diff options
| author | Sean Bright <sean.bright@gmail.com> | 2017-12-22 09:23:22 -0500 |
|---|---|---|
| committer | Sean Bright <sean.bright@gmail.com> | 2017-12-22 09:23:22 -0500 |
| commit | fd0ca1c3f9b972a52d48a82b492fd6bac772dc78 (patch) | |
| tree | 42d2a87726d196f4db1c68489007520a4c597062 /codecs | |
| parent | 9ef97b5a9191e51f1edc66bb17728fd9fe552c35 (diff) | |
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
Diffstat (limited to 'codecs')
123 files changed, 690 insertions, 838 deletions
diff --git a/codecs/Makefile b/codecs/Makefile index 64f77ebaa..86a7dec30 100644 --- a/codecs/Makefile +++ b/codecs/Makefile @@ -1,6 +1,6 @@ # # Asterisk -- An open source telephony toolkit. -# +# # Makefile for codec modules # # Copyright (C) 1999-2006, Digium, Inc. diff --git a/codecs/codec_adpcm.c b/codecs/codec_adpcm.c index 0b2f90f39..9c202a5e8 100644 --- a/codecs/codec_adpcm.c +++ b/codecs/codec_adpcm.c @@ -23,7 +23,7 @@ /*! \file * * \brief codec_adpcm.c - translate between signed linear and Dialogic ADPCM - * + * * \ingroup codecs */ @@ -50,7 +50,7 @@ #include "ex_adpcm.h" /* - * Step size index shift table + * Step size index shift table */ static int indsft[8] = { -1, -1, -1, -1, 2, 4, 6, 8 }; @@ -167,7 +167,7 @@ static inline int adpcm(short csig, struct adpcm_state *state) int step; int encoded; - /* + /* * Clip csig if too large or too small */ csig >>= 4; @@ -208,7 +208,7 @@ static inline int adpcm(short csig, struct adpcm_state *state) /* feedback to state */ decode(encoded, state); - + return encoded; } @@ -259,7 +259,7 @@ static struct ast_frame *lintoadpcm_frameout(struct ast_trans_pvt *pvt) struct ast_frame *f; int i; int samples = pvt->samples; /* save original number */ - + if (samples < 2) return NULL; diff --git a/codecs/codec_alaw.c b/codecs/codec_alaw.c index ebaca74c5..873a59e3e 100644 --- a/codecs/codec_alaw.c +++ b/codecs/codec_alaw.c @@ -19,7 +19,7 @@ /*! \file * * \brief codec_alaw.c - translate between signed linear and alaw - * + * * \ingroup codecs */ @@ -50,7 +50,7 @@ static int alawtolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) pvt->samples += i; pvt->datalen += i * 2; /* 2 bytes/sample */ - + while (i--) *dst++ = AST_ALAW(*src++); @@ -67,7 +67,7 @@ static int lintoalaw_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) pvt->samples += i; pvt->datalen += i; /* 1 byte/sample */ - while (i--) + while (i--) *dst++ = AST_LIN2A(*src++); return 0; diff --git a/codecs/codec_g722.c b/codecs/codec_g722.c index 48b5e4a27..6046dd382 100644 --- a/codecs/codec_g722.c +++ b/codecs/codec_g722.c @@ -109,7 +109,7 @@ static int g722tolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) /* g722_decode expects the samples to be in the invalid samples / 2 format */ in_samples = f->samples / 2; - out_samples = g722_decode(&tmp->g722, &pvt->outbuf.i16[pvt->samples * sizeof(int16_t)], + out_samples = g722_decode(&tmp->g722, &pvt->outbuf.i16[pvt->samples * sizeof(int16_t)], (uint8_t *) f->data.ptr, in_samples); pvt->samples += out_samples; @@ -124,7 +124,7 @@ static int lintog722_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) struct g722_encoder_pvt *tmp = pvt->pvt; int outlen; - outlen = g722_encode(&tmp->g722, (&pvt->outbuf.ui8[pvt->datalen]), + outlen = g722_encode(&tmp->g722, (&pvt->outbuf.ui8[pvt->datalen]), (int16_t *) f->data.ptr, f->samples); pvt->samples += outlen * 2; @@ -242,7 +242,7 @@ static int load_module(void) if (res) { unload_module(); return AST_MODULE_LOAD_DECLINE; - } + } return AST_MODULE_LOAD_SUCCESS; } diff --git a/codecs/codec_g726.c b/codecs/codec_g726.c index 3b76628d5..249e4f448 100644 --- a/codecs/codec_g726.c +++ b/codecs/codec_g726.c @@ -891,7 +891,7 @@ static int load_module(void) if (res) { unload_module(); return AST_MODULE_LOAD_DECLINE; - } + } return AST_MODULE_LOAD_SUCCESS; } diff --git a/codecs/codec_gsm.c b/codecs/codec_gsm.c index 296747ee3..e5367a558 100644 --- a/codecs/codec_gsm.c +++ b/codecs/codec_gsm.c @@ -64,7 +64,7 @@ struct gsm_translator_pvt { /* both gsm2lin and lin2gsm */ static int gsm_new(struct ast_trans_pvt *pvt) { struct gsm_translator_pvt *tmp = pvt->pvt; - + return (tmp->gsm = gsm_create()) ? 