FreeCalypso > hg > gsm-codec-lib
view libtwamr/cod_amr.h @ 499:446a4a2955ba
libgsmhr1/typedefs.h: initial port
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Wed, 19 Jun 2024 00:50:29 +0000 |
parents | 93d6c6960a46 |
children |
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/* ***************************************************************************** * * GSM AMR-NB speech codec R98 Version 7.6.0 December 12, 2001 * R99 Version 3.3.0 * REL-4 Version 4.1.0 * ***************************************************************************** * * File : cod_amr.h * Purpose : Main encoder routine operating on a frame basis. * ***************************************************************************** */ #ifndef cod_amr_h #define cod_amr_h "$Id $" /* ***************************************************************************** * INCLUDE FILES ***************************************************************************** */ #include "tw_amr.h" #include "typedef.h" #include "cnst.h" #include "lpc.h" #include "lsp.h" #include "cl_ltp.h" #include "gain_q.h" #include "p_ol_wgh.h" #include "ton_stab.h" #include "vad.h" #include "dtx_enc.h" /* ***************************************************************************** * DEFINITION OF DATA TYPES ***************************************************************************** */ /*-----------------------------------------------------------* * Coder constant parameters (defined in "cnst.h") * *-----------------------------------------------------------* * L_WINDOW : LPC analysis window size. * * L_NEXT : Samples of next frame needed for autocor. * * L_FRAME : Frame size. * * L_FRAME_BY2 : Half the frame size. * * L_SUBFR : Sub-frame size. * * M : LPC order. * * MP1 : LPC order+1 * * L_TOTAL7k4 : Total size of speech buffer. * * PIT_MIN7k4 : Minimum pitch lag. * * PIT_MAX : Maximum pitch lag. * * L_INTERPOL : Length of filter for interpolation * *-----------------------------------------------------------*/ typedef struct { /* Speech vector */ Word16 old_speech[L_TOTAL]; Word16 *speech, *p_window, *p_window_12k2; Word16 *new_speech; /* Global variable */ /* Weight speech vector */ Word16 old_wsp[L_FRAME + PIT_MAX]; Word16 *wsp; /* OL LTP states */ Word16 old_lags[5]; Word16 ol_gain_flg[2]; /* Excitation vector */ Word16 old_exc[L_FRAME + PIT_MAX + L_INTERPOL]; Word16 *exc; /* Zero vector */ Word16 ai_zero[L_SUBFR + MP1]; Word16 *zero; /* Impulse response vector */ Word16 *h1; Word16 hvec[L_SUBFR * 2]; /* Substates */ lpcState lpcSt; lspState lspSt; clLtpState clLtpSt; gainQuantState gainQuantSt; pitchOLWghtState pitchOLWghtSt; tonStabState tonStabSt; vadState vadSt; Flag dtx; dtx_encState dtx_encSt; /* Filter's memory */ Word16 mem_syn[M], mem_w0[M], mem_w[M]; Word16 mem_err[M + L_SUBFR], *error; Word16 sharp; } cod_amrState; /* ******************************************************************************** * DECLARATION OF PROTOTYPES ******************************************************************************** */ /* ************************************************************************** * * Function : cod_amr_reset * Purpose : Resets state memory * Returns : 0 on success * ************************************************************************** */ void cod_amr_reset (cod_amrState *st, Flag dtx, Flag use_vad2); /*************************************************************************** * FUNCTION: cod_amr_first * * PURPOSE: Copes with look-ahead. * * INPUTS: * No input argument are passed to this function. However, before * calling this function, 40 new speech data should be copied to the * vector new_speech[]. This is a global pointer which is declared in * this file (it points to the end of speech buffer minus 200). * ***************************************************************************/ int cod_amr_first(cod_amrState *st, /* i/o : State struct */ Word16 new_speech[] /* i : speech input (L_FRAME) */ ); /*************************************************************************** * FUNCTION: cod_amr * * PURPOSE: Main encoder routine. * * DESCRIPTION: This function is called every 20 ms speech frame, * operating on the newly read 160 speech samples. It performs the * principle encoding functions to produce the set of encoded parameters * which include the LSP, adaptive codebook, and fixed codebook * quantization indices (addresses and gains). * * INPUTS: * No input argument are passed to this function. However, before * calling this function, 160 new speech data should be copied to the * vector new_speech[]. This is a global pointer which is declared in * this file (it points to the end of speech buffer minus 160). * * OUTPUTS: * * ana[]: vector of analysis parameters. * synth[]: Local synthesis speech (for debugging purposes) * ***************************************************************************/ int cod_amr(cod_amrState *st, /* i/o : State struct */ enum Mode mode, /* i : AMR mode */ Word16 new_speech[], /* i : speech input (L_FRAME) */ Word16 ana[], /* o : Analysis parameters */ enum Mode *usedMode, /* o : used mode */ Word16 synth[] /* o : Local synthesis */ ); #endif