From c4da749362b2842ba0478dcdb309c29dd353c795 Mon Sep 17 00:00:00 2001 From: Mark Spencer Date: Tue, 15 Apr 2003 04:36:52 +0000 Subject: Add iLBC codec git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@852 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- CHANGES | 1 + channel.c | 2 + channels/chan_iax2.c | 3 + codecs/Makefile | 11 +- codecs/codec_ilbc.c | 284 +++++++++++++++++ codecs/ilbc/FrameClassify.c | 100 ++++++ codecs/ilbc/FrameClassify.h | 23 ++ codecs/ilbc/LPCdecode.c | 121 ++++++++ codecs/ilbc/LPCdecode.h | 42 +++ codecs/ilbc/LPCencode.c | 184 ++++++++++++ codecs/ilbc/LPCencode.h | 30 ++ codecs/ilbc/Makefile | 19 ++ codecs/ilbc/StateConstructW.c | 72 +++++ codecs/ilbc/StateConstructW.h | 28 ++ codecs/ilbc/StateSearchW.c | 177 +++++++++++ codecs/ilbc/StateSearchW.h | 43 +++ codecs/ilbc/anaFilter.c | 68 +++++ codecs/ilbc/anaFilter.h | 27 ++ codecs/ilbc/constants.c | 685 ++++++++++++++++++++++++++++++++++++++++++ codecs/ilbc/constants.h | 85 ++++++ codecs/ilbc/createCB.c | 207 +++++++++++++ codecs/ilbc/createCB.h | 55 ++++ codecs/ilbc/doCPLC.c | 303 +++++++++++++++++++ codecs/ilbc/doCPLC.h | 31 ++ codecs/ilbc/enhancer.c | 600 ++++++++++++++++++++++++++++++++++++ codecs/ilbc/enhancer.h | 34 +++ codecs/ilbc/filter.c | 161 ++++++++++ codecs/ilbc/filter.h | 72 +++++ codecs/ilbc/gainquant.c | 103 +++++++ codecs/ilbc/gainquant.h | 32 ++ codecs/ilbc/getCBvec.c | 171 +++++++++++ codecs/ilbc/getCBvec.h | 27 ++ codecs/ilbc/helpfun.c | 293 ++++++++++++++++++ codecs/ilbc/helpfun.h | 93 ++++++ codecs/ilbc/hpInput.c | 58 ++++ codecs/ilbc/hpInput.h | 26 ++ codecs/ilbc/hpOutput.c | 58 ++++ codecs/ilbc/hpOutput.h | 26 ++ codecs/ilbc/iCBConstruct.c | 104 +++++++ codecs/ilbc/iCBConstruct.h | 37 +++ codecs/ilbc/iCBSearch.c | 461 ++++++++++++++++++++++++++++ codecs/ilbc/iCBSearch.h | 32 ++ codecs/ilbc/iLBC_decode.c | 519 ++++++++++++++++++++++++++++++++ codecs/ilbc/iLBC_decode.h | 38 +++ codecs/ilbc/iLBC_define.h | 157 ++++++++++ codecs/ilbc/iLBC_encode.c | 446 +++++++++++++++++++++++++++ codecs/ilbc/iLBC_encode.h | 32 ++ codecs/ilbc/lsf.c | 252 ++++++++++++++++ codecs/ilbc/lsf.h | 29 ++ codecs/ilbc/packing.c | 169 +++++++++++ codecs/ilbc/packing.h | 65 ++++ codecs/ilbc/syntFilter.c | 65 ++++ codecs/ilbc/syntFilter.h | 26 ++ codecs/ilbc_slin_ex.h | 18 ++ codecs/slin_ilbc_ex.h | 28 ++ frame.c | 2 + include/asterisk/frame.h | 2 + rtp.c | 21 ++ 58 files changed, 6857 insertions(+), 1 deletion(-) create mode 100755 codecs/codec_ilbc.c create mode 100755 codecs/ilbc/FrameClassify.c create mode 100755 codecs/ilbc/FrameClassify.h create mode 100755 codecs/ilbc/LPCdecode.c create mode 100755 codecs/ilbc/LPCdecode.h create mode 100755 codecs/ilbc/LPCencode.c create mode 100755 codecs/ilbc/LPCencode.h create mode 100755 codecs/ilbc/Makefile create mode 100755 codecs/ilbc/StateConstructW.c create mode 100755 codecs/ilbc/StateConstructW.h create mode 100755 codecs/ilbc/StateSearchW.c create mode 100755 codecs/ilbc/StateSearchW.h create mode 100755 codecs/ilbc/anaFilter.c create mode 100755 codecs/ilbc/anaFilter.h create mode 100755 codecs/ilbc/constants.c create mode 100755 codecs/ilbc/constants.h create mode 100755 codecs/ilbc/createCB.c create mode 100755 codecs/ilbc/createCB.h create mode 100755 codecs/ilbc/doCPLC.c create mode 100755 codecs/ilbc/doCPLC.h create mode 100755 codecs/ilbc/enhancer.c create mode 100755 codecs/ilbc/enhancer.h create mode 100755 codecs/ilbc/filter.c create mode 100755 codecs/ilbc/filter.h create mode 100755 codecs/ilbc/gainquant.c create mode 100755 codecs/ilbc/gainquant.h create mode 100755 codecs/ilbc/getCBvec.c create mode 100755 codecs/ilbc/getCBvec.h create mode 100755 codecs/ilbc/helpfun.c create mode 100755 codecs/ilbc/helpfun.h create mode 100755 codecs/ilbc/hpInput.c create mode 100755 codecs/ilbc/hpInput.h create mode 100755 codecs/ilbc/hpOutput.c create mode 100755 codecs/ilbc/hpOutput.h create mode 100755 codecs/ilbc/iCBConstruct.c create mode 100755 codecs/ilbc/iCBConstruct.h create mode 100755 codecs/ilbc/iCBSearch.c create mode 100755 codecs/ilbc/iCBSearch.h create mode 100755 codecs/ilbc/iLBC_decode.c create mode 100755 codecs/ilbc/iLBC_decode.h create mode 100755 codecs/ilbc/iLBC_define.h create mode 100755 codecs/ilbc/iLBC_encode.c create mode 100755 codecs/ilbc/iLBC_encode.h create mode 100755 codecs/ilbc/lsf.c create mode 100755 codecs/ilbc/lsf.h create mode 100755 codecs/ilbc/packing.c create mode 100755 codecs/ilbc/packing.h create mode 100755 codecs/ilbc/syntFilter.c create mode 100755 codecs/ilbc/syntFilter.h create mode 100755 codecs/ilbc_slin_ex.h create mode 100755 codecs/slin_ilbc_ex.h diff --git a/CHANGES b/CHANGES index d0c970c56..39db2dca9 100755 --- a/CHANGES +++ b/CHANGES @@ -1,3 +1,4 @@ + -- Add iLBC codec Asterisk 0.4.0 -- Merge and edit Nick's FXO dial support -- Reengineer SIP registration (outbound) diff --git a/channel.c b/channel.c index 49982d849..33538e7a2 100755 --- a/channel.c +++ b/channel.c @@ -234,6 +234,8 @@ int ast_best_codec(int fmts) /* Okay, we're down to vocoders now, so pick GSM because it's small and easier to translate and sounds pretty good */ AST_FORMAT_GSM, + /* iLBC is not too bad */ + AST_FORMAT_ILBC, /* Speex is free, but computationally more expensive than GSM */ AST_FORMAT_SPEEX, /* Ick, LPC10 sounds terrible, but at least we have code for it, if you're tacky enough diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c index f13f8fa3d..eec8d0e33 100755 --- a/channels/chan_iax2.c +++ b/channels/chan_iax2.c @@ -538,6 +538,9 @@ static int get_samples(struct ast_frame *f) case AST_FORMAT_G723_1: samples = 240 /* XXX Not necessarily true XXX */; break; + case AST_FORMAT_ILBC: + samples = 240 * (f->datalen / 52); + break; case AST_FORMAT_GSM: samples = 160 * (f->datalen / 33); break; diff --git a/codecs/Makefile b/codecs/Makefile index e4beeea28..00778b821 100755 --- a/codecs/Makefile +++ b/codecs/Makefile @@ -20,6 +20,7 @@ MODG723=$(shell [ -f g723.1/coder.c ] && echo "codec_g723_1.so") MODG723+=$(shell [ -f g723.1b/coder2.c ] && echo "codec_g723_1b.so") MODSPEEX=$(shell [ -f /usr/include/speex.h ] || [ -f /usr/local/include/speex.h ] && echo "codec_speex.so") +MODILBC=$(shell [ -f ilbc/iLBC_decode.h ] && echo "codec_ilbc.so") CFLAGS+= LIBG723=g723.1/libg723.a @@ -30,8 +31,9 @@ LIBGSMT=$(shell if uname -m | grep -q 86; then echo gsm/lib/libgsm.a; fi) LIBMP3=mp3/libmp3.a LIBLPC10=lpc10/liblpc10.a LIBSPEEX=-lspeex -lm +LIBILBC=ilbc/libilbc.a -CODECS+=$(MODG723) $(MODSPEEX) codec_gsm.so codec_mp3_d.so codec_lpc10.so \ +CODECS+=$(MODG723) $(MODSPEEX) $(MODILBC) codec_gsm.so codec_mp3_d.so codec_lpc10.so \ codec_adpcm.so codec_ulaw.so codec_alaw.so codec_a_mu.so all: $(CODECS) @@ -43,6 +45,7 @@ clean: make -C gsm clean make -C mp3 clean make -C lpc10 clean + make -C ilbc clean $(LIBG723): make -C g723.1 all @@ -59,6 +62,12 @@ $(LIBMP3): $(LIBLPC10): make -C lpc10 all +$(LIBILBC): + make -C ilbc all + +codec_ilbc.so: codec_ilbc.o $(LIBILBC) + $(CC) -shared -Xlinker -x -o $@ $< $(LIBILBC) + codec_g723_1.so : codec_g723_1.o $(LIBG723) $(CC) -shared -Xlinker -x -o $@ $< $(LIBG723) diff --git a/codecs/codec_ilbc.c b/codecs/codec_ilbc.c new file mode 100755 index 000000000..c5e67f881 --- /dev/null +++ b/codecs/codec_ilbc.c @@ -0,0 +1,284 @@ +/* + * Asterisk -- A telephony toolkit for Linux. + * + * Translate between signed linear and Internet Low Bitrate Codec + * + * The iLBC code is from The IETF code base and is copyright GlobalSound, AB + * + * Copyright (C) 1999, Mark Spencer + * + * Mark Spencer + * + * This program is free software, distributed under the terms of + * the GNU General Public License + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ilbc/iLBC_encode.h" +#include "ilbc/iLBC_decode.h" + +/* Sample frame data */ +#include "slin_ilbc_ex.h" +#include "ilbc_slin_ex.h" + +#define USE_ILBC_ENHANCER 0 + +static pthread_mutex_t localuser_lock = AST_MUTEX_INITIALIZER; +static int localusecnt=0; + +static char *tdesc = "iLBC/PCM16 (signed linear) Codec Translator"; + +struct ast_translator_pvt { + iLBC_Enc_Inst_t enc; + iLBC_Dec_Inst_t dec; + struct ast_frame f; + /* Space to build offset */ + char offset[AST_FRIENDLY_OFFSET]; + /* Buffer for our outgoing frame */ + short outbuf[8000]; + /* Enough to store a full second */ + short buf[8000]; + int tail; +}; + +#define ilbc_coder_pvt ast_translator_pvt + +static struct ast_translator_pvt *lintoilbc_new(void) +{ + struct ilbc_coder_pvt *tmp; + tmp = malloc(sizeof(struct ilbc_coder_pvt)); + if (tmp) { + initEncode(&tmp->enc); + tmp->tail = 0; + localusecnt++; + } + return tmp; +} + +static struct ast_translator_pvt *ilbctolin_new(void) +{ + struct ilbc_coder_pvt *tmp; + tmp = malloc(sizeof(struct ilbc_coder_pvt)); + if (tmp) { + initDecode(&tmp->dec, USE_ILBC_ENHANCER); + tmp->tail = 0; + localusecnt++; + } + return tmp; +} + +static struct ast_frame *lintoilbc_sample(void) +{ + static struct ast_frame f; + f.frametype = AST_FRAME_VOICE; + f.subclass = AST_FORMAT_SLINEAR; + f.datalen = sizeof(slin_ilbc_ex); + /* Assume 8000 Hz */ + f.samples = sizeof(slin_ilbc_ex)/2; + f.mallocd = 0; + f.offset = 0; + f.src = __PRETTY_FUNCTION__; + f.data = slin_ilbc_ex; + return &f; +} + +static struct ast_frame *ilbctolin_sample(void) +{ + static struct ast_frame f; + f.frametype = AST_FRAME_VOICE; + f.subclass = AST_FORMAT_ILBC; + f.datalen = sizeof(ilbc_slin_ex); + /* All frames are 30 ms long */ + f.samples = 240; + f.mallocd = 0; + f.offset = 0; + f.src = __PRETTY_FUNCTION__; + f.data = ilbc_slin_ex; + return &f; +} + +static struct ast_frame *ilbctolin_frameout(struct ast_translator_pvt *tmp) +{ + if (!tmp->tail) + return NULL; + /* Signed linear is no particular frame size, so just send whatever + we have in the buffer in one lump sum */ + tmp->f.frametype = AST_FRAME_VOICE; + tmp->f.subclass = AST_FORMAT_SLINEAR; + tmp->f.datalen = tmp->tail * 2; + /* Assume 8000 Hz */ + tmp->f.samples = tmp->tail; + tmp->f.mallocd = 0; + tmp->f.offset = AST_FRIENDLY_OFFSET; + tmp->f.src = __PRETTY_FUNCTION__; + tmp->f.data = tmp->buf; + /* Reset tail pointer */ + tmp->tail = 0; + + return &tmp->f; +} + +static int ilbctolin_framein(struct ast_translator_pvt *tmp, struct ast_frame *f) +{ + /* Assuming there's space left, decode into the current buffer at + the tail location. Read in as many frames as there are */ + int x,i; + float tmpf[240]; + + if (f->datalen % 52) { + ast_log(LOG_WARNING, "Huh? An ilbc frame that isn't a multiple of 52 bytes long from %s (%d)?