Line data Source code
1 : /******************************************************************************************************
2 :
3 : (C) 2022-2025 IVAS codec Public Collaboration with portions copyright Dolby International AB, Ericsson AB,
4 : Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V., Huawei Technologies Co. LTD.,
5 : Koninklijke Philips N.V., Nippon Telegraph and Telephone Corporation, Nokia Technologies Oy, Orange,
6 : Panasonic Holdings Corporation, Qualcomm Technologies, Inc., VoiceAge Corporation, and other
7 : contributors to this repository. All Rights Reserved.
8 :
9 : This software is protected by copyright law and by international treaties.
10 : The IVAS codec Public Collaboration consisting of Dolby International AB, Ericsson AB,
11 : Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V., Huawei Technologies Co. LTD.,
12 : Koninklijke Philips N.V., Nippon Telegraph and Telephone Corporation, Nokia Technologies Oy, Orange,
13 : Panasonic Holdings Corporation, Qualcomm Technologies, Inc., VoiceAge Corporation, and other
14 : contributors to this repository retain full ownership rights in their respective contributions in
15 : the software. This notice grants no license of any kind, including but not limited to patent
16 : license, nor is any license granted by implication, estoppel or otherwise.
17 :
18 : Contributors are required to enter into the IVAS codec Public Collaboration agreement before making
19 : contributions.
20 :
21 : This software is provided "AS IS", without any express or implied warranties. The software is in the
22 : development stage. It is intended exclusively for experts who have experience with such software and
23 : solely for the purpose of inspection. All implied warranties of non-infringement, merchantability
24 : and fitness for a particular purpose are hereby disclaimed and excluded.
25 :
26 : Any dispute, controversy or claim arising under or in relation to providing this software shall be
27 : submitted to and settled by the final, binding jurisdiction of the courts of Munich, Germany in
28 : accordance with the laws of the Federal Republic of Germany excluding its conflict of law rules and
29 : the United Nations Convention on Contracts on the International Sales of Goods.
30 :
31 : *******************************************************************************************************/
32 :
33 : /*====================================================================================
34 : EVS Codec 3GPP TS26.443 Nov 04, 2021. Version 12.14.0 / 13.10.0 / 14.6.0 / 15.4.0 / 16.3.0
35 : ====================================================================================*/
36 :
37 : #include <stdint.h>
38 : #include "options.h"
39 : #ifdef DEBUGGING
40 : #include "debug.h"
41 : #endif
42 : #include "cnst.h"
43 : #include "rom_com.h"
44 : #include "prot.h"
45 : #include "wmc_auto.h"
46 :
47 : /*-------------------------------------------------------------------*
48 : * analy_lp()
49 : *
50 : * Perform LP analysis
51 : *
52 : * - autocorrelations + lag windowing
53 : * - Levinson-Durbin algorithm to find A(z)
54 : * - convert A(z) to LSPs
55 : * - find interpolated LSPs and convert back to A(z) for all subframes
56 : * - update LSPs for the next frame
57 : *-------------------------------------------------------------------*/
58 :
59 1319689 : void analy_lp(
60 : const float speech[], /* i : pointer to the speech frame */
61 : const int16_t L_frame, /* i : length of the frame */
62 : const int16_t L_look, /* i : look-ahead */
63 : float *ener, /* o : residual energy from Levinson-Durbin */
64 : float A[], /* o : A(z) filter coefficients */
65 : float epsP[], /* o : LP analysis residual energies for each iteration */
66 : float lsp_new[], /* o : current frame LSPs */