0 : -1; } @@ -95,7 +95,7 @@ static int gsmtolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) src = f->data.ptr + x; } /* XXX maybe we don't need to check */ - if (pvt->samples + len > BUFFER_SAMPLES) { + if (pvt->samples + len > BUFFER_SAMPLES) { ast_log(LOG_WARNING, "Out of buffer space\n"); return -1; } diff --git a/codecs/codec_lpc10.c b/codecs/codec_lpc10.c index 00e302267..dd9d649a5 100644 --- a/codecs/codec_lpc10.c +++ b/codecs/codec_lpc10.c @@ -7,7 +7,7 @@ * * The lpc10 code is from a library used by nautilus, modified to be a bit * nicer to the compiler. - * See http://www.arl.wustl.edu/~jaf/ + * See http://www.arl.wustl.edu/~jaf/ * * See http://www.asterisk.org for more information about * the Asterisk project. Please do not directly contact @@ -137,7 +137,7 @@ static int lpc10tolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f) pvt->datalen += 2*LPC10_SAMPLES_PER_FRAME; len += LPC10_BYTES_IN_COMPRESSED_FRAME; } - if (len != f->datalen) + if (len != f->datalen) printf("Decoded %d, expected %d\n", len, f->datalen); return 0; } diff --git a/codecs/codec_resample.c b/codecs/codec_resample.c index e0d530d85..7cb4391b3 100644 --- a/codecs/codec_resample.c +++ b/codecs/codec_resample.c @@ -17,11 +17,11 @@ * at the top of the source tree. */ -/*! +/*! * \file * * \brief Resample slinear audio - * + * * \ingroup codecs */ diff --git a/codecs/codec_speex.c b/codecs/codec_speex.c index 72ac22023..0a9359620 100644 --- a/codecs/codec_speex.c +++ b/codecs/codec_speex.c @@ -21,7 +21,7 @@ * * \brief Translate between signed linear and Speex (Open Codec) * - * \note This work was motivated by Jeremy McNamara + * \note This work was motivated by Jeremy McNamara * hacked to be configurable by anthm and bkw 9/28/2004 * * \ingroup codecs @@ -43,7 +43,7 @@ /* We require a post 1.1.8 version of Speex to enable preprocessing * and better type handling - */ + */ #ifdef _SPEEX_TYPES_H #include <speex/speex_preprocess.h> #endif @@ -143,7 +143,7 @@ static int speex_encoder_construct(struct ast_trans_pvt *pvt, const SpeexMode *p if (abr) speex_encoder_ctl(tmp->speex, SPEEX_SET_ABR, &abr); if (dtx) - speex_encoder_ctl(tmp->speex, SPEEX_SET_DTX, &dtx); + speex_encoder_ctl(tmp->speex, SPEEX_SET_DTX, &dtx); tmp->silent_state = 0; tmp->fraction_lost = 0; @@ -172,7 +172,7 @@ static int lin32tospeexuwb_new(struct ast_trans_pvt *pvt) static int speex_decoder_construct(struct ast_trans_pvt *pvt, const SpeexMode *profile) { struct speex_coder_pvt *tmp = pvt->pvt; - + if (!(tmp->speex = speex_decoder_init(profile))) return -1; @@ -462,7 +462,7 @@ static struct ast_translator speextolin = { }; static struct ast_translator lintospeex = { - .name = "lintospeex", + .name = "lintospeex", .src_codec = { .name = "slin", .type = AST_MEDIA_TYPE_AUDIO, @@ -496,7 +496,7 @@ static struct ast_translator speexwbtolin16 = { .name = "slin", .type = AST_MEDIA_TYPE_AUDIO, .sample_rate = 16000, - }, + }, .format = "slin16", .newpvt = speexwbtolin16_new, .framein = speextolin_framein, @@ -577,7 +577,7 @@ static struct ast_translator lin32tospeexuwb = { .buf_size = BUFFER_SAMPLES * 2, /* XXX maybe a lot less ? */ }; -static int parse_config(int reload) +static int parse_config(int reload) { struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 }; struct ast_config *cfg = ast_config_load("codecs.conf", config_flags); @@ -594,14 +594,14 @@ static int parse_config(int reload) if (res > -1 && res < 11) { ast_verb(3, "CODEC SPEEX: Setting Quality to %d\n",res); quality = res; - } else + } else ast_log(LOG_ERROR,"Error Quality must be 0-10\n"); } else if (!strcasecmp(var->name, "complexity")) { res = abs(atoi(var->value)); if (res > -1 && res < 11) { ast_verb(3, "CODEC SPEEX: Setting Complexity to %d\n",res); complexity = res; - } else + } else ast_log(LOG_ERROR,"Error! Complexity must be 0-10\n"); } else if (!strcasecmp(var->name, "vbr_quality")) { if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0 && res_f <= 10) { @@ -625,7 +625,7 @@ static int parse_config(int reload) else ast_verb(3, "CODEC SPEEX: Disabling ABR\n"); abr = res; - } else + } else ast_log(LOG_ERROR,"Error! ABR target bitrate must be >= 0\n"); } else if (!strcasecmp(var->name, "vad")) { vad = ast_true(var->value) ? 1 : 0; @@ -675,7 +675,7 @@ static int parse_config(int reload) return 0; } -static int reload(void) +static int reload(void) { if (parse_config(1)) return AST_MODULE_LOAD_DECLINE; diff --git a/codecs/codec_ulaw.c b/codecs/codec_ulaw.c index e0a4f6841..609d06d60 100644 --- a/codecs/codec_ulaw.c +++ b/codecs/codec_ulaw.c @@ -19,7 +19,7 @@ /*! \file * * \brief codec_ulaw.c - translate between signed linear and ulaw - * + * * \ingroup codecs */ diff --git a/codecs/ex_ilbc.h b/codecs/ex_ilbc.h index 3fe2749dd..18c7983ce 100644 --- a/codecs/ex_ilbc.h +++ b/codecs/ex_ilbc.h @@ -11,10 +11,10 @@ #include "asterisk/frame.h" /* for ast_frame, etc */ static uint8_t ex_ilbc[] = { - 0xff, 0xa0, 0xff, 0xfa, 0x0f, 0x60, 0x12, 0x11, 0xa2, 0x47, - 0x22, 0x8c, 0x00, 0x00, 0x01, 0x02, 0x80, 0x43, 0xa0, 0x40, - 0x33, 0xff, 0xcf, 0xc0, 0xf3, 0xf3, 0x3f, 0x8f, 0x3f, 0xff, - 0xff, 0xff, 0xff, 0xfc, 0xf9, 0xe5, 0x55, 0x78, 0x0b, 0xca, + 0xff, 0xa0, 0xff, 0xfa, 0x0f, 0x60, 0x12, 0x11, 0xa2, 0x47, + 0x22, 0x8c, 0x00, 0x00, 0x01, 0x02, 0x80, 0x43, 0xa0, 0x40, + 0x33, 0xff, 0xcf, 0xc0, 0xf3, 0xf3, 0x3f, 0x8f, 0x3f, 0xff, + 0xff, 0xff, 0xff, 0xfc, 0xf9, 0xe5, 0x55, 