\n", f->src, f->datalen); + return -1; + } + + for (x=0;xdatalen;x+=52) { + if (tmp->tail + 240 < sizeof(tmp->buf)/2) { + iLBC_decode(tmpf, f->data + x, &tmp->dec, 1); + for (i=0;i<240;i++) + tmp->buf[tmp->tail + i] = tmpf[i]; + tmp->tail+=240; + } else { + ast_log(LOG_WARNING, "Out of buffer space\n"); + return -1; + } + } + return 0; +} + +static int lintoilbc_framein(struct ast_translator_pvt *tmp, struct ast_frame *f) +{ + /* Just add the frames to our stream */ + /* XXX We should look at how old the rest of our stream is, and if it + is too old, then we should overwrite it entirely, otherwise we can + get artifacts of earlier talk that do not belong */ + if (tmp->tail + f->datalen/2 < sizeof(tmp->buf) / 2) { + memcpy((tmp->buf + tmp->tail), f->data, f->datalen); + tmp->tail += f->datalen/2; + } else { + ast_log(LOG_WARNING, "Out of buffer space\n"); + return -1; + } + return 0; +} + +static struct ast_frame *lintoilbc_frameout(struct ast_translator_pvt *tmp) +{ + int x=0,i; + float tmpf[240]; + /* We can't work on anything less than a frame in size */ + if (tmp->tail < 240) + return NULL; + tmp->f.frametype = AST_FRAME_VOICE; + tmp->f.subclass = AST_FORMAT_ILBC; + tmp->f.mallocd = 0; + tmp->f.offset = AST_FRIENDLY_OFFSET; + tmp->f.src = __PRETTY_FUNCTION__; + tmp->f.data = tmp->outbuf; + while(tmp->tail >= 240) { + if ((x+1) * 52 >= sizeof(tmp->outbuf)) { + ast_log(LOG_WARNING, "Out of buffer space\n"); + break; + } + for (i=0;i<240;i++) + tmpf[i] = tmp->buf[i]; + /* Encode a frame of data */ + iLBC_encode(((unsigned char *)(tmp->outbuf)) + (x * 52), tmpf, &tmp->enc); + /* Assume 8000 Hz -- 20 ms */ + tmp->tail -= 240; + /* Move the data at the end of the buffer to the front */ + if (tmp->tail) + memmove(tmp->buf, tmp->buf + 240, tmp->tail * 2); + x++; + } + tmp->f.datalen = x * 52; + tmp->f.samples = x * 240; +#if 0 + { + static int fd = -1; + if (fd == -1) { + fd = open("ilbc.out", O_CREAT|O_TRUNC|O_WRONLY, 0666); + write(fd, tmp->f.data, tmp->f.datalen); + close(fd); + } + } +#endif + return &tmp->f; +} + +static void ilbc_destroy_stuff(struct ast_translator_pvt *pvt) +{ + free(pvt); + localusecnt--; +} + +static struct ast_translator ilbctolin = + { "ilbctolin", + AST_FORMAT_ILBC, AST_FORMAT_SLINEAR, + ilbctolin_new, + ilbctolin_framein, + ilbctolin_frameout, + ilbc_destroy_stuff, + ilbctolin_sample + }; + +static struct ast_translator lintoilbc = + { "lintoilbc", + AST_FORMAT_SLINEAR, AST_FORMAT_ILBC, + lintoilbc_new, + lintoilbc_framein, + lintoilbc_frameout, + ilbc_destroy_stuff, + lintoilbc_sample + }; + +int unload_module(void) +{ + int res; + ast_pthread_mutex_lock(&localuser_lock); + res = ast_unregister_translator(&lintoilbc); + if (!res) + res = ast_unregister_translator(&ilbctolin); + if (localusecnt) + res = -1; + ast_pthread_mutex_unlock(&localuser_lock); + return res; +} + +int load_module(void) +{ + int res; + res=ast_register_translator(&ilbctolin); + if (!res) + res=ast_register_translator(&lintoilbc); + else + ast_unregister_translator(&ilbctolin); + return res; +} + +char *description(void) +{ + return tdesc; +} + +int usecount(void) +{ + int res; + STANDARD_USECOUNT(res); + return res; +} + +char *key() +{ + return ASTERISK_GPL_KEY; +} diff --git a/codecs/ilbc/FrameClassify.c b/codecs/ilbc/FrameClassify.c new file mode 100755 index 000000000..61ede133b --- /dev/null +++ b/codecs/ilbc/FrameClassify.c @@ -0,0 +1,100 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + FrameClassify.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include "iLBC_define.h" + +/*----------------------------------------------------------------* + * Classification of subframes to localize start state + *---------------------------------------------------------------*/ + +int FrameClassify( /* index to the max-energy sub frame */ + float *residual /* (i) lpc residual signal */ +){ + float max_ssqEn, fssqEn[NSUB], bssqEn[NSUB], *pp; + int n, l, max_ssqEn_n; + const float ssqEn_win[NSUB-1]={(float)0.8,(float)0.9, + (float)1.0,(float)0.9,(float)0.8}; + const float sampEn_win[5]={(float)1.0/(float)6.0, + (float)2.0/(float)6.0, (float)3.0/(float)6.0, + (float)4.0/(float)6.0, (float)5.0/(float)6.0}; + + /* init the front and back energies to zero */ + + memset(fssqEn, 0, NSUB*sizeof(float)); + memset(bssqEn, 0, NSUB*sizeof(float)); + + /* Calculate front of first seqence */ + + n=0; + pp=residual; + for(l=0;l<5;l++){ + fssqEn[n] += sampEn_win[l] * (*pp) * (*pp); + pp++; + } + for(l=5;l max_ssqEn) { + max_ssqEn=(fssqEn[n-1]+bssqEn[n]) * + ssqEn_win[n-1]; + max_ssqEn_n=n; + } + } + + return max_ssqEn_n; +} + + diff --git a/codecs/ilbc/FrameClassify.h b/codecs/ilbc/FrameClassify.h new file mode 100755 index 000000000..c0d4b4f2e --- /dev/null +++ b/codecs/ilbc/FrameClassify.h @@ -0,0 +1,23 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + FrameClassify.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __iLBC_FRAMECLASSIFY_H +#define __iLBC_FRAMECLASSIFY_H + +int FrameClassify( /* Index to the max-energy sub frame */ + float *residual /* (i) lpc residual signal */ +); + +#endif + + diff --git a/codecs/ilbc/LPCdecode.c b/codecs/ilbc/LPCdecode.c new file mode 100755 index 000000000..5ca53d055 --- /dev/null +++ b/codecs/ilbc/LPCdecode.c @@ -0,0 +1,121 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + LPC_decode.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include +#include + +#include "helpfun.h" +#include "lsf.h" +#include "iLBC_define.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * interpolation of lsf coefficients for the decoder + *---------------------------------------------------------------*/ + +void LSFinterpolate2a_dec( + float *a, /* (o) lpc coefficients for a sub frame */ + float *lsf1, /* (i) first lsf coefficient vector */ + float *lsf2, /* (i) second lsf coefficient vector */ + float coef, /* (i) interpolation weight */ + int length /* (i) length of lsf vectors */ +){ + float lsftmp[LPC_FILTERORDER]; + + interpolate(lsftmp, lsf1, lsf2, coef, length); + lsf2a(a, lsftmp); +} + +/*----------------------------------------------------------------* + * obtain dequantized lsf coefficients from quantization index + *---------------------------------------------------------------*/ + +void SimplelsfDEQ( + float *lsfdeq, /* (o) dequantized lsf coefficients */ + int *index /* (i) quantization index */ +){ + int i,j, pos, cb_pos; + + /* decode first LSF */ + + pos = 0; + cb_pos = 0; + for (i = 0; i < LSF_NSPLIT; i++) { + for (j = 0; j < dim_lsfCbTbl[i]; j++) { + lsfdeq[pos + j] = lsfCbTbl[cb_pos + + (long)(index[i])*dim_lsfCbTbl[i] + j]; + } + pos += dim_lsfCbTbl[i]; + cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i]; + } + + /* decode last LSF */ + + pos = 0; + cb_pos = 0; + for (i = 0; i < LSF_NSPLIT; i++) { + for (j = 0; j < dim_lsfCbTbl[i]; j++) { + lsfdeq[LPC_FILTERORDER + pos + j] = lsfCbTbl[cb_pos + + (long)(index[LSF_NSPLIT + i])*dim_lsfCbTbl[i] + j]; + } + pos += dim_lsfCbTbl[i]; + cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i]; + } +} + +/*----------------------------------------------------------------* + * obtain synthesis and weighting filters form lsf coefficients + *---------------------------------------------------------------*/ + +void DecoderInterpolateLSF( + float *syntdenum, /* (o) synthesis filter coefficients */ + float *weightdenum, /* (o) weighting denumerator + coefficients */ + float *lsfdeq, /* (i) dequantized lsf coefficients */ + int length, /* (i) length of lsf coefficient vector */ + iLBC_Dec_Inst_t *iLBCdec_inst + /* (i) the decoder state structure */ +){ + int i, pos, lp_length; + float lp[LPC_FILTERORDER + 1], *lsfdeq2; + + lsfdeq2 = lsfdeq + length; + lp_length = length + 1; + + /* subframe 1: Interpolation between old and first */ + + LSFinterpolate2a_dec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq, + lsf_weightTbl[0], length); + memcpy(syntdenum,lp,lp_length*sizeof(float)); + bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM, lp_length); + + /* subframes 2 to 6: interpolation between first and last + LSF */ + + pos = lp_length; + for (i = 1; i < 6; i++) { + LSFinterpolate2a_dec(lp, lsfdeq, lsfdeq2, lsf_weightTbl[i], + length); + memcpy(syntdenum + pos,lp,lp_length*sizeof(float)); + bwexpand(weightdenum + pos, lp, + LPC_CHIRP_WEIGHTDENUM, lp_length); + pos += lp_length; + } + + /* update memory */ + + memcpy((*iLBCdec_inst).lsfdeqold, lsfdeq2, length*sizeof(float)); + +} + + diff --git a/codecs/ilbc/LPCdecode.h b/codecs/ilbc/LPCdecode.h new file mode 100755 index 000000000..f57264882 --- /dev/null +++ b/codecs/ilbc/LPCdecode.h @@ -0,0 +1,42 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + LPC_decode.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __iLBC_LPC_DECODE_H +#define __iLBC_LPC_DECODE_H + +void LSFinterpolate2a_dec( + float *a, /* (o) lpc coefficients for a sub frame */ + float *lsf1, /* (i) first lsf coefficient vector */ + float *lsf2, /* (i) second lsf coefficient vector */ + float coef, /* (i) interpolation weight */ + int length /* (i) length of lsf vectors */ +); + +void SimplelsfDEQ( + float *lsfdeq, /* (o) dequantized lsf coefficients */ + int *index /* (i) quantization index */ +); + +void DecoderInterpolateLSF( + float *syntdenum, /* (o) synthesis filter coefficients */ + float *weightdenum, /* (o) weighting denumerator + coefficients */ + float *lsfdeq, /* (i) dequantized lsf coefficients */ + int length, /* (i) length of lsf coefficient vector */ + iLBC_Dec_Inst_t *iLBCdec_inst + /* (i) the decoder state structure */ +); + +#endif + + diff --git a/codecs/ilbc/LPCencode.c b/codecs/ilbc/LPCencode.c new file mode 100755 index 000000000..330796d7b --- /dev/null +++ b/codecs/ilbc/LPCencode.c @@ -0,0 +1,184 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + LPCencode.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include + +#include "iLBC_define.h" +#include "helpfun.h" +#include "lsf.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * lpc analysis (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void SimpleAnalysis( + float *lsf, /* (o) lsf coefficients */ + float *data, /* (i) new data vector */ + float *lpc_buffer /* (i) buffer containing old data */ +){ + int k, is; + float temp[BLOCKL], lp[LPC_FILTERORDER + 1]; + float lp2[LPC_FILTERORDER + 1]; + float r[LPC_FILTERORDER + 1]; + + memcpy(lpc_buffer+LPC_LOOKBACK,data,BLOCKL*sizeof(float)); + + /* No lookahead, last window is asymmetric */ + + for (k = 0; k < LPC_N; k++) { + + is = LPC_LOOKBACK; + + if (k < (LPC_N - 1)) { + window(temp, lpc_winTbl, lpc_buffer, BLOCKL); + } else { + window(temp, lpc_asymwinTbl, lpc_buffer + is, BLOCKL); + } + + autocorr(r, temp, BLOCKL, LPC_FILTERORDER); + window(r, r, lpc_lagwinTbl, LPC_FILTERORDER + 1); + + levdurb(lp, temp, r, LPC_FILTERORDER); + bwexpand(lp2, lp, LPC_CHIRP_SYNTDENUM, LPC_FILTERORDER+1); + + a2lsf(lsf + k*LPC_FILTERORDER, lp2); + } + memcpy(lpc_buffer, lpc_buffer+BLOCKL, + LPC_LOOKBACK*sizeof(float)); +} + +/*----------------------------------------------------------------* + * lsf interpolator and conversion from lsf to a coefficients + * (subrutine to SimpleInterpolateLSF) + *---------------------------------------------------------------*/ + +void LSFinterpolate2a_enc( + float *a, /* (o) lpc coefficients */ + float *lsf1,/* (i) first set of lsf coefficients */ + float *lsf2,/* (i) second set of lsf coefficients */ + float coef, /* (i) weighting coefficient to use between lsf1 + and lsf2 */ + long length /* (i) length of coefficient vectors */ +){ + float lsftmp[LPC_FILTERORDER]; + + interpolate(lsftmp, lsf1, lsf2, coef, length); + lsf2a(a, lsftmp); +} + +/*----------------------------------------------------------------* + * lsf interpolator (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void SimpleInterpolateLSF( + float *syntdenum, /* (o) the synthesis filter denominator + resulting from the quantized + interpolated lsf */ + float *weightdenum, /* (o) the weighting filter denominator + resulting from the unquantized + interpolated lsf */ + float *lsf, /* (i) the unquantized lsf coefficients */ + float *lsfdeq, /* (i) the dequantized lsf coefficients */ + float *lsfold, /* (i) the unquantized lsf coefficients of + the previous signal frame */ + float *lsfdeqold, /* (i) the dequantized lsf coefficients of + the previous signal frame */ + int length /* (i) should equate FILTERORDER */ +){ + int i, pos, lp_length; + float lp[LPC_FILTERORDER + 1], *lsf2, *lsfdeq2; + + lsf2 = lsf + length; + lsfdeq2 = lsfdeq + length; + lp_length = length + 1; + + /* subframe 1: Interpolation between old and first set of + lsf coefficients */ + + LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq, + lsf_weightTbl[0], length); + memcpy(syntdenum,lp,lp_length*sizeof(float)); + LSFinterpolate2a_enc(lp, lsfold, lsf, lsf_weightTbl[0], length); + bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM, lp_length); + + /* subframe 2 to 6: Interpolation between first and second + set of lsf coefficients */ + + pos = lp_length; + for (i = 1; i < NSUB; i++) { + LSFinterpolate2a_enc(lp, lsfdeq, lsfdeq2, + lsf_weightTbl[i], length); + memcpy(syntdenum + pos,lp,lp_length*sizeof(float)); + + LSFinterpolate2a_enc(lp, lsf, lsf2, + lsf_weightTbl[i], length); + bwexpand(weightdenum + pos, lp, + LPC_CHIRP_WEIGHTDENUM, lp_length); + pos += lp_length; + } + + /* update memory */ + + memcpy(lsfold, lsf2, length*sizeof(float)); + memcpy(lsfdeqold, lsfdeq2, length*sizeof(float)); +} + +/*----------------------------------------------------------------* + * lsf quantizer (subrutine to LPCencode) + *---------------------------------------------------------------*/ + +void SimplelsfQ( + float *lsfdeq, /* (o) dequantized lsf coefficients + (dimension FILTERORDER) */ + int *index, /* (o) quantization index */ + float *lsf /* (i) the lsf coefficient vector to be + quantized (dimension FILTERORDER ) */ +){ + /* Quantize first LSF with memoryless split VQ */ + SplitVQ(lsfdeq, index, lsf, lsfCbTbl, LSF_NSPLIT, + dim_lsfCbTbl, size_lsfCbTbl); + + /* Quantize second LSF with memoryless split VQ */ + SplitVQ(lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT, + lsf + LPC_FILTERORDER, lsfCbTbl, LSF_NSPLIT, + dim_lsfCbTbl, size_lsfCbTbl); +} + +/*----------------------------------------------------------------* + * lpc encoder + *---------------------------------------------------------------*/ + +void LPCencode( + float *syntdenum, /* (i/o) synthesis filter coefficients + before/after encoding */ + float *weightdenum, /* (i/o) weighting denumerator coefficients + before/after encoding */ + int *lsf_index, /* (o) lsf quantization index */ + float *data, /* (i) lsf coefficients to quantize */ + iLBC_Enc_Inst_t *iLBCenc_inst + /* (i/o) the encoder state structure */ +){ + float lsf[LPC_FILTERORDER * LPC_N]; + float lsfdeq[LPC_FILTERORDER * LPC_N]; + int change=0; + + SimpleAnalysis(lsf, data, (*iLBCenc_inst).lpc_buffer); + SimplelsfQ(lsfdeq, lsf_index, lsf); + change=LSF_check(lsfdeq, LPC_FILTERORDER, LPC_N); + SimpleInterpolateLSF(syntdenum, weightdenum, + lsf, lsfdeq, (*iLBCenc_inst).lsfold, + (*iLBCenc_inst).lsfdeqold, LPC_FILTERORDER); +} + + diff --git a/codecs/ilbc/LPCencode.h b/codecs/ilbc/LPCencode.h new file mode 100755 index 000000000..1b6b83749 --- /dev/null +++ b/codecs/ilbc/LPCencode.h @@ -0,0 +1,30 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + LPCencode.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __iLBC_LPCENCOD_H +#define __iLBC_LPCENCOD_H + +void LPCencode( + float *syntdenum, /* (i/o) synthesis filter coefficients + before/after encoding */ + float *weightdenum, /* (i/o) weighting denumerator coefficients + before/after encoding */ + int *lsf_index, /* (o) lsf quantization index */ + float *data, /* (i) lsf coefficients to quantize */ + iLBC_Enc_Inst_t *iLBCenc_inst + /* (i/o) the encoder state structure */ +); + +#endif + + diff --git a/codecs/ilbc/Makefile b/codecs/ilbc/Makefile new file mode 100755 index 000000000..ba7466ee0 --- /dev/null +++ b/codecs/ilbc/Makefile @@ -0,0 +1,19 @@ +LIB=libilbc.a + +OBJS= anaFilter.o iCBSearch.o packing.o \ + constants.o gainquant.o iLBC_decode.o StateConstructW.o \ + createCB.o getCBvec.o iLBC_encode.o StateSearchW.o doCPLC.o \ + helpfun.o syntFilter.o enhancer.o hpInput.o LPCdecode.o \ + filter.o hpOutput.o LPCencode.o FrameClassify.o iCBConstruct.o lsf.o + +all: $(LIB) + + +$(LIB): $(OBJS) + ar cr $(LIB) $(OBJS) + +clean: + rm -f $(LIB) *.o + +install: + diff --git a/codecs/ilbc/StateConstructW.c b/codecs/ilbc/StateConstructW.c new file mode 100755 index 000000000..c694106ae --- /dev/null +++ b/codecs/ilbc/StateConstructW.c @@ -0,0 +1,72 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + StateConstructW.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include +#include + +#include "iLBC_define.h" +#include "constants.h" +#include "filter.h" + +/*----------------------------------------------------------------* + * decoding of the start state + *---------------------------------------------------------------*/ + +void StateConstructW( + int idxForMax, /* (i) 6-bit index for the quantization of + max amplitude */ + int *idxVec, /* (i) vector of quantization indexes */ + float *syntDenum, /* (i) synthesis filter denumerator */ + float *out, /* (o) the decoded state vector */ + int len /* (i) length of a state vector */ +){ + float maxVal, tmpbuf[LPC_FILTERORDER+2*STATE_LEN], *tmp, + numerator[LPC_FILTERORDER+1]; + float foutbuf[LPC_FILTERORDER+2*STATE_LEN], *fout; + int k,tmpi; + + /* decoding of the maximum value */ + + maxVal = state_frgqTbl[idxForMax]; + maxVal = (float)pow(10,maxVal)/(float)4.5; + + /* initialization of buffers and coefficients */ + + memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float)); + memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float)); + for(k=0; k +#include + +#include "iLBC_define.h" +#include "constants.h" +#include "filter.h" +#include "helpfun.h" + +/*----------------------------------------------------------------* + * predictive noise shaping encoding of scaled start state + * (subrutine for StateSearchW) + *---------------------------------------------------------------*/ + +void AbsQuantW( + float *in, /* (i) vector to encode */ + float *syntDenum, /* (i) denominator of synthesis filter */ + float *weightDenum, /* (i) denominator of weighting filter */ + int *out, /* (o) vector of quantizer indexes */ + int len, /* (i) length of vector to encode and + vector of quantizer indexes */ + int state_first /* (i) position of start state in the + 80 vec */ +){ + float *syntOut, syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN]; + float toQ, xq; + int n; + int index; + + /* initialization of buffer for filtering */ + + memset(syntOutBuf, 0, LPC_FILTERORDER*sizeof(float)); + + /* initialization of pointer for filtering */ + + syntOut = &syntOutBuf[LPC_FILTERORDER]; + + /* synthesis and weighting filters on input */ + + if (state_first) { + AllPoleFilter (in, weightDenum, SUBL, LPC_FILTERORDER); + } else { + AllPoleFilter (in, weightDenum, STATE_SHORT_LEN-SUBL, + LPC_FILTERORDER); + } + + /* encoding loop */ + + for(n=0;n maxVal*maxVal){ + maxVal = fout[k]; + } + } + maxVal=(float)fabs(maxVal); + + /* encoding of the maximum amplitude value */ + + if(maxVal < 10.0){ + maxVal = 10.0; + } + maxVal = (float)log10(maxVal); + sort_sq(&dtmp, idxForMax, maxVal, state_frgqTbl, 64); + + /* decoding of the maximum amplitude representation value, + and corresponding scaling of start state */ + + maxVal=state_frgqTbl[*idxForMax]; + qmax = (float)pow(10,maxVal); + scal = (float)(4.5)/qmax; + for(k=0;k +#include "iLBC_define.h" + +/*----------------------------------------------------------------* + * LP analysis filter. + *---------------------------------------------------------------*/ + +void anaFilter( + float *In, /* (i) Signal to be filtered */ + float *a, /* (i) LP parameters */ + int len,/* (i) Length of signal */ + float *Out, /* (o) Filtered signal */ + float *mem /* (i/o) Filter state */ +){ + int i, j; + float *po, *pi, *pm, *pa; + + po = Out; + + /* Filter first part using memory from past */ + + for (i=0;i +#include + +/*----------------------------------------------------------------* + * Construct an additional codebook vector by filtering the + * initial codebook buffer. This vector is then used to expand + * the codebook with an additional section. + *---------------------------------------------------------------*/ + +void filteredCBvecs( + float *cbvectors, /* (o) Codebook vectors for the higher +section */ + float *mem, /* (i) Buffer to create codebook vector from +*/ + int lMem /* (i) Length of buffer */ +){ + int j, k; + float *pp, *pp1; + float tempbuff2[CB_MEML+CB_FILTERLEN]; + float *pos; + + memset(tempbuff2, 0, (CB_HALFFILTERLEN-1)*sizeof(float)); + memcpy(&tempbuff2[CB_HALFFILTERLEN-1], mem, lMem*sizeof(float)); + memset(&tempbuff2[lMem+CB_HALFFILTERLEN-1], 0, + (CB_HALFFILTERLEN+1)*sizeof(float)); + + /* Create codebook vector for higher section by filtering */ + + /* do filtering */ + pos=cbvectors; + memset(pos, 0, lMem*sizeof(float)); + for (k=0; k0.0) { + invenergy[tmpIndex]=(float)1.0/(energy[tmpIndex]+EPS); + } else { + invenergy[tmpIndex] = (float) 0.0; + } + + if (stage==0) { + measure = (float)-10000000.0; + + if (crossDot > 0.0) { + measure = crossDot*crossDot*invenergy[tmpIndex]; + } + } + else { + measure = crossDot*crossDot*invenergy[tmpIndex]; + } + + /* check if measure is better */ + ftmp = crossDot*invenergy[tmpIndex]; + + if ((measure>*max_measure) && (fabs(ftmp) +#include + +#include "iLBC_define.h" + +/*----------------------------------------------------------------* + * Compute cross correlation and pitch gain for pitch prediction + * of last subframe at given lag. + *---------------------------------------------------------------*/ + +void compCorr( + float *cc, /* (o) cross correlation coefficient */ + float *gc, /* (o) gain */ + float *buffer, /* (i) signal buffer */ + int lag, /* (i) pitch lag */ + int bLen, /* (i) length of buffer */ + int sRange /* (i) correlation search length */ +){ + int i; + float ftmp1, ftmp2; + + ftmp1 = 0.0; + ftmp2 = 0.0; + for (i=0; i 0.0) { + *cc = ftmp1*ftmp1/ftmp2; + *gc = (float)fabs(ftmp1/ftmp2); + } + else { + *cc = 0.0; + *gc = 0.0; + } +} + +/*----------------------------------------------------------------* + * Packet loss concealment routine. Conceals a residual signal + * and LP parameters. If no packet loss, update state. + *---------------------------------------------------------------*/ + +void doThePLC( + float *PLCresidual, /* (o) concealed residual */ + float *PLClpc, /* (o) concealed LP parameters */ + int PLI, /* (i) packet loss indicator + 0 - no PL, 1 = PL */ + float *decresidual, /* (i) decoded residual */ + float *lpc, /* (i) decoded LPC (only used for no PL) */ + int inlag, /* (i) pitch lag */ + iLBC_Dec_Inst_t *iLBCdec_inst + /* (i/o) decoder instance */ +){ + int lag=20, randlag; + float gain, maxcc; + float gain_comp, maxcc_comp; + int i, pick, offset; + float ftmp, ftmp1, randvec[BLOCKL], pitchfact; + + /* Packet