67 : float lsp_mid[], /* o : current mid-frame LSPs */
68 : float lsp_old[], /* i/o: previous frame unquantized LSPs */
69 : const int16_t Top[2], /* i : open loop pitch lag */
70 : const float Tnc[2], /* i : open loop pitch gain */
71 : const int32_t sr_core, /* i : internal sampling rate */
72 : const int16_t sec_chan_low_rate /* i : TD secondary channel flag */
73 : )
74 : {
75 : int16_t i_subfr, wind_length, half_frame;
76 : float r[M + 1], *lsp;
77 : const float *wind, *pt;
78 : int16_t half_frame_idx;
79 :
80 1319689 : if ( L_frame == L_FRAME )
81 : {
82 1153834 : wind_length = L_LP;
83 1153834 : wind = LP_assym_window;
84 : }
85 : else /* L_frame == L_FRAME16k */
86 : {
87 165855 : wind_length = L_LP_16k;
88 165855 : wind = LP_assym_window_16k;
89 : }
90 1319689 : lsp = lsp_mid;
91 1319689 : half_frame = L_frame >> 1;
92 :
93 1319689 : half_frame_idx = 0;
94 3959067 : for ( i_subfr = half_frame; i_subfr <= L_frame; i_subfr = i_subfr + half_frame )
95 : {
96 2639378 : pt = speech + i_subfr + L_look - wind_length;
97 :
98 : /* Autocorrelations */
99 2639378 : autocorr( pt, r, M, wind_length, wind, 0, 0, 0 );
100 :
101 : /* Lag windowing */
102 2639378 : adapt_lag_wind( r, M, Top[half_frame_idx], Tnc[half_frame_idx], sr_core );
103 2639378 : ++half_frame_idx;
104 :
105 : /* Levinson-Durbin */
106 2639378 : lev_dur( A, r, M, epsP );
107 :
108 : /* Conversion of A(z) to LSPs */
109 2639378 : a2lsp_stab( A, lsp, lsp_old );
110 :
111 2639378 : lsp = lsp_new;
112 : }
113 :
114 1319689 : if ( sec_chan_low_rate == 1 )
115 : {
116 0 : int_lsp4( L_frame, lsp_old, lsp_mid, lsp_new, A, M, -2 );
117 : }
118 : else
119 : {
120 : /* LSP interpolation */
121 1319689 : int_lsp4( L_frame, lsp_old, lsp_mid, lsp_new, A, M, 0 );
122 : }
123 :
124 : /* updates */
125 1319689 : mvr2r( lsp_new, lsp_old, M );
126 :
127 1319689 : *ener = epsP[M];
128 :
129 1319689 : return;
130 : }
131 :
132 :
133 : /*-------------------------------------------------------------------*
134 : * analy_lp_AMR_WB()
135 : *
136 : * Perform LP analysis for AMR-WB IO mode
137 : *
138 : * - autocorrelations + lag windowing
139 : * - Levinson-Durbin algorithm to find A(z)
140 : * - convert A(z) to ISPs
141 : * - find interpolated ISPs and convert back to A(z) for all subframes
142 : * - update ISPs for the next frame
143 : *-------------------------------------------------------------------*/
144 :
145 0 : void analy_lp_AMR_WB(
146 : const float speech[], /* i : pointer to the speech frame */
147 : float *ener, /* o : residual energy from Levinson-Durbin */
148 : float A[], /* o : A(z) filter coefficients */
149 : float epsP[], /* o : LP analysis residual energies for each iteration */
150 : float isp_new[], /* o : current frame ISPs */
151 : float isp_old[], /* i/o: previous frame unquantized ISPs */
152 : float isf_new[], /* o : current frame ISFs */
153 : const int16_t Top, /* i : open loop pitch lag */
154 : const float Tnc /* i : open loop pitch gain */
155 : )
156 : {
157 : int16_t wind_length;
158 : float r[M + 1];
159 : const float *wind;
160 :
161 : /* Initialization */
162 0 : wind_length = L_LP_AMR_WB;
163 0 : wind = hamcos_window;
164 :
165 : /* Autocorrelations */
166 0 : autocorr( speech - L_SUBFR, r, M, wind_length, wind, 0, 0, 0 );
167 :
168 : /* Lag windowing */
169 0 : adapt_lag_wind( r, M, Top, Tnc, 12800 );
170 :
171 : /* Levinson-Durbin */
172 0 : lev_dur( A, r, M, epsP );
173 :
174 0 : a2isf( A, isf_new, stable_ISF, M );
175 0 : isf2isp( isf_new, isp_new, M, INT_FS_12k8 );
176 :
177 : /* ISP interpolation */
178 0 : int_lsp( L_FRAME, isp_old, isp_new, A, M, interpol_isp_amr_wb, 1 );
179 :
180 0 : *ener = epsP[M];
181 :
182 : /* updates */
183 0 : mvr2r( isp_new, isp_old, M );
184 :
185 0 : return;
186 : }
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