0x78, 0x0b, 0xca, 0xe1, 0x27, 0x94, 0x7b, 0xa8, 0x91, 0x2c, 0x36, 0x08, 0x56, }; diff --git a/codecs/g722/g722.h b/codecs/g722/g722.h index f57b1c882..1a1e4f9de 100644 --- a/codecs/g722/g722.h +++ b/codecs/g722/g722.h @@ -7,7 +7,7 @@ * * Copyright (C) 2005 Steve Underwood * - * Despite my general liking of the GPL, I place my own contributions + * Despite my general liking of the GPL, I place my own contributions * to this code in the public domain for the benefit of all mankind - * even the slimy ones who might try to proprietize my work and use it * to my detriment. @@ -122,7 +122,7 @@ typedef struct int nb; int det; } band[2]; - + unsigned int in_buffer; int in_bits; unsigned int out_buffer; diff --git a/codecs/g722/g722_decode.c b/codecs/g722/g722_decode.c index 3e8f7d0c6..47e63eb17 100644 --- a/codecs/g722/g722_decode.c +++ b/codecs/g722/g722_decode.c @@ -7,7 +7,7 @@ * * Copyright (C) 2005 Steve Underwood * - * Despite my general liking of the GPL, I place my own contributions + * Despite my general liking of the GPL, I place my own contributions * to this code in the public domain for the benefit of all mankind - * even the slimy ones who might try to proprietize my work and use it * to my detriment. @@ -121,7 +121,7 @@ static void block4(g722_decode_state_t *s, int band, int d) s->band[band].d[i] = s->band[band].d[i - 1]; s->band[band].b[i] = s->band[band].bp[i]; } - + for (i = 2; i > 0; i--) { s->band[band].r[i] = s->band[band].r[i - 1]; @@ -198,9 +198,9 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[] static const int wh[3] = {0, -214, 798}; static const int rh2[4] = {2, 1, 2, 1}; static const int qm2[4] = {-7408, -1616, 7408, 1616}; - static const int qm4[16] = + static const int qm4[16] = { - 0, -20456, -12896, -8968, + 0, -20456, -12896, -8968, -6288, -4240, -2584, -1200, 20456, 12896, 8968, 6288, 4240, 2584, 1200, 0 @@ -320,7 +320,7 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[] else if (wd1 > 18432) wd1 = 18432; s->band[0].nb = wd1; - + /* Block 3L, SCALEL */ wd1 = (s->band[0].nb >> 6) & 31; wd2 = 8 - (s->band[0].nb >> 11); @@ -328,7 +328,7 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[] s->band[0].det = wd3 << 2; block4(s, 0, dlowt); - + if (!s->eight_k) { /* Block 2H, INVQAH */ @@ -351,7 +351,7 @@ int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[] else if (wd1 > 22528) wd1 = 22528; s->band[1].nb = wd1; - + /* Block 3H, SCALEH */ wd1 = (s->band[1].nb >> 6) & 31; wd2 = 10 - (s->band[1].nb >> 11); diff --git a/codecs/g722/g722_encode.c b/codecs/g722/g722_encode.c index 5890fbf24..cf53c74eb 100644 --- a/codecs/g722/g722_encode.c +++ b/codecs/g722/g722_encode.c @@ -9,7 +9,7 @@ * * All rights reserved. * - * Despite my general liking of the GPL, I place my own contributions + * Despite my general liking of the GPL, I place my own contributions * to this code in the public domain for the benefit of all mankind - * even the slimy ones who might try to proprietize my work and use it * to my detriment. @@ -120,7 +120,7 @@ static void block4(g722_encode_state_t *s, int band, int d) s->band[band].d[i] = s->band[band].d[i - 1]; s->band[band].b[i] = s->band[band].bp[i]; } - + for (i = 2; i > 0; i--) { s->band[band].r[i] = s->band[band].r[i - 1]; @@ -289,7 +289,7 @@ int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[] s->x[i] = s->x[i + 2]; s->x[22] = amp[j++]; s->x[23] = amp[j++]; - + /* Discard every other QMF output */ sumeven = 0; sumodd = 0; @@ -337,7 +337,7 @@ int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[] s->band[0].det = wd3 << 2; block4(s, 0, dlow); - + if (s->eight_k) { /* Just leave the high bits as zero */ diff --git a/codecs/gsm/Makefile b/codecs/gsm/Makefile index 06f082983..489a0f377 100644 --- a/codecs/gsm/Makefile +++ b/codecs/gsm/Makefile @@ -17,14 +17,14 @@ SASR = -DSASR # LTP_CUT = -DLTP_CUT LTP_CUT = -######### Define to enable the GSM library's long-term correlation +######### Define to enable the GSM library's long-term correlation ######### approximation option---faster, but worse; works for ######### both integer and floating point multiplications. ######### This flag is still in the experimental stage. WAV49 = -DWAV49 #WAV49 = -######### Define to enable the GSM library's option to pack GSM frames +######### Define to enable the GSM library's option to pack GSM frames ######### in the style used by the WAV #49 format. If you want to write ######### a tool that produces .WAV files which contain GSM-encoded data, ######### define this, and read about the GSM_OPT_WAV49 option in the @@ -35,7 +35,7 @@ WAV49 = -DWAV49 ######### Define to enable MMXTM optimizations for x86 architecture CPU's ######### which support MMX instructions. This should be newer pentiums, ######### ppro's, etc, as well as the AMD K6 