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Working codec2 support
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Brian West
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/*---------------------------------------------------------------------------*\
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FILE........: codec2.c
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AUTHOR......: David Rowe
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DATE CREATED: 21/8/2010
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Codec2 fully quantised encoder and decoder functions. If you want use
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codec2, the codec2_xxx functions are for you.
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\*---------------------------------------------------------------------------*/
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/*
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Copyright (C) 2010 David Rowe
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All rights reserved.
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU Lesser General Public License version 2.1, as
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published by the Free Software Foundation. This program is
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distributed in the hope that it will be useful, but WITHOUT ANY
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WARRANTY; without even the implied warranty of MERCHANTABILITY or
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FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public
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License for more details.
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You should have received a copy of the GNU Lesser General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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#include <assert.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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#include "defines.h"
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#include "sine.h"
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#include "nlp.h"
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#include "dump.h"
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#include "lpc.h"
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#include "quantise.h"
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#include "phase.h"
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#include "interp.h"
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#include "postfilter.h"
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#include "codec2.h"
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typedef struct {
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float Sn[M]; /* input speech */
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float w[M]; /* time domain hamming window */
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COMP W[FFT_ENC]; /* DFT of w[] */
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float Pn[2*N]; /* trapezoidal synthesis window */
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float Sn_[2*N]; /* synthesised speech */
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float prev_Wo; /* previous frame's pitch estimate */
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float ex_phase; /* excitation model phase track */
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float bg_est; /* background noise estimate for post filter */
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MODEL prev_model; /* model parameters from 20ms ago */
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void *nlp; /* pitch predictor states */
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} CODEC2;
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/*---------------------------------------------------------------------------*\
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FUNCTION HEADERS
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\*---------------------------------------------------------------------------*/
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void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[]);
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void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model,float ak[]);
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/*---------------------------------------------------------------------------*\
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FUNCTIONS
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\*---------------------------------------------------------------------------*/
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/*---------------------------------------------------------------------------*\
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FUNCTION....: codec2_create
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AUTHOR......: David Rowe
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DATE CREATED: 21/8/2010
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Create and initialise an instance of the codec. Returns a pointer
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to the codec states or NULL on failure. One set of states is
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sufficient for a full duuplex codec (i.e. an encoder and decoder).
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You don't need separate states for encoders and decoders. See
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c2enc.c and c2dec.c for examples.
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\*---------------------------------------------------------------------------*/
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void *codec2_create()
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{
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CODEC2 *c2;
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int i,l;
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c2 = (CODEC2*)malloc(sizeof(CODEC2));
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if (c2 == NULL)
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return NULL;
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for(i=0; i<M; i++)
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c2->Sn[i] = 1.0;
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for(i=0; i<2*N; i++)
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c2->Sn_[i] = 0;
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make_analysis_window(c2->w,c2->W);
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make_synthesis_window(c2->Pn);
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quantise_init();
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c2->prev_Wo = 0.0;
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c2->bg_est = 0.0;
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c2->ex_phase = 0.0;
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for(l=1; l<=MAX_AMP; l++)
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c2->prev_model.A[l] = 0.0;
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c2->prev_model.Wo = TWO_PI/P_MAX;
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c2->nlp = nlp_create();
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if (c2->nlp == NULL) {
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free (c2);
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return NULL;
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}
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return (void*)c2;
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}
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/*---------------------------------------------------------------------------*\
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FUNCTION....: codec2_create
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AUTHOR......: David Rowe
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DATE CREATED: 21/8/2010
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Destroy an instance of the codec.
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\*---------------------------------------------------------------------------*/
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void codec2_destroy(void *codec2_state)
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{
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CODEC2 *c2;
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assert(codec2_state != NULL);
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c2 = (CODEC2*)codec2_state;
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nlp_destroy(c2->nlp);
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free(codec2_state);
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}
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/*---------------------------------------------------------------------------*\
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FUNCTION....: codec2_encode
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AUTHOR......: David Rowe
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DATE CREATED: 21/8/2010
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Encodes 160 speech samples (20ms of speech) into 51 bits.
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The codec2 algorithm actually operates internally on 10ms (80
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sample) frames, so we run the encoding algorithm twice. On the
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first frame we just send the voicing bit. One the second frame we
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send all model parameters.
