain(fixed_gain_factor, 
1003 
ff_dot_productf(p>fixed_vector, p>fixed_vector, 
1004 
AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE, 
1005 
p>prediction_error, 
1006 
energy_mean[p>cur_frame_mode], energy_pred_fac); 
1007  
1008 
// The excitation feedback is calculated without any processing such

1009 
// as fixed gain smoothing. This isn't mentioned in the specification.

1010 
for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 
1011 
p>excitation[i] *= p>pitch_gain[4];

1012 
ff_set_fixed_vector(p>excitation, &fixed_sparse, p>fixed_gain[4],

1013 
AMR_SUBFRAME_SIZE); 
1014  
1015 
// In the ref decoder, excitation is stored with no fractional bits.

1016 
// This step prevents buzz in silent periods. The ref encoder can

1017 
// emit long sequences with pitch factor greater than one. This

1018 
// creates unwanted feedback if the excitation vector is nonzero.

1019 
// (e.g. test sequence T19_795.COD in 3GPP TS 26.074)

1020 
for (i = 0; i < AMR_SUBFRAME_SIZE; i++) 
1021 
p>excitation[i] = truncf(p>excitation[i]); 
1022  
1023 
// Smooth fixed gain.

1024 
// The specification is ambiguous, but in the reference source, the

1025 
// smoothed value is NOT fed back into later fixed gain smoothing.

1026 
synth_fixed_gain = fixed_gain_smooth(p, p>lsf_q[subframe], 
1027 
p>lsf_avg, p>cur_frame_mode); 
1028  
1029 
synth_fixed_vector = anti_sparseness(p, &fixed_sparse, p>fixed_vector, 
1030 
synth_fixed_gain, spare_vector); 
1031  
1032 
if (synthesis(p, p>lpc[subframe], synth_fixed_gain,

1033 
synth_fixed_vector, &p>samples_in[LP_FILTER_ORDER], 0))

1034 
// overflow detected > rerun synthesis scaling pitch vector down

1035 
// by a factor of 4, skipping pitch vector contribution emphasis

1036 
// and adaptive gain control

1037 
synthesis(p, p>lpc[subframe], synth_fixed_gain, 
1038 
synth_fixed_vector, &p>samples_in[LP_FILTER_ORDER], 1);

1039  
1040 
postfilter(p, p>lpc[subframe], buf_out + subframe * AMR_SUBFRAME_SIZE); 
1041  
1042 
// update buffers and history

1043 
ff_clear_fixed_vector(p>fixed_vector, &fixed_sparse, AMR_SUBFRAME_SIZE); 
1044 
update_state(p); 
1045 
} 
1046  
1047 
ff_acelp_apply_order_2_transfer_function(buf_out, highpass_zeros, 
1048 
highpass_poles, highpass_gain, 
1049 
p>high_pass_mem, AMR_BLOCK_SIZE); 
1050  
1051 
for (i = 0; i < AMR_BLOCK_SIZE; i++) 
1052 
buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE, 
1053 
1.0, 32767.0 / 32768.0); 
1054  
1055 
/* Update averaged lsf vector (used for fixed gain smoothing).

1056 
*

1057 
* Note that lsf_avg should not incorporate the current frame's LSFs

1058 
* for fixed_gain_smooth.

1059 
* The specification has an incorrect formula: the reference decoder uses

1060 
* qbar(n1) rather than qbar(n) in section 6.1(4) equation 71. */

1061 
ff_weighted_vector_sumf(p>lsf_avg, p>lsf_avg, p>lsf_q[3],

1062 
0.84, 0.16, LP_FILTER_ORDER); 
1063  
1064 
/* report how many samples we got */

1065 
*data_size = AMR_BLOCK_SIZE * sizeof(float); 
1066  
1067 
/* return the amount of bytes consumed if everything was OK */

1068 
return frame_sizes_nb[p>cur_frame_mode] + 1; // +7 for rounding and +8 for TOC 
1069 
} 
1070  
1071  
1072 
AVCodec amrnb_decoder = { 
1073 
.name = "amrnb",

1074 
.type = AVMEDIA_TYPE_AUDIO, 
1075 
.id = CODEC_ID_AMR_NB, 
1076 
.priv_data_size = sizeof(AMRContext),

1077 
.init = amrnb_decode_init, 
1078 
.decode = amrnb_decode_frame, 
1079 
.long_name = NULL_IF_CONFIG_SMALL("Adaptive MultiRate NarrowBand"),

1080 
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_FLT,SAMPLE_FMT_NONE},

1081 
}; 