and K7. The compile will -######### probably require gcc. +######### probably require gcc. # Due to the gsm codec beeing broken when compiled with gcc version 4.2 # and optimization higher than -O2 we are checking for that version and @@ -48,7 +48,7 @@ endif PG = #PG = -g -pg ######### Profiling flags. If you don't know what that means, leave it blank. - + # Choose a compiler. The code works both with ANSI and K&R-C. # Use -DNeedFunctionPrototypes to compile with, -UNeedFunctionPrototypes to # compile without, function prototypes in the header files. @@ -119,13 +119,13 @@ ARFLAGS = cr RMFLAGS = -f FIND = find COMPRESS = compress -COMPRESSFLAGS = +COMPRESSFLAGS = # RANLIB = true RANLIB = ranlib # # You shouldn't have to configure below this line if you're porting. -# +# # Local Directories @@ -484,7 +484,7 @@ $(ADDTST)/add: $(ADDTST)/add_test.o # format used by the tests we ran (.cod). We paid for the test data, # so I guess we can't just provide them with this package. Still, # if you happen to have them lying around, here's the code. -# +# # You can use gsm2cod | cod2txt independently to look at what's # coded inside the compressed frames, although this shouldn't be # hard to roll on your own using the gsm_print() function from @@ -492,7 +492,7 @@ $(ADDTST)/add: $(ADDTST)/add_test.o $(TST)/test-result: $(TST)/lin2cod $(TST)/cod2lin $(TOAST) $(TST)/run - ( cd $(TST); ./run ) + ( cd $(TST); ./run ) $(TST)/lin2txt: $(TST)/lin2txt.o $(LIBGSM) $(LD) $(LFLAGS) -o $(TST)/lin2txt \ diff --git a/codecs/gsm/README b/codecs/gsm/README index cb6af85cf..10470e04e 100644 --- a/codecs/gsm/README +++ b/codecs/gsm/README @@ -17,7 +17,7 @@ rate, i.e. a frame rate of 50 Hz) into 260 bits; for compatibility with typical UNIX applications, our implementation turns frames of 160 16-bit linear samples into 33-byte frames (1650 Bytes/s). The quality of the algorithm is good enough for reliable speaker -recognition; even music often survives transcoding in recognizable +recognition; even music often survives transcoding in recognizable form (given the bandwidth limitations of 8 kHz sampling rate). The interfaces offered are a front end modelled after compress(1), and diff --git a/codecs/gsm/inc/gsm.h b/codecs/gsm/inc/gsm.h index 81065e512..c98b06fd4 100644 --- a/codecs/gsm/inc/gsm.h +++ b/codecs/gsm/inc/gsm.h @@ -55,7 +55,7 @@ typedef gsm_byte gsm_frame[33]; /* 33 * 8 bits */ #define GSM_OPT_FRAME_CHAIN 6 extern gsm gsm_create GSM_P((void)); -extern void gsm_destroy GSM_P((gsm)); +extern void gsm_destroy GSM_P((gsm)); extern int gsm_print GSM_P((FILE *, gsm, gsm_byte *)); extern int gsm_option GSM_P((gsm, int, int *)); diff --git a/codecs/gsm/inc/private.h b/codecs/gsm/inc/private.h index 80ecbc59f..21634337c 100644 --- a/codecs/gsm/inc/private.h +++ b/codecs/gsm/inc/private.h @@ -82,10 +82,10 @@ extern longword gsm_L_asr P((longword a, int n)); extern word gsm_asr P((word a, int n)); /* - * Inlined functions from add.h + * Inlined functions from add.h */ -/* +/* * #define GSM_MULT_R(a, b) (* word a, word b, !(a == b == MIN_WORD) *) \ * (0x0FFFF & SASR(((longword)(a) * (longword)(b) + 16384), 15)) */ @@ -103,7 +103,7 @@ extern word gsm_asr P((word a, int n)); static __inline__ int GSM_L_ADD(int a, int b) { __asm__ __volatile__( - + "addl %2,%0; jno 0f; movl $0x7fffffff,%0; adcl $0,%0; 0:" : "=&r" (a) : "0" (a), "ir" (b) @@ -139,7 +139,7 @@ static __inline__ short GSM_SUB(short a, short b) #ifdef WIN32 #define inline __inline #define __inline__ __inline -#endif +#endif # define GSM_L_ADD(a, b) \ ( (a) < 0 ? ( (b) >= 0 ? (a) + (b) \ @@ -214,10 +214,10 @@ extern void Gsm_Preprocess P(( extern void Gsm_Encoding P(( struct gsm_state * S, - word * e, - word * ep, + word * e, + word * ep, word * xmaxc, - word * Mc, + word * Mc, word * xMc)); extern void Gsm_Short_Term_Analysis_Filter P(( diff --git a/codecs/gsm/inc/proto.h b/codecs/gsm/inc/proto.h index 68e1ecf05..20682c35d 100644 --- a/codecs/gsm/inc/proto.h +++ b/codecs/gsm/inc/proto.h @@ -40,7 +40,7 @@ # define P1(x, a) (a) # define P2(x, a, b) (a, b) # define P3(x, a, b, c) (a, b, c) -# define P4(x, a, b, c, d) (a, b, c, d) +# define P4(x, a, b, c, d) (a, b, c, d) # define P5(x, a, b, c, d, e) (a, b, c, d, e) # define P6(x, a, b, c, d, e, f) (a, b, c, d, e, f) # define P7(x, a, b, c, d, e, f, g) (a, b, c, d, e, f, g) diff --git a/codecs/gsm/src/add.c b/codecs/gsm/src/add.c index f23d27f16..d3e722de6 100644 --- a/codecs/gsm/src/add.c +++ b/codecs/gsm/src/add.c @@ -88,7 +88,7 @@ longword gsm_L_sub P2((a,b), longword a, longword b) } else if (b <= 0) return a - b; else { - /* a<0, b>0 */ + /* a<0, b>0 */ ulongword A = (ulongword)-(a + 1) + b; return A >= MAX_LONGWORD ? MIN_LONGWORD : -(longword)A - 1; @@ -120,7 +120,7 @@ word gsm_norm P1((a), longword a ) * variable