Loss */ + + if (PLI == 1) { + + (*iLBCdec_inst).consPLICount += 1; + + /* if previous frame not lost, + determine pitch pred. gain */ + + if ((*iLBCdec_inst).prevPLI != 1) { + + /* Search around the previous lag to find the + best pitch period */ + + lag=inlag-3; + compCorr(&maxcc, &gain, (*iLBCdec_inst).prevResidual, + lag, BLOCKL, 60); + for (i=inlag-2;i<=inlag+3;i++) { + compCorr(&maxcc_comp, &gain_comp, + (*iLBCdec_inst).prevResidual, + i, BLOCKL, 60); + + if (maxcc_comp>maxcc) { + maxcc=maxcc_comp; + gain=gain_comp; + lag=i; + } + } + + if (gain > 1.0) { + gain = 1.0; + } + } + + /* previous frame lost, use recorded lag and gain */ + + else { + lag=(*iLBCdec_inst).prevLag; + gain=(*iLBCdec_inst).prevGain; + } + + /* Attenuate signal and scale down pitch pred gain if + several frames lost consecutively */ + + + if ((*iLBCdec_inst).consPLICount > 1) { + gain *= (float)0.9; + } + + /* Compute mixing factor of picth repeatition and noise */ + + + if (gain > PLC_XT_MIX) { + pitchfact = PLC_YT_MIX; + } else if (gain < PLC_XB_MIX) { + pitchfact = PLC_YB_MIX; + } else { + pitchfact = PLC_YB_MIX + (gain - PLC_XB_MIX) * + (PLC_YT_MIX-PLC_YB_MIX)/(PLC_XT_MIX-PLC_XB_MIX); + } + + /* compute concealed residual */ + + (*iLBCdec_inst).energy = 0.0; + for (i=0; i= PLC_GAINTHRESHOLD) { + + /* Compute mixing factor of pitch repeatition + and noise */ + + if (gain > PLC_XT_MIX) { + pitchfact = PLC_YT_MIX; + } else if (gain < PLC_XB_MIX) { + pitchfact = PLC_YB_MIX; + } else { + pitchfact = PLC_YB_MIX + (gain - PLC_XB_MIX) * + (PLC_YT_MIX-PLC_YB_MIX)/(PLC_XT_MIX-PLC_XB_MIX); + } + + /* compute concealed residual for 3 subframes */ + + for (i=0; i<3*SUBL; i++) { + + (*iLBCdec_inst).seed=((*iLBCdec_inst).seed* + 69069L+1) & (0x80000000L-1); + randlag = 50 + ((signed long) + (*iLBCdec_inst).seed)%70; + + /* noise component */ + + pick = i - randlag; + + if (pick < 0) { + randvec[i] = gain * + (*iLBCdec_inst).prevResidual[BLOCKL+pick]; + } else { + randvec[i] = gain * randvec[pick]; + } + + /* pitch repeatition component */ + + pick = i - lag; + + if (pick < 0) { + PLCresidual[i] = gain * + (*iLBCdec_inst).prevResidual[BLOCKL+pick]; + } else { + PLCresidual[i] = gain * PLCresidual[pick]; + } + + /* mix noise and pitch repeatition */ + + PLCresidual[i] = (pitchfact * PLCresidual[i] + + ((float)1.0 - pitchfact) * randvec[i]); + } + + /* interpolate concealed residual with actual + residual */ + + offset = 3*SUBL; + for (i=0; i +#include +#include "iLBC_define.h" +#include "constants.h" +#include "filter.h" + +/*----------------------------------------------------------------* + * Find index in array such that the array element with said + * index is the element of said array closest to "value" + * according to the squared-error criterion + *---------------------------------------------------------------*/ + +void NearestNeighbor( + int *index, /* (o) index of array element closest to value */ + float *array, /* (i) data array */ + float value,/* (i) value */ + int arlength/* (i) dimension of data array */ +){ + int i; + float bestcrit,crit; + + crit=array[0]-value; + bestcrit=crit*crit; + *index=0; + for(i=1;i dim1){ + hfl2=(int) (dim1/2); + for(j=0;j= idatal) { + searchSegEndPos=idatal-ENH_BLOCKL-1; + } + corrdim=searchSegEndPos-searchSegStartPos+1; + + /* compute upsampled correlation (corr33) and find + location of max */ + + mycorr1(corrVec,idata+searchSegStartPos, + corrdim+ENH_BLOCKL-1,idata+centerStartPos,ENH_BLOCKL); + enh_upsample(corrVecUps,corrVec,corrdim,ENH_FL0); + tloc=0; maxv=corrVecUps[0]; + for(i=1;imaxv){ + tloc=i; + maxv=corrVecUps[i]; + } + } + + /* make vector can be upsampled without ever running outside + bounds */ + + *updStartPos= (float)searchSegStartPos + + (float)tloc/(float)ENH_UPS0+(float)1.0; + tloc2=(int)(tloc/ENH_UPS0); + + if (tloc>tloc2*ENH_UPS0) { + tloc2++; + } + st=searchSegStartPos+tloc2-ENH_FL0; + + if(st<0){ + memset(vect,0,-st*sizeof(float)); + memcpy(&vect[-st],idata, (ENH_VECTL+st)*sizeof(float)); + } + else{ + en=st+ENH_VECTL; + + if(en>idatal){ + memcpy(vect, &idata[st], + (ENH_VECTL-(en-idatal))*sizeof(float)); + memset(&vect[ENH_VECTL-(en-idatal)], 0, + (en-idatal)*sizeof(float)); + } + else { + memcpy(vect, &idata[st], ENH_VECTL*sizeof(float)); + } + } + fraction=tloc2*ENH_UPS0-tloc; + + /* compute the segment (this is actually a convolution) */ + + mycorr1(seg,vect,ENH_VECTL,polyphaserTbl+(2*ENH_FL0+1)*fraction, + 2*ENH_FL0+1); +} + +/*----------------------------------------------------------------* + * find the smoothed output data + *---------------------------------------------------------------*/ + +void smath( + float *odata, /* (o) smoothed output */ + float *sseq,/* (i) said second sequence of waveforms */ + int hl, /* (i) 2*hl+1 is sseq dimension */ + float alpha0/* (i) max smoothing energy fraction */ +){ + int i,k; + float w00,w10,w11,A,B,C,*psseq,err,errs; + float surround[BLOCKL]; /* shape contributed by other than + current */ + float wt[2*ENH_HL+1]; /* waveform weighting to get surround + shape */ + float denom; + + /* create shape of contribution from all waveforms except the + current one */ + + for(i=1;i<=2*hl+1; i++) { + wt[i-1] = (float)0.5*(1 - (float)cos(2*PI*i/(2*hl+2))); + } + wt[hl]=0.0; /* for clarity, not used */ + for(i=0;i alpha0 * w00){ + if( w00 < 1) { + w00=1; + } + denom = (w11*w00-w10*w10)/(w00*w00); + + if( denom > 0.0001){ /* eliminates numerical problems + for if smooth */ + A = (float)sqrt( (alpha0- alpha0*alpha0/4)/denom); + B = -alpha0/2 - A * w10/w00; + B = B+1; + } + else{ /* essentially no difference between cycles; + smoothing not needed */ + A= 0.0; + B= 1.0; + } + + /* create smoothed sequence */ + + psseq=sseq+hl*ENH_BLOCKL; + for(i=0;i=0;q--) { + blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]]; + NearestNeighbor(lagBlock+q,plocs, + blockStartPos[q]+ENH_BLOCKL_HALF-period[lagBlock[q+1]], + periodl); + + + if(blockStartPos[q]-ENH_OVERHANG>=0) { + refiner(sseq+q*ENH_BLOCKL,blockStartPos+q,idata,idatal, + centerStartPos,blockStartPos[q], + period[lagBlock[q+1]]); + } else { + + psseq=sseq+q*ENH_BLOCKL; + memset(psseq, 0, ENH_BLOCKL*sizeof(float)); + } + } + + /* future */ + + for(i=0;i 0.0) { + return (float)(ftmp1*ftmp1/ftmp2); + } + else { + return (float)0.0; + } +} + +/*----------------------------------------------------------------* + * interface for enhancer + *---------------------------------------------------------------*/ + +int enhancerInterface( + float *out, /* (o) enhanced signal */ + float *in, /* (i) unenhanced signal */ + iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */ +){ + float *enh_buf, *enh_period; + int iblock, isample; + int lag, ilag, i; + float cc, maxcc; + float ftmp1, ftmp2, gain; + float *inPtr, *enh_bufPtr1, *enh_bufPtr2; + + float lpState[6], downsampled[(ENH_NBLOCKS*ENH_BLOCKL+120)/2]; + int inLen=ENH_NBLOCKS*ENH_BLOCKL+120; + int start; + + enh_buf=iLBCdec_inst->enh_buf; + enh_period=iLBCdec_inst->enh_period; + + + memmove(enh_buf, &enh_buf[ENH_NBLOCKS*ENH_BLOCKL], + (ENH_NBLOCKS_EXTRA*ENH_BLOCKL)*sizeof(float)); + + memcpy(&enh_buf[ENH_NBLOCKS_EXTRA*ENH_BLOCKL], in, + (ENH_NBLOCKS*ENH_BLOCKL)*sizeof(float)); + + if (iLBCdec_inst->prev_enh_pl==1) { + /* PLC was performed on the previous packet */ + + lag = 20; + maxcc = xCorrCoef(in, in+lag, ENH_BLOCKL); + for (ilag=21; ilag<120; ilag++) { + cc = xCorrCoef(in, in+ilag, ENH_BLOCKL); + + if (cc > maxcc) { + maxcc = cc; + lag = ilag; + } + } + + ftmp1 = 0.0; + ftmp2 = 0.0; + for (i=0; i 0.0) { + gain=(float)(ftmp1/ftmp2); + } + else { + gain=(float)0.0; + } + if (gain>1.0) { + gain=1.0; + } else if (gain<-1.0) { + gain=-1.0; + } + + inPtr=&in[lag-1]; + + enh_bufPtr1=&enh_buf[ENH_NBLOCKS_EXTRA*ENH_BLOCKL-1]; + + if (lag>ENH_BLOCKL) { + start=ENH_BLOCKL; + } else { + start=lag; + } + + for (isample = start; isample>0; isample--) { + *enh_bufPtr1-- = gain*(*inPtr--); + } + + enh_bufPtr2=&enh_buf[ENH_NBLOCKS_EXTRA*ENH_BLOCKL-1]; + for (isample = (ENH_BLOCKL-1-lag); isample>=0; isample--) { + *enh_bufPtr1-- = gain*(*enh_bufPtr2--); + } + + } + + memmove(enh_period, &enh_period[ENH_NBLOCKS], + ENH_NBLOCKS_EXTRA*sizeof(float)); + + + /* Set state information to the 6 samples right before + the samples to be downsampled. */ + + memcpy(lpState, enh_buf+ENH_NBLOCKS_EXTRA*ENH_BLOCKL-126, + 6*sizeof(float)); + + /* Down sample a factor 2 to save computations */ + + DownSample(enh_buf+ENH_NBLOCKS_EXTRA*ENH_BLOCKL-120, + lpFilt_coefsTbl, inLen, + lpState, downsampled); + + /* Estimate the pitch in the down sampled domain. */ + for(iblock = 0; iblock maxcc) { + maxcc = cc; + lag = ilag; + } + } + + /* Store the estimated lag in the non-downsampled domain */ + enh_period[iblock+ENH_NBLOCKS_EXTRA] = (float)lag*2; + } + + for(iblock = 0; iblock +#include +#include "constants.h" +#include "filter.h" + +/*----------------------------------------------------------------* + * quantizer for the gain in the gain-shape coding of residual + *---------------------------------------------------------------*/ + +float gainquant(/* (o) quantized gain value */ + float in, /* (i) gain value */ + float maxIn,/* (i) maximum of gain value */ + int cblen, /* (i) number of quantization indices */ + int *index /* (o) quantization index */ +){ + int i, tindex; + float minmeasure,measure, *cb, scale; + + /* ensure a lower bound on the scaling factor */ + + scale=maxIn; + + if (scale<0.1) { + scale=(float)0.1; + } + + /* select the quantization table */ + + if (cblen == 8) { + cb = gain_sq3Tbl; + } else if (cblen == 16) { + cb = gain_sq4Tbl; + } else { + cb = gain_sq5Tbl; + } + + /* select the best index in the quantization table */ + + minmeasure=10000000.0; + tindex=0; + for (i=0;i + +/*----------------------------------------------------------------* + * Construct codebook vector for given index. + *---------------------------------------------------------------*/ + +void getCBvec( + float *cbvec, /* (o) Constructed codebook vector */ + float *mem, /* (i) Codebook buffer */ + int index, /* (i) Codebook index */ + int lMem, /* (i) Length of codebook buffer */ + int cbveclen/* (i) Codebook vector length */ +){ + int j, k, n, memInd, sFilt; + float tmpbuf[CB_MEML]; + int base_size; + int ilow, ihigh; + float alfa, alfa1; + + /* Determine size of codebook sections */ + + base_size=lMem-cbveclen+1; + + if (cbveclen==SUBL) { + base_size+=cbveclen/2; + } + + /* No filter -> First codebook section */ + + if (index + +#include "iLBC_define.