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The bit allocation is:
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Parameter bits/frame
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--------------------------------------
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Harmonic magnitudes (LSPs) 36
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Low frequency LPC correction 1
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Energy 5
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Wo (fundamental frequnecy) 7
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Voicing (10ms update) 2
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TOTAL 51
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\*---------------------------------------------------------------------------*/
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void codec2_encode(void *codec2_state, unsigned char * bits, short speech[])
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{
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CODEC2 *c2;
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MODEL model;
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int voiced1, voiced2;
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int lsp_indexes[LPC_ORD];
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int lpc_correction;
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int energy_index;
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int Wo_index;
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int i;
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unsigned int nbit = 0;
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assert(codec2_state != NULL);
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c2 = (CODEC2*)codec2_state;
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/* first 10ms analysis frame - we just want voicing */
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analyse_one_frame(c2, &model, speech);
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voiced1 = model.voiced;
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/* second 10ms analysis frame */
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analyse_one_frame(c2, &model, &speech[N]);
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voiced2 = model.voiced;
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Wo_index = encode_Wo(model.Wo);
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encode_amplitudes(lsp_indexes,
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&lpc_correction,
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&energy_index,
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&model,
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c2->Sn,
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c2->w);
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memset(bits, '\0', ((CODEC2_BITS_PER_FRAME + 7) / 8));
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pack(bits, &nbit, Wo_index, WO_BITS);
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for(i=0; i<LPC_ORD; i++) {
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pack(bits, &nbit, lsp_indexes[i], lsp_bits(i));
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}
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pack(bits, &nbit, lpc_correction, 1);
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pack(bits, &nbit, energy_index, E_BITS);
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pack(bits, &nbit, voiced1, 1);
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pack(bits, &nbit, voiced2, 1);
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assert(nbit == CODEC2_BITS_PER_FRAME);
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}
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/*---------------------------------------------------------------------------*\
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FUNCTION....: codec2_decode
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AUTHOR......: David Rowe
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DATE CREATED: 21/8/2010
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Decodes frames of 51 bits into 160 samples (20ms) of speech.
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\*---------------------------------------------------------------------------*/
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void codec2_decode(void *codec2_state, short speech[],
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const unsigned char * bits)
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{
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CODEC2 *c2;
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MODEL model;
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int voiced1, voiced2;
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int lsp_indexes[LPC_ORD];
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int lpc_correction;
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int energy_index;
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int Wo_index;
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float ak[LPC_ORD+1];
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int i;
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unsigned int nbit = 0;
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MODEL model_interp;
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assert(codec2_state != NULL);
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c2 = (CODEC2*)codec2_state;
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Wo_index = unpack(bits, &nbit, WO_BITS);
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for(i=0; i<LPC_ORD; i++) {
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lsp_indexes[i] = unpack(bits, &nbit, lsp_bits(i));
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}
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lpc_correction = unpack(bits, &nbit, 1);
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energy_index = unpack(bits, &nbit, E_BITS);
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voiced1 = unpack(bits, &nbit, 1);
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voiced2 = unpack(bits, &nbit, 1);
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assert(nbit == CODEC2_BITS_PER_FRAME);
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model.Wo = decode_Wo(Wo_index);
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model.L = PI/model.Wo;
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decode_amplitudes(&model,
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ak,
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lsp_indexes,
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lpc_correction,
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energy_index);
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model.voiced = voiced2;
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model_interp.voiced = voiced1;
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interpolate(&model_interp, &c2->prev_model, &model);
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synthesise_one_frame(c2, speech, &model_interp, ak);
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synthesise_one_frame(c2, &speech[N], &model, ak);
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memcpy(&c2->prev_model, &model, sizeof(MODEL));
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}
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/*---------------------------------------------------------------------------*\
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FUNCTION....: synthesise_one_frame()
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AUTHOR......: David Rowe
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DATE CREATED: 23/8/2010
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Synthesise 80 speech samples (10ms) from model parameters.
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\*---------------------------------------------------------------------------*/
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void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model, float ak[])
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{
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int i;
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phase_synth_zero_order(model, ak, &c2->ex_phase);
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postfilter(model, &c2->bg_est);
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synthesise(c2->Sn_, model, c2->Pn, 1);
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for(i=0; i<N; i++) {
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if (c2->Sn_[i] > 32767.0)
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speech[i] = 32767;
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else if (c2->Sn_[i] < -32767.0)
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speech[i] = -32767;
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else
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speech[i] = c2->Sn_[i];
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}
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}
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/*---------------------------------------------------------------------------*\
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FUNCTION....: analyse_one_frame()
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AUTHOR......: David Rowe
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DATE CREATED: 23/8/2010
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Extract sinusoidal model parameters from 80 speech samples (10ms of
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speech).
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\*---------------------------------------------------------------------------*/
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void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[])
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{
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COMP Sw[FFT_ENC];
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COMP Sw_[FFT_ENC];
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float pitch;
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int i;
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/* Read input speech */
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for(i=0; i<M-N; i++)
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c2->Sn[i] = c2->Sn[i+N];
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for(i=0; i<N; i++)
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c2->Sn[i+M-N] = speech[i];
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dft_speech(Sw, c2->Sn, c2->w);
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/* Estimate pitch */
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nlp(c2->nlp,c2->Sn,N,M,P_MIN,P_MAX,&pitch,Sw,&c2->prev_Wo);
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c2->prev_Wo = TWO_PI/pitch;
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model->Wo = TWO_PI/pitch;
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model->L = PI/model->Wo;
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/* estimate model parameters */
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dft_speech(Sw, c2->Sn, c2->w);
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two_stage_pitch_refinement(model, Sw);
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estimate_amplitudes(model, Sw, c2->W);
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est_voicing_mbe(model, Sw, c2->W, (FS/TWO_PI)*model->Wo, Sw_);
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}
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