L_var1 for positive values on the interval * * with minimum of - * minimum of 1073741824 (01000000000000000000000000000000) and + * minimum of 1073741824 (01000000000000000000000000000000) and * maximum of 2147483647 (01111111111111111111111111111111) * * @@ -141,7 +141,7 @@ word gsm_norm P1((a), longword a ) a = ~a; } - return a & 0xffff0000 + return a & 0xffff0000 ? ( a & 0xff000000 ? -1 + bitoff[ 0xFF & (a >> 24) ] : 7 + bitoff[ 0xFF & (a >> 16) ] ) @@ -194,7 +194,7 @@ word gsm_asr P2((a,n), word a, int n) # endif } -/* +/* * (From p. 46, end of section 4.2.5) * * NOTE: The following lines gives [sic] one correct implementation diff --git a/codecs/gsm/src/code.c b/codecs/gsm/src/code.c index 9f6b00f43..dd7e6190e 100644 --- a/codecs/gsm/src/code.c +++ b/codecs/gsm/src/code.c @@ -19,8 +19,8 @@ #include "gsm.h" #include "proto.h" -/* - * 4.2 FIXED POINT IMPLEMENTATION OF THE RPE-LTP CODER +/* + * 4.2 FIXED POINT IMPLEMENTATION OF THE RPE-LTP CODER */ void Gsm_Coder P8((S,s,LARc,Nc,bc,Mc,xmaxc,xMc), @@ -33,7 +33,7 @@ void Gsm_Coder P8((S,s,LARc,Nc,bc,Mc,xmaxc,xMc), * The RPE-LTD coder works on a frame by frame basis. The length of * the frame is equal to 160 samples. Some computations are done * once per frame to produce at the output of the coder the - * LARc[1..8] parameters which are the coded LAR coefficients and + * LARc[1..8] parameters which are the coded LAR coefficients and * also to realize the inverse filtering operation for the entire * frame (160 samples of signal d[0..159]). These parts produce at * the output of the coder: diff --git a/codecs/gsm/src/debug.c b/codecs/gsm/src/debug.c index 22dfa8082..c2469df1a 100644 --- a/codecs/gsm/src/debug.c +++ b/codecs/gsm/src/debug.c @@ -18,7 +18,7 @@ #include <stdio.h> #include "proto.h" -void gsm_debug_words P4( (name, from, to, ptr), +void gsm_debug_words P4( (name, from, to, ptr), char * name, int from, int to, diff --git a/codecs/gsm/src/gsm_decode.c b/codecs/gsm/src/gsm_decode.c index 7318ba2d4..7ebf35dd0 100644 --- a/codecs/gsm/src/gsm_decode.c +++ b/codecs/gsm/src/gsm_decode.c @@ -206,7 +206,7 @@ int gsm_decode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target) xmaxc[2] = sr & 0x3f; sr >>= 6; xmc[26] = sr & 0x7; sr >>= 3; xmc[27] = sr & 0x7; sr >>= 3; - sr |= (uword)*c++ << 1; + sr |= (uword)*c++ << 1; xmc[28] = sr & 0x7; sr >>= 3; xmc[29] = sr & 0x7; sr >>= 3; xmc[30] = sr & 0x7; sr >>= 3; @@ -223,7 +223,7 @@ int gsm_decode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target) xmc[38] = sr & 0x7; sr >>= 3; sr = *c++; Nc[3] = sr & 0x7f; sr >>= 7; - sr |= (uword)*c++ << 1; + sr |= (uword)*c++ << 1; bc[3] = sr & 0x3; sr >>= 2; Mc[3] = sr & 0x3; sr >>= 2; sr |= (uword)*c++ << 5; diff --git a/codecs/gsm/src/gsm_explode.c b/codecs/gsm/src/gsm_explode.c index a906fc2ed..744ded5af 100644 --- a/codecs/gsm/src/gsm_explode.c +++ b/codecs/gsm/src/gsm_explode.c @@ -228,7 +228,7 @@ int gsm_explode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target) #define xmc (target + 46 - 26) xmc[26] = sr & 0x7; sr >>= 3; xmc[27] = sr & 0x7; sr >>= 3; - sr |= (uword)*c++ << 1; + sr |= (uword)*c++ << 1; xmc[28] = sr & 0x7; sr >>= 3; xmc[29] = sr & 0x7; sr >>= 3; xmc[30] = sr & 0x7; sr >>= 3; @@ -245,7 +245,7 @@ int gsm_explode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target) xmc[38] = sr & 0x7; sr >>= 3; sr = *c++; Nc[3] = sr & 0x7f; sr >>= 7; - sr |= (uword)*c++ << 1; + sr |= (uword)*c++ << 1; bc[3] = sr & 0x3; sr >>= 2; Mc[3] = sr & 0x3; sr >>= 2; sr |= (uword)*c++ << 5; @@ -273,7 +273,7 @@ int gsm_explode P3((s, c, target), gsm s, gsm_byte * c, gsm_signal * target) xmc[51] = sr & 0x7; sr >>= 3; } } - else + else #endif { /* GSM_MAGIC = (*c >> 4) & 0xF; */ diff --git a/codecs/gsm/src/gsm_implode.c b/codecs/gsm/src/gsm_implode.c index 453b8cf39..08ebf5853 100644 --- a/codecs/gsm/src/gsm_implode.c +++ b/codecs/gsm/src/gsm_implode.c @@ -316,7 +316,7 @@ void gsm_implode P3((s, source, c), gsm s, gsm_signal * source, gsm_byte * c) #define xmc (source + 46 - 26) xmc[26] = sr & 0x7; sr >>= 3; xmc[27] = sr & 0x7; sr >>= 3; - sr |= (uword)*c++ << 1; + sr |= (uword)*c++ << 1; xmc[28] = sr & 0x7; sr >>= 3; xmc[29] = sr & 0x7; sr >>= 3; xmc[30] = sr & 0x7; sr >>= 3; @@ -333,7 +333,7 @@ void gsm_implode P3((s, source, c), gsm s, gsm_signal * source, gsm_byte * c) xmc[38] = sr & 0x7; sr >>= 3; sr = *c++; Nc[3] = sr & 0x7f; sr >>= 7; - sr |= (uword)*c++ << 1; + sr |= (uword)*c++ << 1; bc[3] = sr & 0x3; sr >>= 2; Mc[3] = sr & 0x3; sr >>= 2; sr |= (uword)*c++ << 5; @@ -361,7 +361,7 @@ void gsm_implode P3((s, source, c), gsm s, gsm_signal * source, gsm_byte * c) } } else -#endif +#endif { *c++ = ((GSM_MAGIC & 0xF) << 4) /* 1 */ diff --git a/codecs/gsm/src/gsm_option.c b/codecs/gsm/src/gsm_option.c index 280780132..303170a9c 100644 --- a/codecs/gsm/src/gsm_option.c +++ b/codecs/gsm/src/gsm_option.c @@ -56,7 +56,7 @@ int gsm_option P3((r, opt, val), gsm r, int opt, int * val) case GSM_OPT_WAV49: -#ifdef WAV49 +#ifdef WAV49 result = r->wav_fmt; if (val) r->wav_fmt = !!