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * calculation of auto correlation + *---------------------------------------------------------------*/ + +void autocorr( + float *r, /* (o) autocorrelation vector */ + const float *x, /* (i) data vector */ + int N, /* (i) length of data vector */ + int order /* largest lag for calculated autocorrelations */ +){ + int lag, n; + float sum; + + for (lag = 0; lag <= order; lag++) { + sum = 0; + for (n = 0; n < N - lag; n++) { + sum += x[n] * x[n+lag]; + } + r[lag] = sum; + } +} + +/*----------------------------------------------------------------* + * window multiplication + *---------------------------------------------------------------*/ + +void window( + float *z, /* (o) the windowed data */ + const float *x, /* (i) the original data vector */ + const float *y, /* (i) the window */ + int N /* (i) length of all vectors */ +){ + int i; + + for (i = 0; i < N; i++) { + z[i] = x[i] * y[i]; + } +} + +/*----------------------------------------------------------------* + * levinson-durbin solution for lpc coefficients + *---------------------------------------------------------------*/ + +void levdurb( + float *a, /* (o) lpc coefficient vector starting with 1.0 +*/ + float *k, /* (o) reflection coefficients */ + float *r, /* (i) autocorrelation vector */ + int order /* (i) order of lpc filter */ +){ + float sum, alpha; + int m, m_h, i; + + a[0] = 1.0; + + if (r[0] < EPS) { /* if r[0] <= 0, set LPC coeff. to zero */ + for (i = 0; i < order; i++) { + k[i] = 0; + a[i+1] = 0; + } + } else { + a[1] = k[0] = -r[1]/r[0]; + alpha = r[0] + r[1] * k[0]; + for (m = 1; m < order; m++){ + sum = r[m + 1]; + for (i = 0; i < m; i++){ + sum += a[i+1] * r[m - i]; + } + k[m] = -sum / alpha; + alpha += k[m] * sum; + m_h = (m + 1) >> 1; + for (i = 0; i < m_h; i++){ + sum = a[i+1] + k[m] * a[m - i]; + a[m - i] += k[m] * a[i+1]; + a[i+1] = sum; + } + a[m+1] = k[m]; + } + } +} + +/*----------------------------------------------------------------* + * interpolation between vectors + *---------------------------------------------------------------*/ + +void interpolate( + float *out, /* (o) the interpolated vector */ + float *in1, /* (i) the first vector for the interpolation */ + float *in2, /* (i) the second vector for the interpolation */ + float coef, /* (i) interpolation weights */ + int length /* (i) length of all vectors */ +){ + int i; + float invcoef; + + invcoef = (float)1.0 - coef; + for (i = 0; i < length; i++) { + out[i] = coef * in1[i] + invcoef * in2[i]; + } +} + +/*----------------------------------------------------------------* + * lpc bandwidth expansion + *---------------------------------------------------------------*/ + +void bwexpand( + float *out, /* (o) the bandwidth expanded lpc coefficients */ + float *in, /* (i) the lpc coefficients before bandwidth + expansion */ + float coef, /* (i) the bandwidth expansion factor */ + int length /* (i) the length of lpc coefficient vectors */ +){ + int i; + float chirp; + + chirp = coef; + + out[0] = in[0]; + for (i = 1; i < length; i++) { + out[i] = chirp * in[i]; + chirp *= coef; + } +} + +/*----------------------------------------------------------------* + * vector quantization + *---------------------------------------------------------------*/ + +void vq( + float *Xq, /* (o) the quantized vector */ + int *index, /* (o) the quantization index */ + const float *CB,/* (i) the vector quantization codebook */ + float *X, /* (i) the vector to quantize */ + int n_cb, /* (i) the number of vectors in the codebook */ + int dim /* (i) the dimension of all vectors */ +){ + int i, j; + int pos, minindex; + float dist, tmp, mindist; + + pos = 0; + mindist = FLOAT_MAX; + minindex = 0; + for (j = 0; j < n_cb; j++) { + dist = X[0] - CB[pos]; + dist *= dist; + for (i = 1; i < dim; i++) { + tmp = X[i] - CB[pos + i]; + dist += tmp*tmp; + } + + if (dist < mindist) { + mindist = dist; + minindex = j; + } + pos += dim; + } + for (i = 0; i < dim; i++) { + Xq[i] = CB[minindex*dim + i]; + } + *index = minindex; +} + +/*----------------------------------------------------------------* + * split vector quantization + *---------------------------------------------------------------*/ + +void SplitVQ( + float *qX, /* (o) the quantized vector */ + int *index, /* (o) a vector of indexes for all vector + codebooks in the split */ + float *X, /* (i) the vector to quantize */ + const float *CB,/* (i) the quantizer codebook */ + int nsplit, /* the number of vector splits */ + const int *dim, /* the dimension of X and qX */ + const int *cbsize /* the number of vectors in the codebook */ +){ + int cb_pos, X_pos, i; + + cb_pos = 0; + X_pos= 0; + for (i = 0; i < nsplit; i++) { + vq(qX + X_pos, index + i, CB + cb_pos, X + X_pos, + cbsize[i], dim[i]); + X_pos += dim[i]; + cb_pos += dim[i] * cbsize[i]; + } +} + +/*----------------------------------------------------------------* + * scalar quantization + *---------------------------------------------------------------*/ + +void sort_sq( + float *xq, /* (o) the quantized value */ + int *index, /* (o) the quantization index */ + float x, /* (i) the value to quantize */ + const float *cb,/* (i) the quantization codebook */ + int cb_size /* (i) the size of the quantization codebook */ +){ + int i; + + if (x <= cb[0]) { + *index = 0; + *xq = cb[0]; + } else { + i = 0; + while ((x > cb[i]) && i < cb_size - 1) { + i++; + } + + if (x > ((cb[i] + cb[i - 1])/2)) { + *index = i; + *xq = cb[i]; + } else { + *index = i - 1; + *xq = cb[i - 1]; + } + } +} + +/*----------------------------------------------------------------* + * check for stability of lsf coefficients + *---------------------------------------------------------------*/ + +int LSF_check( /* (o) 1 for stable lsf vectors and 0 for + nonstable ones */ + float *lsf, /* (i) a table of lsf vectors */ + int dim, /* (i) the dimension of each lsf vector */ + int NoAn /* (i) the number of lsf vectors in the table */ +){ + int k,n,m, Nit=2, change=0,pos; + float tmp; + static float eps=(float)0.039; /* 50 Hz */ + static float eps2=(float)0.0195; + static float maxlsf=(float)3.14; /* 4000 Hz */ + static float minlsf=(float)0.01; /* 0 Hz */ + + /* LSF separation check*/ + + for (n=0;nmaxlsf) { + lsf[pos]=maxlsf; + change=1; + } + } + } + } + + return change; +} + + diff --git a/codecs/ilbc/helpfun.h b/codecs/ilbc/helpfun.h new file mode 100755 index 000000000..be6e727b7 --- /dev/null +++ b/codecs/ilbc/helpfun.h @@ -0,0 +1,93 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + helpfun.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __iLBC_HELPFUN_H +#define __iLBC_HELPFUN_H + +void autocorr( + float *r, /* (o) autocorrelation vector */ + const float *x, /* (i) data vector */ + int N, /* (i) length of data vector */ + int order /* largest lag for calculated autocorrelations */ +); + +void window( + float *z, /* (o) the windowed data */ + const float *x, /* (i) the original data vector */ + const float *y, /* (i) the window */ + int N /* (i) length of all vectors */ +); + +void levdurb( + float *a, /* (o) lpc coefficient vector starting + with 1.0 */ + float *k, /* (o) reflection coefficients */ + float *r, /* (i) autocorrelation vector */ + int order /* (i) order of lpc filter */ +); + +void interpolate( + float *out, /* (o) the interpolated vector */ + float *in1, /* (i) the first vector for the interpolation */ + float *in2, /* (i) the second vector for the interpolation */ + float coef, /* (i) interpolation weights */ + int length /* (i) length of all vectors */ +); + +void bwexpand( + float *out, /* (o) the bandwidth expanded lpc coefficients */ + float *in, /* (i) the lpc coefficients before bandwidth + expansion */ + float coef, /* (i) the bandwidth expansion factor */ + int length /* (i) the length of lpc coefficient vectors */ +); + +void vq( + float *Xq, /* (o) the quantized vector */ + int *index, /* (o) the quantization index */ + const float *CB,/* (i) the vector quantization codebook */ + float *X, /* (i) the vector to quantize */ + int n_cb, /* (i) the number of vectors in the codebook */ + int dim /* (i) the dimension of all vectors */ +); + +void SplitVQ( + float *qX, /* (o) the quantized vector */ + int *index, /* (o) a vector of indexes for all vector + codebooks in the split */ + float *X, /* (i) the vector to quantize */ + const float *CB,/* (i) the quantizer codebook */ + int nsplit, /* the number of vector splits */ + const int *dim, /* the dimension of X and qX */ + const int *cbsize /* the number of vectors in the codebook */ +); + + +void sort_sq( + float *xq, /* (o) the quantized value */ + int *index, /* (o) the quantization index */ + float x, /* (i) the value to quantize */ + const float *cb,/* (i) the quantization codebook */ + int cb_size /* (i) the size of the quantization codebook */ +); + +int LSF_check( /* (o) 1 for stable lsf vectors and 0 for + nonstable ones */ + float *lsf, /* (i) a table of lsf vectors */ + int dim, /* (i) the dimension of each lsf vector */ + int NoAn /* (i) the number of lsf vectors in the table */ +); + +#endif + + diff --git a/codecs/ilbc/hpInput.c b/codecs/ilbc/hpInput.c new file mode 100755 index 000000000..1599c2923 --- /dev/null +++ b/codecs/ilbc/hpInput.c @@ -0,0 +1,58 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + hpInput.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include "constants.h" + +/*----------------------------------------------------------------* + * Input high-pass filter + *---------------------------------------------------------------*/ + +void hpInput( + float *In, /* (i) vector to filter */ + int len, /* (i) length of vector to filter */ + float *Out, /* (o) the resulting filtered vector */ + float *mem /* (i/o) the filter state */ +){ + int i; + float *pi, *po; + + /* all-zero section*/ + + pi = &In[0]; + po = &Out[0]; + for (i=0; i + +#include "iLBC_define.h" +#include "gainquant.h" +#include "getCBvec.h" + +/*----------------------------------------------------------------* + * Convert the codebook indexes to make the search easier + *---------------------------------------------------------------*/ + +void index_conv_enc( + int *index /* (i/o) Codebook indexes */ +){ + int k; + + for (k=1;k=108)&&(index[k]<172)) { + index[k]-=64; + } else if (index[k]>=236) { + index[k]-=128; + } else { + /* ERROR */ + } + } +} + +void index_conv_dec( + int *index /* (i/o) Codebook indexes */ +){ + int k; + + for (k=1;k=44)&&(index[k]<108)) { + index[k]+=64; + } else if ((index[k]>=108)&&(index[k]<128)) { + index[k]+=128; + } else { + /* ERROR */ + } + } +} + +/*----------------------------------------------------------------* + * Construct decoded vector from codebook and gains. + *---------------------------------------------------------------*/ + +void iCBConstruct( + float *decvector, /* (o) Decoded vector */ + int *index, /* (i) Codebook indices */ + int *gain_index,/* (i) Gain quantization indices */ + float *mem, /* (i) Buffer for codevector construction */ + int lMem, /* (i) Length of buffer */ + int veclen, /* (i) Length of vector */ + int nStages /* (i) Number of codebook stages */ +){ + int j,k; + float gain[CB_NSTAGES]; + float cbvec[SUBL]; + + /* gain de-quantization */ + + gain[0] = gaindequant(gain_index[0], 1.0, 32); + if (nStages > 1) { + gain[1] = gaindequant(gain_index[1], + (float)fabs(gain[0]), 16); + } + if (nStages > 2) { + gain[2] = gaindequant(gain_index[2], + (float)fabs(gain[1]), 8); + } + + /* codebook vector construction and construction of + total vector */ + + getCBvec(cbvec, mem, index[0], lMem, veclen); + for (j=0;j 1) { + for (k=1; k +#include + +#include "iLBC_define.h" +#include "gainquant.h" +#include "createCB.h" +#include "filter.h" +#include "constants.h" + +/*----------------------------------------------------------------* + * Search routine for codebook encoding and gain quantization. + *---------------------------------------------------------------*/ + +void iCBSearch( + int *index, /* (o) Codebook indices */ + int *gain_index,/* (o) Gain quantization indices */ + float *intarget,/* (i) Target vector for encoding */ + float *mem, /* (i) Buffer for codebook construction */ + int lMem, /* (i) Length of buffer */ + int lTarget, /* (i) Length of vector */ + int nStages, /* (i) Number of codebook stages */ + float *weightDenum, /* (i) weighting filter coefficients */ + float *weightState, /* (i) weighting filter state */ + int block /* (i) the subblock number */ +){ + int i, j, icount, stage, best_index, range, counter; + float max_measure, gain, measure, crossDot, ftmp; + float gains[CB_NSTAGES]; + float target[SUBL]; + int base_index, sInd, eInd, base_size; + int sIndAug=0, eIndAug=0; + float buf[CB_MEML+SUBL+2*LPC_FILTERORDER]; + float invenergy[CB_EXPAND*128], energy[CB_EXPAND*128]; + float *pp, *ppi=0, *ppo=0, *ppe=0; + float cbvectors[CB_MEML]; + float tene, cene, cvec[SUBL]; + float aug_vec[SUBL]; + + memset(cvec,0,SUBL*sizeof(float)); + + /* Determine size of codebook sections */ + + base_size=lMem-lTarget+1; + + if (lTarget==SUBL) { + base_size=lMem-lTarget+1+lTarget/2; + } + + /* setup buffer for weighting */ + + memcpy(buf,weightState,sizeof(float)*LPC_FILTERORDER); + memcpy(buf+LPC_FILTERORDER,mem,lMem*sizeof(float)); + memcpy(buf+LPC_FILTERORDER+lMem,intarget,lTarget*sizeof(float)); + + /* weighting */ + + AllPoleFilter(buf+LPC_FILTERORDER, weightDenum, + lMem+lTarget, LPC_FILTERORDER); + + /* Construct the codebook and target needed */ + + memcpy(target, buf+LPC_FILTERORDER+lMem, lTarget*sizeof(float)); + + tene=0.0; + for (i=0;i0.0) { + invenergy[0] = (float) 1.0 / (*ppe + EPS); + } else { + invenergy[0] = (float) 0.0; + } + ppe++; + + measure=(float)-10000000.0; + + if (crossDot > 0.0) { + measure = crossDot*crossDot*invenergy[0]; + } + } + else { + measure = crossDot*crossDot*invenergy[0]; + } + + /* check if measure is better */ + ftmp = crossDot*invenergy[0]; + + if ((measure>max_measure) && (fabs(ftmp)0.0) { + invenergy[icount] = + (float)1.0/(energy[icount]+EPS); + } else { + invenergy[icount] = (float) 0.0; + } + + measure=(float)-10000000.0; + + if (crossDot > 0.0) { + measure = crossDot*crossDot*invenergy[icount]; + } + } + else { + measure = crossDot*crossDot*invenergy[icount]; + } + + /* check if measure is better */ + ftmp = crossDot*invenergy[icount]; + + + if ((measure>max_measure) && (fabs(ftmp) range) { + sInd -= (eInd-range); + eInd = range; + } + } else { /* base_index >= (base_size-20) */ + + if(sInd < (base_size-20)) { + sIndAug = 20; + sInd = 0; + eInd = 0; + eIndAug = 19 + CB_RESRANGE; + + if(eIndAug > 39) { + eInd = eIndAug-39; + eIndAug = 39; + } + } else { + sIndAug = 20 + sInd - (base_size-20); + eIndAug = 39; + sInd = 0; + eInd = CB_RESRANGE - (eIndAug-sIndAug+1); + } + } + + } else { /* lTarget = 22 */ + + if (sInd < 0) { + eInd -= sInd; + sInd = 0; + } + + if(eInd > range) { + sInd -= (eInd - range); + eInd = range; + } + } + } + + /* search of higher codebook section */ + + /* index search range */ + counter = sInd; + sInd += base_size; + eInd += base_size; + + + if(stage==0) { + ppe = energy+base_size; + *ppe=0.0; + + pp=cbvectors+lMem-lTarget; + for (j=0; j0.0) { + invenergy[icount] = (float) 1.0/(energy[icount]+EPS); + } else { + invenergy[icount] = (float) 0.0; + } + + if (stage==0) { + + measure=(float)-10000000.0; + + if (crossDot > 0.0) { + measure = crossDot*crossDot* + invenergy[icount]; + } + } + else { + measure = crossDot*crossDot*invenergy[icount]; + } + + /* check if measure is better */ + ftmp = crossDot*invenergy[icount]; + + if ((measure>max_measure) && (fabs(ftmp)CB_MAXGAIN) { + gain = (float)CB_MAXGAIN; + } + gain = gainquant(gain, 1.0, 32, &gain_index[stage]); + } + else { + if (stage==1) { + gain = gainquant(gain, (float)fabs(gains[stage-1]), + 16, &gain_index[stage]); + } else { + gain = gainquant(gain, (float)fabs(gains[stage-1]), + 8, &gain_index[stage]); + } + } + + /* Extract the best (according to measure) codebook vector */ + + if(lTarget==(STATE_LEN-STATE_SHORT_LEN)) { + + if(index[stage] +#include + +#include "iLBC_define.h" +#include "StateConstructW.h" +#include "LPCdecode.h" +#include "iCBConstruct.h" +#include "doCPLC.h" +#include "helpfun.h" +#include "constants.h" +#include "packing.h" +#include "string.h" +#include "enhancer.h" +#include "hpOutput.h" +#include "syntFilter.h" + +/*----------------------------------------------------------------* + * Initiation of decoder instance. + *---------------------------------------------------------------*/ + +short initDecode( /* (o) Number of decoded + samples */ + iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */ + int use_enhancer /* (i) 1 to use enhancer + 0 to run without + enhancer */ +){ + int i; + + memset((*iLBCdec_inst).syntMem, 0, + LPC_FILTERORDER*sizeof(float)); + memcpy((*iLBCdec_inst).lsfdeqold, lsfmeanTbl, + LPC_FILTERORDER*sizeof(float)); + + memset((*iLBCdec_inst).old_syntdenum, 0, + ((LPC_FILTERORDER + 1)*NSUB)*sizeof(float)); + for (i=0; iprev_enh_pl = 0; + + return (BLOCKL); +} + +/*----------------------------------------------------------------* + * frame residual decoder function (subrutine to iLBC_decode) + *---------------------------------------------------------------*/ + +void Decode( + float *decresidual, /* (o) decoded residual frame */ + int start, /* (i) location of start state */ + int idxForMax, /* (i) codebook index for the maximum + value */ + int *idxVec, /* (i) codebook indexes for the samples + in the start state */ + float *syntdenum, /* (i) the decoded synthesis filter + coefficients */ + int *cb_index, /* (i) the indexes for the adaptive + codebook */ + int *gain_index, /* (i) the indexes for the corresponding + gains */ + int *extra_cb_index,/* (i) the indexes for the adaptive + codebook part of start state */ + int *extra_gain_index, /* (i) the indexes for the corresponding + gains */ + int state_first /* (i) 1 if non adaptive part of start + state comes first 0 if that part + comes last */ +){ + float reverseDecresidual[BLOCKL], mem[CB_MEML]; + int k, meml_gotten, Nfor, Nback, i; + int diff, start_pos; + int subcount, subframe; + + diff = STATE_LEN - STATE_SHORT_LEN; + + if (state_first == 1) { + start_pos = (start-1)*SUBL; + } else { + start_pos = (start-1)*SUBL + diff; + } + + /* decode scalar part of start state */ + + StateConstructW(idxForMax, idxVec, + &syntdenum[(start-1)*(LPC_FILTERORDER+1)], + &decresidual[start_pos], STATE_SHORT_LEN); + + + if (state_first) { /* put adaptive part in the end */ + + /* setup memory */ + + memset(mem, 0, (CB_MEML-STATE_SHORT_LEN)*sizeof(float)); + memcpy(mem+CB_MEML-STATE_SHORT_LEN, decresidual+start_pos, + STATE_SHORT_LEN*sizeof(float)); + + /* construct decoded vector */ + + iCBConstruct(&decresidual[start_pos+STATE_SHORT_LEN], + extra_cb_index, extra_gain_index, mem+CB_MEML-stMemLTbl, + stMemLTbl, diff, CB_NSTAGES); + + } + else {/* put adaptive part in the beginning */ + + /* create reversed vectors for prediction */ + + for(k=0; k 0 ){ + + /* setup memory */ + + memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float)); + memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL, + STATE_LEN*sizeof(float)); + + /* loop over subframes to encode */ + + for (subframe=0; subframe 0 ){ + + /* setup memory */ + + meml_gotten = SUBL*(NSUB+1-start); + + + if( meml_gotten > CB_MEML ) { + meml_gotten=CB_MEML; + } + for( k=0; k0) { /* the data are good */ + + /* decode data */ + + pbytes=bytes; + pos=0; + + /* Set everything to zero before decoding */ + + for (k=0;k<6;k++) { + lsf_i[k]=0; + } + start=0; + state_first=0; + idxForMax=0; + for (k=0; k5) ) + mode = 0; + + if (mode==1) { /* No bit errors was detected, + continue decoding */ + + /* adjust index */ + index_conv_dec(cb_index); + + /* decode the lsf */ + + SimplelsfDEQ(lsfdeq, lsf_i); + check=LSF_check(lsfdeq, LPC_FILTERORDER, LPC_N); + DecoderInterpolateLSF(syntdenum, weightdenum, + lsfdeq, LPC_FILTERORDER, iLBCdec_inst); + + Decode(decresidual, start, idxForMax, idxVec, + syntdenum, cb_index, gain_index, + extra_cb_index, extra_gain_index, + state_first); + + /* preparing the plc for a future loss! */ + + doThePLC(PLCresidual, PLClpc, 0, decresidual, + syntdenum + (LPC_FILTERORDER + 1)*(NSUB - 1), + (*iLBCdec_inst).last_lag, iLBCdec_inst); + + + memcpy(decresidual, PLCresidual, BLOCKL*sizeof(float)); + } + + } + + if (mode == 0) { + /* the data is bad (either a PLC call + * was made or a bit error was detected) + */ + + /* packet loss conceal */ + + memset(zeros, 0, BLOCKL*sizeof(float)); + + one[0] = 1; + memset(one+1, 0, LPC_FILTERORDER*sizeof(float)); + + start=0; + + doThePLC(PLCresidual, PLClpc, 1, zeros, one, + (*iLBCdec_inst).last_lag, iLBCdec_inst); + memcpy(decresidual, PLCresidual, BLOCKL*sizeof(float)); + + order_plus_one = LPC_FILTERORDER + 1; + for (i = 0; i < NSUB; i++) { + memcpy(syntdenum+(i*order_plus_one), PLClpc, + order_plus_one*sizeof(float)); + } + } + + if ((*iLBCdec_inst).use_enhancer == 1) { + + /* post filtering */ + + (*iLBCdec_inst).last_lag = + enhancerInterface(data, decresidual, iLBCdec_inst); + + /* synthesis filtering */ + + for (i=0; i < 2; i++) { + syntFilter(data + i*SUBL, + (*iLBCdec_inst).old_syntdenum + + (i+4)*(LPC_FILTERORDER+1), SUBL, + (*iLBCdec_inst).syntMem); + } + for (i=2; i < NSUB; i++) { + syntFilter(data + i*SUBL, + syntdenum + (i-2)*(LPC_FILTERORDER+1), SUBL, + (*iLBCdec_inst).syntMem); + } + + } else { + + /* Find last lag */ + lag = 20; + maxcc = xCorrCoef(&decresidual[BLOCKL-ENH_BLOCKL], + &decresidual[BLOCKL-ENH_BLOCKL-lag], ENH_BLOCKL); + + for (ilag=21; ilag<120; ilag++) { + cc = xCorrCoef(&decresidual[BLOCKL-ENH_BLOCKL], + &decresidual[BLOCKL-ENH_BLOCKL-ilag], ENH_BLOCKL); + + if (cc > maxcc) { + maxcc = cc; + lag = ilag; + } + } + (*iLBCdec_inst).last_lag = lag; + + /* copy data and run synthesis filter */ + + memcpy(data, decresidual, BLOCKL*sizeof(float)); + for (i=0; i < NSUB; i++) { + syntFilter(data + i*SUBL, + syntdenum + i*(LPC_FILTERORDER+1), SUBL, + (*iLBCdec_inst).syntMem); + } + } + + /* high pass filtering on output if desired, otherwise + copy to out */ + + /*hpOutput(data, BLOCKL, decblock, (*iLBCdec_inst).hpomem);*/ + memcpy(decblock,data,BLOCKL*sizeof(float)); + + memcpy((*iLBCdec_inst).old_syntdenum, syntdenum, + NSUB*(LPC_FILTERORDER+1)*sizeof(float)); + + iLBCdec_inst->prev_enh_pl=0; + + if (mode==0) { /* PLC was used */ + iLBCdec_inst->prev_enh_pl=1; + } +} + + diff --git a/codecs/ilbc/iLBC_decode.h b/codecs/ilbc/iLBC_decode.h new file mode 100755 index 000000000..70e992619 --- /dev/null +++ b/codecs/ilbc/iLBC_decode.h @@ -0,0 +1,38 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + iLBC_decode.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __iLBC_ILBCDECODE_H +#define __iLBC_ILBCDECODE_H + +#include "iLBC_define.h" + +short initDecode( /* (o) Number of decoded + samples */ + iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */ + int use_enhancer /* (i) 1 to use enhancer + 0 to run without + enhancer */ +); + +void iLBC_decode( + float *decblock, /* (o) decoded signal block */ + unsigned char *bytes, /* (i) encoded signal bits */ + iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state + structure */ + int mode /* (i) 0: bad packet, PLC, + 1: normal */ +); + +#endif + + diff --git a/codecs/ilbc/iLBC_define.h b/codecs/ilbc/iLBC_define.h new file mode 100755 index 000000000..e1c821b3d --- /dev/null +++ b/codecs/ilbc/iLBC_define.h @@ -0,0 +1,157 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + iLBC_define.