*val; #endif diff --git a/codecs/gsm/src/k6opt.h b/codecs/gsm/src/k6opt.h index 16ea2ac8d..5c65b378c 100644 --- a/codecs/gsm/src/k6opt.h +++ b/codecs/gsm/src/k6opt.h @@ -1,7 +1,7 @@ /* k6opt.h vector functions optimized for MMX extensions to x86 * * Copyright (C) 1999 by Stanley J. Brooks <stabro@megsinet.net> - * + * * Any use of this software is permitted provided that this notice is not * removed and that neither the authors nor the Technische Universitaet Berlin * are deemed to have made any representations as to the suitability of this @@ -9,7 +9,7 @@ * this software. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE; * not even the implied warranty of MERCHANTABILITY or FITNESS FOR * A PARTICULAR PURPOSE. - * + * * Chicago, 03.12.1999 * Stanley J. Brooks */ @@ -22,7 +22,7 @@ extern void Weighting_filter P2((e, x), extern longword k6maxcc P3((wt,dp,Nc_out), const word *wt, - const word *dp, + const word *dp, word * Nc_out /* OUT */ ) ; @@ -34,7 +34,7 @@ extern longword k6maxcc P3((wt,dp,Nc_out), */ extern longword k6maxmin P3((p,n,out), const word *p, - int n, + int n, word *out /* out[0] is max, out[1] is min */ ) ; @@ -53,7 +53,7 @@ extern longword k6iprod P3((p,q,n), */ extern void k6vsraw P3((p,n,bits), const word *p, - int n, + int n, int bits ) ; @@ -65,7 +65,7 @@ extern void k6vsraw P3((p,n,bits), */ extern void k6vsllw P3((p,n,bits), const word *p, - int n, + int n, int bits ) ; diff --git a/codecs/gsm/src/long_term.c b/codecs/gsm/src/long_term.c index 83b6fdf85..571d9c7db 100644 --- a/codecs/gsm/src/long_term.c +++ b/codecs/gsm/src/long_term.c @@ -338,7 +338,7 @@ static void Cut_Calculation_of_the_LTP_parameters P5((st, d,dp,bc_out,Nc_out), else scal = 6 - temp; assert(scal >= 0); - ltp_cut = (longword)SASR(dmax, scal) * st->ltp_cut / 100; + ltp_cut = (longword)SASR(dmax, scal) * st->ltp_cut / 100; /* Initialization of a working array wt @@ -370,7 +370,7 @@ static void Cut_Calculation_of_the_LTP_parameters P5((st, d,dp,bc_out,Nc_out), register float a = lp[-8], b = lp[-7], c = lp[-6], d = lp[-5], e = lp[-4], f = lp[-3], g = lp[-2], h = lp[-1]; - register float E; + register float E; register float S0 = 0, S1 = 0, S2 = 0, S3 = 0, S4 = 0, S5 = 0, S6 = 0, S7 = 0, S8 = 0; @@ -536,7 +536,7 @@ static void Calculation_of_the_LTP_parameters P4((d,dp,bc_out,Nc_out), register float a = lp[-8], b = lp[-7], c = lp[-6], d = lp[-5], e = lp[-4], f = lp[-3], g = lp[-2], h = lp[-1]; - register float E; + register float E; register float S0 = 0, S1 = 0, S2 = 0, S3 = 0, S4 = 0, S5 = 0, S6 = 0, S7 = 0, S8 = 0; @@ -750,7 +750,7 @@ static void Fast_Calculation_of_the_LTP_parameters P4((d,dp,bc_out,Nc_out), register float a = lp[-8], b = lp[-7], c = lp[-6], d = lp[-5], e = lp[-4], f = lp[-3], g = lp[-2], h = lp[-1]; - register float E; + register float E; register float S0 = 0, S1 = 0, S2 = 0, S3 = 0, S4 = 0, S5 = 0, S6 = 0, S7 = 0, S8 = 0; @@ -867,7 +867,7 @@ static void Long_term_analysis_filtering P6((bc,Nc,dp,d,dpp,e), case 0: STEP( 3277 ); break; case 1: STEP( 11469 ); break; case 2: STEP( 21299 ); break; - case 3: STEP( 32767 ); break; + case 3: STEP( 32767 ); break; } } @@ -888,7 +888,7 @@ void Gsm_Long_Term_Predictor P7((S,d,dp,e,dpp,Nc,bc), /* 4x for 160 samples */ assert( dpp); assert( Nc ); assert( bc ); #if defined(FAST) && defined(USE_FLOAT_MUL) - if (S->fast) + if (S->fast) #if defined (LTP_CUT) if (S->ltp_cut) Cut_Fast_Calculation_of_the_LTP_parameters(S, @@ -896,7 +896,7 @@ void Gsm_Long_Term_Predictor P7((S,d,dp,e,dpp,Nc,bc), /* 4x for 160 samples */ else #endif /* LTP_CUT */ Fast_Calculation_of_the_LTP_parameters(d, dp, bc, Nc ); - else + else #endif /* FAST & USE_FLOAT_MUL */ #ifdef LTP_CUT if (S->ltp_cut) @@ -936,7 +936,7 @@ void Gsm_Long_Term_Synthesis_Filtering P5((S,Ncr,bcr,erp,drp), */ brp = gsm_QLB[ bcr ]; - /* Computation of the reconstructed short term residual + /* Computation of the reconstructed short term residual * signal drp[0..39] */ assert(brp != MIN_WORD); diff --git a/codecs/gsm/src/lpc.c b/codecs/gsm/src/lpc.c index 744149e02..7ce2c2ff3 100644 --- a/codecs/gsm/src/lpc.c +++ b/codecs/gsm/src/lpc.c @@ -84,7 +84,7 @@ static void Autocorrelation P2((s, L_ACF), float_s[k] = (float) \ (s[k] = GSM_MULT_R(s[k], 16384 >> (n-1)));\ break; -# else +# else # define SCALE(n) \ case n: for (k = 0; k <= 159; k++) \ s[k] = (word)GSM_MULT_R( s[k], 16384 >> (n-1) );\ @@ -153,7 +153,7 @@ static void Autocorrelation P2((s, L_ACF), STEP(5); STEP(6); STEP(7); STEP(8); } - for (k = 9; k--; L_ACF[k] <<= 1) ; + for (k = 9; k--; L_ACF[k] <<= 1) ; } @@ -168,7 +168,7 @@ static void Autocorrelation P2((s, L_ACF), /* Rescaling of the array s[0..159] */ if (scalauto > 0) { - assert(scalauto <= 4); + assert(scalauto <= 4); #ifndef K6OPT for (k = 160; k--; *s++ <<= scalauto) ; # else /* K6OPT */ @@ -256,7 +256,7 @@ static void Reflection_coefficients P2( (L_ACF, r), assert(*r >= 0); if (P[1] > 0) *r = -*r; /* r[n] = sub(0, r[n]) */ assert (*r != MIN_WORD); - if (n == 8) return; + if (n == 8) return; /* Schur recursion */ @@ -325,7 +325,7 @@ static void Quantization_and_coding P1((LAR), /* This procedure needs four tables; the following equations * give the optimum scaling for the constants: - * + * * A[0..7] = integer( real_A[0..7] * 1024 ) * B[0..7] = integer( real_B[0..7] * 512 ) * MAC[0..7] = maximum of the LARc[0..7] diff --git a/codecs/gsm/src/preprocess.c b/codecs/gsm/src/preprocess.c index eacdac851..da7baae13 100644 --- a/codecs/gsm/src/preprocess.c +++ b/codecs/gsm/src/preprocess.c @@ -15,7 +15,7 @@ #include "proto.h" /* 4.2.0 .. 4.2.3 PREPROCESSING SECTION - * + * * After A-law to linear conversion (or directly from the * Ato D converter) the following scaling is assumed for * input to the RPE-LTP algorithm: @@ -26,7 +26,7 @@ * Where S is the sign bit, v a valid bit, and * a "don't care" bit. * The original signal is called sop[..] * - * out: 0.1................... 12 + * out: 0.1................... 12 * S.S.v.v.v.v.v.v.v.v.v.v.v.v.0.0 */ @@ -59,7 +59,7 @@ void Gsm_Preprocess P3((S, s, so), /* 4.2.2 Offset compensation - * + * * This part implements a high-pass filter and requires extended * arithmetic precision for the recursive part of this filter. * The input of this procedure is the array so[0...159] and the @@ -82,15 +82,15 @@ void Gsm_Preprocess P3((S, s, so), */ { word msp; -#ifndef __GNUC__ +#ifndef __GNUC__ word lsp; #endif longword L_s2; longword L_temp; - + L_s2 = s1; L_s2 <<= 15; -#ifndef __GNUC__ +#ifndef __GNUC__ msp = (word)SASR( L_z2, 15 ); lsp = (word)(L_z2 & 0x7fff); /* gsm_L_sub(L_z2,(msp<<15)); */ diff --git a/codecs/gsm/src/rpe.c b/codecs/gsm/src/rpe.c index 1c354795d..2ace69f2f 100644 --- a/codecs/gsm/src/rpe.c +++ b/codecs/gsm/src/rpe.c @@ -29,7 +29,7 @@ static void Weighting_filter P2((e, x), * The coefficients of the weighting filter are stored in a table * (see table 4.4). The following scaling is used: * - * H[0..10] = integer( real_H[ 0..10] * 8192 ); + * H[0..10] = integer( real_H[ 0..10] * 8192 ); */ { /* word wt[ 50 ]; */ @@ -50,7 +50,7 @@ static void Weighting_filter P2((e, x), e -= 5; /* Compute the signal x[0..39] - */ + */ for (k = 0; k <= 39; k++) { L_result = 8192 >> 1; @@ -65,7 +65,7 @@ static void Weighting_filter P2((e, x), #define STEP( i, H ) (e[ k + i ] * (longword)H) /* Every one of these multiplications is done twice -- - * but I don't see an elegant way to optimize this. + * but I don't see an elegant way to optimize this. * Do you? */ @@ -83,16 +83,16 @@ static void Weighting_filter P2((e, x), L_result += STEP( 10, -134 ) ; #else L_result += - STEP( 0, -134 ) - + STEP( 1, -374 ) + STEP( 0, -134 ) + + STEP( 1, -374 ) /* + STEP( 2, 0 ) */ - + STEP( 3, 2054 ) - + STEP( 4, 5741 ) - + STEP( 5, 8192 ) - + STEP( 6, 5741 ) - + STEP( 7, 2054 ) + + STEP( 3, 2054 ) + + STEP( 4, 5741 ) + + STEP( 5, 8192 ) + + STEP( 6, 5741 ) + + STEP( 7, 2054 ) /* + STEP( 8, 0 ) */ - + STEP( 9, -374 ) + + STEP( 9, -374 ) + STEP(10, -134 ) ; #endif @@ -117,7 +117,7 @@ static void Weighting_filter P2((e, x), /* 4.2.14 */ static void RPE_grid_selection P3((x,xM,Mc_out), - word * x, /* [0..39] IN */ + word * x, /* [0..39] IN */ word * xM, /* [0..12] OUT */ word * Mc_out /* OUT */ ) @@ -150,7 +150,7 @@ static void RPE_grid_selection P3((x,xM,Mc_out), * L_temp = GSM_L_MULT( temp1, temp1 ); * L_result = GSM_L_ADD( L_temp, L_result ); * } - * + * * if (L_result > EM) { * Mc = m; * EM = L_result; @@ -313,7 +313,7 @@ static void APCM_quantization P5((xM,xMc,mant_out,exp_out,xmaxc_out), * can be calculated by using the exponent and the mantissa part of * xmaxc (logarithmic table). * So, this method avoids any division and uses only a scaling - * of the RPE samples by a function of the exponent. A direct + * of the RPE samples by a function of the exponent. A direct * multiplication by the inverse of the mantissa (NRFAC[0..7] * found in table 4.5) gives the 3 bit coded version xMc[0..12] * of the RPE samples. @@ -324,7 +324,7 @@ static void APCM_quantization P5((xM,xMc,mant_out,exp_out,xmaxc_out), */ assert( exp <= 4096 && exp >= -4096); - assert( mant >= 0 && mant <= 7 ); + assert( mant >= 0 && mant <= 7 ); temp1 = 6 - exp; /* normalization by the exponent */ temp2 = gsm_NRFAC[ mant ]; /* inverse mantissa */ @@ -354,7 +354,7 @@ static void APCM_inverse_quantization P4((xMc,mant,exp,xMp), word mant, word