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ +#include + +#ifndef __iLBC_ILBCDEFINE_H +#define __iLBC_ILBCDEFINE_H + +/* general codec settings */ + +#define FS (float)8000.0 +#define BLOCKL 240 +#define NSUB 6 +#define NASUB 4 +#define SUBL 40 +#define STATE_LEN 80 +#define STATE_SHORT_LEN 58 + +/* LPC settings */ + +#define LPC_FILTERORDER 10 +#define LPC_CHIRP_SYNTDENUM (float)0.9025 +#define LPC_CHIRP_WEIGHTDENUM (float)0.4222 +#define LPC_LOOKBACK 60 +#define LPC_N 2 +#define LPC_ASYMDIFF 20 +#define LPC_BW (float)60.0 +#define LPC_WN (float)1.0001 +#define LSF_NSPLIT 3 +#define LSF_NUMBER_OF_STEPS 4 +#define LPC_HALFORDER LPC_FILTERORDER/2 + +/* cb settings */ + +#define CB_NSTAGES 3 +#define CB_EXPAND 2 +#define CB_MEML 147 +#define CB_FILTERLEN 2*4 +#define CB_HALFFILTERLEN 4 +#define CB_RESRANGE 34 +#define CB_MAXGAIN (float) 1.3 + +/* enhancer */ + +#define ENH_BLOCKL 80 /* block length */ +#define ENH_BLOCKL_HALF (ENH_BLOCKL/2) +#define ENH_HL 3 /* 2*ENH_HL+1 is number blocks + in said second sequence */ +#define ENH_SLOP 2 /* max difference estimated and + correct pitch period */ +#define ENH_PLOCSL 20 /* pitch-estimates and + pitch-locations buffer length */ +#define ENH_OVERHANG 2 +#define ENH_UPS0 4 /* upsampling rate */ +#define ENH_FL0 3 /* 2*FLO+1 is the length of each filter */ +#define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0) +#define ENH_CORRDIM (2*ENH_SLOP+1) +#define ENH_NBLOCKS (BLOCKL/ENH_BLOCKL) +#define ENH_NBLOCKS_EXTRA 5 +#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */ +#define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL +#define ENH_ALPHA0 (float)0.05 + +/* PLC */ + +#define PLC_BFIATTENUATE (float)0.9 +#define PLC_GAINTHRESHOLD (float)0.5 +#define PLC_BWEXPAND (float)0.99 +#define PLC_XT_MIX (float)1.0 +#define PLC_XB_MIX (float)0.0 +#define PLC_YT_MIX (float)0.95 +#define PLC_YB_MIX (float)0.0 + +/* Down sampling */ + +#define FILTERORDER_DS 7 +#define DELAY_DS 3 +#define FACTOR_DS 2 + +/* bit stream defs */ + +#define NO_OF_BYTES 50 +#define STATE_BITS 3 +#define BYTE_LEN 8 +#define ULP_CLASSES 3 + +/* help parameters */ + +#define FLOAT_MAX (float)1.0e37 +#define EPS (float)2.220446049250313e-016 +#define PI (float)3.14159265358979323846 +#define MIN_SAMPLE -32768 +#define MAX_SAMPLE 32767 +#define TWO_PI (float)6.283185307 +#define PI2 (float)0.159154943 + +/* type definition encoder instance */ +typedef struct iLBC_Enc_Inst_t_ { + + /* analysis filter state */ + float anaMem[LPC_FILTERORDER]; + + /* old lsf parameters for interpolation */ + float lsfold[LPC_FILTERORDER]; + float lsfdeqold[LPC_FILTERORDER]; + + /* signal buffer for LP analysis */ + float lpc_buffer[LPC_LOOKBACK + BLOCKL]; + + /* state of input HP filter */ + float hpimem[4]; + +} iLBC_Enc_Inst_t; + +/* type definition decoder instance */ +typedef struct iLBC_Dec_Inst_t_ { + /* synthesis filter state */ + float syntMem[LPC_FILTERORDER]; + + /* old LSF for interpolation */ + float lsfdeqold[LPC_FILTERORDER]; + + /* pitch lag estimated in enhancer and used in PLC */ + int last_lag; + + /* PLC state information */ + int prevLag, consPLICount, prevPLI, prev_enh_pl; + float prevGain, prevLpc[LPC_FILTERORDER+1]; + float prevResidual[NSUB*SUBL]; + float energy; + unsigned long seed; + + /* previous synthesis filter parameters */ + float old_syntdenum[(LPC_FILTERORDER + 1)*NSUB]; + + /* state of output HP filter */ + float hpomem[4]; + + /* enhancer state information */ + int use_enhancer; + float enh_buf[ENH_BUFL]; + float enh_period[ENH_NBLOCKS_TOT]; + +} iLBC_Dec_Inst_t; + +#endif + + diff --git a/codecs/ilbc/iLBC_encode.c b/codecs/ilbc/iLBC_encode.c new file mode 100755 index 000000000..6d8e04345 --- /dev/null +++ b/codecs/ilbc/iLBC_encode.c @@ -0,0 +1,446 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + iLBC_encode.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include +#include + +#include "iLBC_define.h" +#include "LPCencode.h" +#include "FrameClassify.h" +#include "StateSearchW.h" +#include "StateConstructW.h" +#include "helpfun.h" +#include "constants.h" +#include "packing.h" +#include "iCBSearch.h" +#include "iCBConstruct.h" +#include "hpInput.h" +#include "anaFilter.h" +#include "syntFilter.h" + +/*----------------------------------------------------------------* + * Initiation of encoder instance. + *---------------------------------------------------------------*/ + +short initEncode( /* (o) Number of bytes encoded */ + iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) Encoder instance */ +){ + memset((*iLBCenc_inst).anaMem, 0, + LPC_FILTERORDER*sizeof(float)); + memcpy((*iLBCenc_inst).lsfold, lsfmeanTbl, + LPC_FILTERORDER*sizeof(float)); + memcpy((*iLBCenc_inst).lsfdeqold, lsfmeanTbl, + LPC_FILTERORDER*sizeof(float)); + memset((*iLBCenc_inst).lpc_buffer, 0, + LPC_LOOKBACK*sizeof(float)); + memset((*iLBCenc_inst).hpimem, 0, 4*sizeof(float)); + + return (NO_OF_BYTES); +} + +/*----------------------------------------------------------------* + * main encoder function + *---------------------------------------------------------------*/ + +void iLBC_encode( + unsigned char *bytes, /* (o) encoded data bits iLBC */ + float *block, /* (o) speech vector to encode */ + iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder + state */ +){ + + float data[BLOCKL]; + float residual[BLOCKL], reverseResidual[BLOCKL]; + + int start, idxForMax, idxVec[STATE_LEN]; + float reverseDecresidual[BLOCKL], mem[CB_MEML]; + int n, k, meml_gotten, Nfor, Nback, i, pos; + int gain_index[CB_NSTAGES*NASUB], extra_gain_index[CB_NSTAGES]; + int cb_index[CB_NSTAGES*NASUB],extra_cb_index[CB_NSTAGES]; + int lsf_i[LSF_NSPLIT*LPC_N]; + unsigned char *pbytes; + int diff, start_pos, state_first; + float en1, en2; + int index, ulp, firstpart; + int subcount, subframe; + float weightState[LPC_FILTERORDER]; + float syntdenum[NSUB*(LPC_FILTERORDER+1)]; + float weightdenum[NSUB*(LPC_FILTERORDER+1)]; + float decresidual[BLOCKL]; + + /* high pass filtering of input signal if such is not done + prior to calling this function */ + + /*hpInput(block, BLOCKL, data, (*iLBCenc_inst).hpimem);*/ + + /* otherwise simply copy */ + + memcpy(data,block,BLOCKL*sizeof(float)); + + /* LPC of hp filtered input data */ + + LPCencode(syntdenum, weightdenum, lsf_i, data, + iLBCenc_inst); + + /* inverse filter to get residual */ + + for (n=0; n en2) { + state_first = 1; + start_pos = (start-1)*SUBL; + } else { + state_first = 0; + start_pos = (start-1)*SUBL + diff; + } + + /* scalar quantization of state */ + + StateSearchW(&residual[start_pos], + &syntdenum[(start-1)*(LPC_FILTERORDER+1)], + &weightdenum[(start-1)*(LPC_FILTERORDER+1)], &idxForMax, + idxVec, STATE_SHORT_LEN, state_first); + + StateConstructW(idxForMax, idxVec, + &syntdenum[(start-1)*(LPC_FILTERORDER+1)], + &decresidual[start_pos], STATE_SHORT_LEN); + + /* predictive quantization in state */ + + if (state_first) { /* put adaptive part in the end */ + + /* setup memory */ + + memset(mem, 0, (CB_MEML-STATE_SHORT_LEN)*sizeof(float)); + memcpy(mem+CB_MEML-STATE_SHORT_LEN, decresidual+start_pos, + STATE_SHORT_LEN*sizeof(float)); + memset(weightState, 0, LPC_FILTERORDER*sizeof(float)); + + /* encode subframes */ + + iCBSearch(extra_cb_index, extra_gain_index, + &residual[start_pos+STATE_SHORT_LEN], + mem+CB_MEML-stMemLTbl, + stMemLTbl, diff, CB_NSTAGES, + &weightdenum[start*(LPC_FILTERORDER+1)], weightState, 0); + + /* construct decoded vector */ + + iCBConstruct(&decresidual[start_pos+STATE_SHORT_LEN], + extra_cb_index, extra_gain_index, mem+CB_MEML-stMemLTbl, + stMemLTbl, diff, CB_NSTAGES); + + } + else { /* put adaptive part in the beginning */ + + /* create reversed vectors for prediction */ + + for(k=0; k 0 ){ + + /* setup memory */ + + memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float)); + memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL, + STATE_LEN*sizeof(float)); + memset(weightState, 0, LPC_FILTERORDER*sizeof(float)); + + /* loop over subframes to encode */ + + for (subframe=0; subframe 0 ){ + + /* create reverse order vectors */ + + for( n=0; n CB_MEML ) { + meml_gotten=CB_MEML; + } + for( k=0; k +#include + +#include "iLBC_define.h" + +/*----------------------------------------------------------------* + * conversion from lpc coefficients to lsf coefficients + *---------------------------------------------------------------*/ + +void a2lsf( + float *freq,/* (o) lsf coefficients */ + float *a /* (i) lpc coefficients */ +){ + float steps[LSF_NUMBER_OF_STEPS] = + {(float)0.00635, (float)0.003175, (float)0.0015875, + (float)0.00079375}; + float step; + int step_idx; + int lsp_index; + float p[LPC_HALFORDER]; + float q[LPC_HALFORDER]; + float p_pre[LPC_HALFORDER]; + float q_pre[LPC_HALFORDER]; + float old_p, old_q, *old; + float *pq_coef; + float omega, old_omega; + int i; + float hlp, hlp1, hlp2, hlp3, hlp4, hlp5; + + for (i = 0; i < LPC_HALFORDER; i++){ + p[i] = (float)-1.0 * (a[i + 1] + a[LPC_FILTERORDER - i]); + q[i] = a[LPC_FILTERORDER - i] - a[i + 1]; + } + + p_pre[0] = (float)-1.0 - p[0]; + p_pre[1] = - p_pre[0] - p[1]; + p_pre[2] = - p_pre[1] - p[2]; + p_pre[3] = - p_pre[2] - p[3]; + p_pre[4] = - p_pre[3] - p[4]; + p_pre[4] = p_pre[4] / 2; + + q_pre[0] = (float)1.0 - q[0]; + q_pre[1] = q_pre[0] - q[1]; + q_pre[2] = q_pre[1] - q[2]; + q_pre[3] = q_pre[2] - q[3]; + q_pre[4] = q_pre[3] - q[4]; + q_pre[4] = q_pre[4] / 2; + + omega = 0.0; + old_omega = 0.0; + + old_p = FLOAT_MAX; + old_q = FLOAT_MAX; + + /* Here we loop through lsp_index to find all the + LPC_FILTERORDER roots for omega. */ + + for (lsp_index = 0; lsp_index < LPC_FILTERORDER; lsp_index++){ + + /* Depending on lsp_index being even or odd, we + alternatively solve the roots for the two LSP equations. */ + + + if ((lsp_index & 0x1) == 0) { + pq_coef = p_pre; + old = &old_p; + } else { + pq_coef = q_pre; + old = &old_q; + } + + /* Start with low resolution grid */ + + for (step_idx = 0, step = steps[step_idx]; + step_idx < LSF_NUMBER_OF_STEPS;){ + + /* cos(10piw) + pq(0)cos(8piw) + pq(1)cos(6piw) + + pq(2)cos(4piw) + pq(3)cod(2piw) + pq(4) */ + + hlp = (float)cos(omega * TWO_PI); + hlp1 = (float)2.0 * hlp + pq_coef[0]; + hlp2 = (float)2.0 * hlp * hlp1 - (float)1.0 + + pq_coef[1]; + hlp3 = (float)2.0 * hlp * hlp2 - hlp1 + pq_coef[2]; + hlp4 = (float)2.0 * hlp * hlp3 - hlp2 + pq_coef[3]; + hlp5 = hlp * hlp4 - hlp3 + pq_coef[4]; + + + if (((hlp5 * (*old)) <= 0.0) || (omega >= 0.5)){ + + if (step_idx == (LSF_NUMBER_OF_STEPS - 1)){ + + if (fabs(hlp5) >= fabs(*old)) { + freq[lsp_index] = omega - step; + } else { + freq[lsp_index] = omega; + } + + + if ((*old) >= 0.0){ + *old = (float)-1.0 * FLOAT_MAX; + } else { + *old = FLOAT_MAX; + } + + omega = old_omega; + step_idx = 0; + + step_idx = LSF_NUMBER_OF_STEPS; + } else { + + if (step_idx == 0) { + old_omega = omega; + } + + step_idx++; + omega -= steps[step_idx]; + + /* Go back one grid step */ + + step = steps[step_idx]; + } + } else { + + /* increment omega until they are of different sign, + and we know there is at least one root between omega + and old_omega */ + *old = hlp5; + omega += step; + } + } + } + + for (i = 0; i < LPC_FILTERORDER; i++) { + freq[i] = freq[i] * TWO_PI; + } +} + +/*----------------------------------------------------------------* + * conversion from lsf coefficients to lpc coefficients + *---------------------------------------------------------------*/ + +void lsf2a( + float *a_coef, /* (o) lpc coefficients */ + float *freq /* (i) lsf coefficients */ +){ + int i, j; + float hlp; + float p[LPC_HALFORDER], q[LPC_HALFORDER]; + float a[LPC_HALFORDER + 1], a1[LPC_HALFORDER], a2[LPC_HALFORDER]; + float b[LPC_HALFORDER + 1], b1[LPC_HALFORDER], b2[LPC_HALFORDER]; + + for (i = 0; i < LPC_FILTERORDER; i++) { + freq[i] = freq[i] * PI2; + } + + /* Check input for ill-conditioned cases. This part is not + found in the TIA standard. It involves the following 2 IF + blocks. If "freq" is judged ill-conditioned, then we first + modify freq[0] and freq[LPC_HALFORDER-1] (normally + LPC_HALFORDER = 10 for LPC applications), then we adjust + the other "freq" values slightly */ + + + if ((freq[0] <= 0.0) || (freq[LPC_FILTERORDER - 1] >= 0.5)){ + + + if (freq[0] <= 0.0) { + freq[0] = (float)0.022; + } + + + if (freq[LPC_FILTERORDER - 1] >= 0.5) { + freq[LPC_FILTERORDER - 1] = (float)0.499; + } + + hlp = (freq[LPC_FILTERORDER - 1] - freq[0]) / + (float) (LPC_FILTERORDER - 1); + + for (i = 1; i < LPC_FILTERORDER; i++) { + freq[i] = freq[i - 1] + hlp; + } + } + + memset(a1, 0, LPC_HALFORDER*sizeof(float)); + memset(a2, 0, LPC_HALFORDER*sizeof(float)); + memset(b1, 0, LPC_HALFORDER*sizeof(float)); + memset(b2, 0, LPC_HALFORDER*sizeof(float)); + memset(a, 0, (LPC_HALFORDER+1)*sizeof(float)); + memset(b, 0, (LPC_HALFORDER+1)*sizeof(float)); + + /* p[i] and q[i] compute cos(2*pi*omega_{2j}) and + cos(2*pi*omega_{2j-1} in eqs. 4.2.2.2-1 and 4.2.2.2-2. + Note that for this code p[i] specifies the coefficients + used in .Q_A(z) while q[i] specifies the coefficients used + in .P_A(z) */ + + for (i = 0; i < LPC_HALFORDER; i++){ + p[i] = (float)cos(TWO_PI * freq[2 * i]); + q[i] = (float)cos(TWO_PI * freq[2 * i + 1]); + } + + a[0] = 0.25; + b[0] = 0.25; + + for (i = 0; i < LPC_HALFORDER; i++){ + a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i]; + b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i]; + a2[i] = a1[i]; + a1[i] = a[i]; + b2[i] = b1[i]; + b1[i] = b[i]; + } + + for (j = 0; j < LPC_FILTERORDER; j++){ + + if (j == 0) { + a[0] = 0.25; + b[0] = -0.25; + } else { + a[0] = b[0] = 0.0; + } + + for (i = 0; i < LPC_HALFORDER; i++){ + a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i]; + b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i]; + a2[i] = a1[i]; + a1[i] = a[i]; + b2[i] = b1[i]; + b1[i] = b[i]; + } + + a_coef[j + 1] = 2 * (a[LPC_HALFORDER] + b[LPC_HALFORDER]); + } + + a_coef[0] = 1.0; +} + + diff --git a/codecs/ilbc/lsf.h b/codecs/ilbc/lsf.h new file mode 100755 index 000000000..825685d41 --- /dev/null +++ b/codecs/ilbc/lsf.h @@ -0,0 +1,29 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + lsf.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __iLBC_LSF_H +#define __iLBC_LSF_H + +void a2lsf( + float *freq,/* (o) lsf coefficients */ + float *a /* (i) lpc coefficients */ +); + +void lsf2a( + float *a_coef, /* (o) lpc coefficients */ + float *freq /* (i) lsf coefficients */ +); + +#endif + + diff --git a/codecs/ilbc/packing.c b/codecs/ilbc/packing.c new file mode 100755 index 000000000..6449e55ff --- /dev/null +++ b/codecs/ilbc/packing.c @@ -0,0 +1,169 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + packing.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include +#include + +#include "iLBC_define.h" +#include "constants.h" +#include "helpfun.h" +#include "string.h" + +/*----------------------------------------------------------------* + * splitting an integer into first most significant bits and + * remaining least significant bits + *---------------------------------------------------------------*/ + +void packsplit( + int *index, /* (i) the value to split */ + int *firstpart, /* (o) the value specified by most + significant bits */ + int *rest, /* (o) the value specified by least + significant bits */ + int bitno_firstpart, /* (i) number of bits in most + significant part */ + int bitno_total /* (i) number of bits in full range + of value */ +){ + int bitno_rest = bitno_total-bitno_firstpart; + + *firstpart = *index>>(bitno_rest); + *rest = *index-(*firstpart<<(bitno_rest)); +} + +/*----------------------------------------------------------------* + * combining a value corresponding to msb's with a value + * corresponding to lsb's + *---------------------------------------------------------------*/ + +void packcombine( + int *index, /* (i/o) the msb value in the + combined value out */ + int rest, /* (i) the lsb value */ + int bitno_rest /* (i) the number of bits in the + lsb part */ +){ + *index = *index<0) { + + /* Jump to the next byte if end of this byte is reached*/ + + if (*pos==8) { + *pos=0; + (*bitstream)++; + **bitstream=0; + } + + posLeft=8-(*pos); + + /* Insert index into the bitstream */ + + if (bitno <= posLeft) { + **bitstream |= (unsigned char)(index<<(posLeft-bitno)); + *pos+=bitno; + bitno=0; + } else { + **bitstream |= (unsigned char)(index>>(bitno-posLeft)); + + *pos=8; + index-=((index>>(bitno-posLeft))<<(bitno-posLeft)); + + bitno-=posLeft; + } + } +} + +/*----------------------------------------------------------------* + * unpacking of bits from bitstream, i.e., vector of bytes + *---------------------------------------------------------------*/ + +void unpack( + unsigned char **bitstream, /* (i/o) on entrance pointer to + place in bitstream to + unpack new data from, on + exit pointer to place in + bitstream to unpack future + data from */ + int *index, /* (o) resulting value */ + int bitno, /* (i) number of bits used to + represent the value */ + int *pos /* (i/o) read position in the + current byte */ +){ + int BitsLeft; + + *index=0; + + while (bitno>0) { + + /* move forward in bitstream when the end of the + byte is reached */ + + if (*pos==8) { + *pos=0; + (*bitstream)++; + } + + BitsLeft=8-(*pos); + + /* Extract bits to index */ + + if (BitsLeft>=bitno) { + *index+=((((**bitstream)<<(*pos)) & 0xFF)>>(8-bitno)); + + *pos+=bitno; + bitno=0; + } else { + + if ((8-bitno)>0) { + *index+=((((**bitstream)<<(*pos)) & 0xFF)>> + (8-bitno)); + *pos=8; + } else { + *index+=(((int)(((**bitstream)<<(*pos)) & 0xFF))<< + (bitno-8)); + *pos=8; + } + bitno-=BitsLeft; + } + } +} + + diff --git a/codecs/ilbc/packing.h b/codecs/ilbc/packing.h new file mode 100755 index 000000000..21261e82d --- /dev/null +++ b/codecs/ilbc/packing.h @@ -0,0 +1,65 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + packing.h + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#ifndef __PACKING_H +#define __PACKING_H + +void packsplit( + int *index, /* (i) the value to split */ + int *firstpart, /* (o) the value specified by most + significant bits */ + int *rest, /* (o) the value specified by least + significant bits */ + int bitno_firstpart, /* (i) number of bits in most + significant part */ + int bitno_total /* (i) number of bits in full range + of value */ +); + +void packcombine( + int *index, /* (i/o) the msb value in the + combined value out */ + int rest, /* (i) the lsb value */ + int bitno_rest /* (i) the number of bits in the + lsb part */ +); + +void dopack( + unsigned char **bitstream, /* (i/o) on entrance pointer to place + in bitstream to pack new data, + on exit pointer to place in + bitstream to pack future + data */ + int index, /* (i) the value to pack */ + int bitno, /* (i) the number of bits that the + value will fit within */ + int *pos /* (i/o) write position in the + current byte */ +); + +void unpack( + unsigned char **bitstream, /* (i/o) on entrance pointer to place + in bitstream to unpack + new data from, on exit pointer + to place in bitstream to + unpack future data from */ + int *index, /* (o) resulting value */ + int bitno, /* (i) number of bits used to + represent the value */ + int *pos /* (i/o) read position in the + current byte */ +); + +#endif + + diff --git a/codecs/ilbc/syntFilter.c b/codecs/ilbc/syntFilter.c new file mode 100755 index 000000000..fe61eaa46 --- /dev/null +++ b/codecs/ilbc/syntFilter.c @@ -0,0 +1,65 @@ + +/****************************************************************** + + iLBC Speech Coder ANSI-C Source Code + + syntFilter.c + + Copyright (c) 2001, + Global IP Sound AB. + All rights reserved. + +******************************************************************/ + +#include "iLBC_define.h" + +/*----------------------------------------------------------------* + * LP synthesis filter. + *---------------------------------------------------------------*/ + +void syntFilter( + float *Out, /* (i/o) Signal to be filtered */ + float *a, /* (i) LP parameters */ + int len, /* (i) Length of signal */ + float *mem /* (i/o) Filter state */ +){ + int i, j; + float *po, *pi, *pa, *pm; + + po=Out; + + /* Filter first part using memory from past */ + + for (i=0;i #include #include +#include #include #define TYPE_SILENCE 0x2 @@ -332,6 +333,9 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp) case AST_FORMAT_GSM: rtp->f.samples = 160 * (rtp->f.datalen / 33); break; + case AST_FORMAT_ILBC: + rtp->f.samples = 240 * (rtp->f.datalen / 52); + break; case AST_FORMAT_ADPCM: rtp->f.samples = rtp->f.datalen * 2; break; @@ -370,6 +374,7 @@ static struct { {{1, AST_FORMAT_LPC10}, "audio", "LPC"}, {{1, AST_FORMAT_G729A}, "audio", "G729"}, {{1, AST_FORMAT_SPEEX}, "audio", "SPEEX"}, + {{1, AST_FORMAT_ILBC}, "audio", "iLBC"}, {{0, AST_RTP_DTMF}, "audio", "telephone-event"}, {{0, AST_RTP_CN}, "audio", "CN"}, {{1, AST_FORMAT_JPEG}, "video", "JPEG"}, @@ -401,6 +406,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = { [34] = {1, AST_FORMAT_H263}, [101] = {0, AST_RTP_DTMF}, [110] = {1, AST_FORMAT_SPEEX}, + [120] = {1, AST_FORMAT_ILBC}, }; void ast_rtp_pt_clear(struct ast_rtp* rtp) @@ -710,6 +716,9 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec case AST_FORMAT_GSM: pred = rtp->lastts + (f->datalen * 160 / 33); break; + case AST_FORMAT_ILBC: + pred = rtp->lastts + (f->datalen * 240 / 52); + break; case AST_FORMAT_G723_1: pred = rtp->lastts + g723_samples(f->data, f->datalen); break; @@ -822,6 +831,18 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f) while((f = ast_smoother_read(rtp->smoother))) ast_rtp_raw_write(rtp, f, codec); break; + case AST_FORMAT_ILBC: + if (!rtp->smoother) { + rtp->smoother = ast_smoother_new(52); + } + if (!rtp->smoother) { + ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n"); + return -1; + } + ast_smoother_feed(rtp->smoother, _f); + while((f = ast_smoother_read(rtp->smoother))) + ast_rtp_raw_write(rtp, f, codec); + break; default: ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", _f->subclass); // fall through to... -- cgit v1.2.3