exp, register word * xMp) /* [0..12] OUT */ -/* +/* * This part is for decoding the RPE sequence of coded xMc[0..12] * samples to obtain the xMp[0..12] array. Table 4.6 is used to get * the mantissa of xmaxc (FAC[0..7]). @@ -363,7 +363,7 @@ static void APCM_inverse_quantization P4((xMc,mant,exp,xMp), int i; word temp, temp1, temp2, temp3; - assert( mant >= 0 && mant <= 7 ); + assert( mant >= 0 && mant <= 7 ); temp1 = gsm_FAC[ mant ]; /* see 4.2-15 for mant */ temp2 = gsm_sub( 6, exp ); /* see 4.2-15 for exp */ @@ -440,7 +440,7 @@ void Gsm_Update_of_reconstructed_short_time_residual_signal P3((dpp, ep, dp), { int k; - for (k = 0; k <= 79; k++) + for (k = 0; k <= 79; k++) dp[ -120 + k ] = dp[ -80 + k ]; for (k = 0; k <= 39; k++) diff --git a/codecs/gsm/src/short_term.c b/codecs/gsm/src/short_term.c index 43c592c04..4172d322c 100644 --- a/codecs/gsm/src/short_term.c +++ b/codecs/gsm/src/short_term.c @@ -77,7 +77,7 @@ static void Decoding_of_the_coded_Log_Area_Ratios P2((LARc,LARpp), } /* 4.2.9 */ -/* Computation of the quantized reflection coefficients +/* Computation of the quantized reflection coefficients */ /* 4.2.9.1 Interpolation of the LARpp[1..8] to get the LARp[1..8] @@ -401,7 +401,7 @@ void Gsm_Short_Term_Analysis_Filter P3((S,LARc,s), Coefficients_40_159( LARpp_j, LARp); LARp_to_rp( LARp ); FILTER( S->u, LARp, 120, s + 40); - + } void Gsm_Short_Term_Synthesis_Filter P4((S, LARcr, wt, s), diff --git a/codecs/gsm/src/table.c b/codecs/gsm/src/table.c index 16a04118c..d8366931e 100644 --- a/codecs/gsm/src/table.c +++ b/codecs/gsm/src/table.c @@ -51,7 +51,7 @@ word gsm_QLB[4] = { 3277, 11469, 21299, 32767 }; word gsm_H[11] = {-134, -374, 0, 2054, 5741, 8192, 5741, 2054, 0, -374, -134 }; -/* Table 4.5 Normalized inverse mantissa used to compute xM/xmax +/* Table 4.5 Normalized inverse mantissa used to compute xM/xmax */ /* i 0 1 2 3 4 5 6 7 */ word gsm_NRFAC[8] = { 29128, 26215, 23832, 21846, 20165, 18725, 17476, 16384 }; diff --git a/codecs/ilbc/FrameClassify.c b/codecs/ilbc/FrameClassify.c index 80d72d917..f4e73ccf2 100644 --- a/codecs/ilbc/FrameClassify.c +++ b/codecs/ilbc/FrameClassify.c @@ -111,4 +111,3 @@ return max_ssqEn_n; } - diff --git a/codecs/ilbc/FrameClassify.h b/codecs/ilbc/FrameClassify.h index 018ddbe5d..6997a0d32 100644 --- a/codecs/ilbc/FrameClassify.h +++ b/codecs/ilbc/FrameClassify.h @@ -24,4 +24,3 @@ #endif - diff --git a/codecs/ilbc/LPCdecode.c b/codecs/ilbc/LPCdecode.c index 81bab9048..a0d77ff24 100644 --- a/codecs/ilbc/LPCdecode.c +++ b/codecs/ilbc/LPCdecode.c @@ -150,9 +150,3 @@ length*sizeof(float)); } - - - - - - diff --git a/codecs/ilbc/LPCdecode.h b/codecs/ilbc/LPCdecode.h index abe1d7b3b..fc23ab7ab 100644 --- a/codecs/ilbc/LPCdecode.h +++ b/codecs/ilbc/LPCdecode.h @@ -38,15 +38,3 @@ ); #endif - - - - - - - - - - - - diff --git a/codecs/ilbc/LPCencode.c b/codecs/ilbc/LPCencode.c index 09c690958..7ef6204e0 100644 --- a/codecs/ilbc/LPCencode.c +++ b/codecs/ilbc/LPCencode.c @@ -236,4 +236,3 @@ lsf, lsfdeq, iLBCenc_inst->lsfold, iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst); } - diff --git a/codecs/ilbc/LPCencode.h b/codecs/ilbc/LPCencode.h index 39a9b0403..c9cdfd471 100644 --- a/codecs/ilbc/LPCencode.h +++ b/codecs/ilbc/LPCencode.h @@ -25,4 +25,3 @@ ); #endif - diff --git a/codecs/ilbc/StateConstructW.c b/codecs/ilbc/StateConstructW.c index 1d3e65fa6..9fd45251d 100644 --- a/codecs/ilbc/StateConstructW.c +++ b/codecs/ilbc/StateConstructW.c @@ -72,15 +72,3 @@ out[k] = fout[len-1-k]+fout[2*len-1-k]; } } - - - - - - - - - - - - diff --git a/codecs/ilbc/StateConstructW.h b/codecs/ilbc/StateConstructW.h index 08f93d281..19ec6cf86 100644 --- a/codecs/ilbc/StateConstructW.h +++ b/codecs/ilbc/StateConstructW.h @@ -28,4 +28,3 @@ ); #endif - diff --git a/codecs/ilbc/StateSearchW.c b/codecs/ilbc/StateSearchW.c index 1eeee936c..7320dded9 100644 --- a/codecs/ilbc/StateSearchW.c +++ b/codecs/ilbc/StateSearchW.c @@ -198,14 +198,3 @@ AbsQuantW(iLBCenc_inst, fout,syntDenum, weightDenum,idxVec, len, state_first); } - - - - - - - - - - - diff --git a/codecs/ilbc/StateSearchW.h b/codecs/ilbc/StateSearchW.h index 9036daa89..730e03960 100644 --- a/codecs/ilbc/StateSearchW.h +++ b/codecs/ilbc/StateSearchW.h @@ -42,11 +42,3 @@ #endif - - - - - - - - diff --git a/codecs/ilbc/anaFilter.c b/codecs/ilbc/anaFilter.c index c161f98a5..3ac2127a1 100644 --- a/codecs/ilbc/anaFilter.c +++ b/codecs/ilbc/anaFilter.c @@ -69,4 +69,3 @@ memcpy(mem, &In[len-LPC_FILTERORDER], LPC_FILTERORDER*sizeof(float)); } - diff --git a/codecs/ilbc/anaFilter.h b/codecs/ilbc/anaFilter.h index 91367f95e..df741bfb8 100644 --- a/codecs/ilbc/anaFilter.h +++ b/codecs/ilbc/anaFilter.h @@ -27,4 +27,3 @@ ); #endif - diff --git a/codecs/ilbc/constants.c b/codecs/ilbc/constants.c index d43dac4d4..bedfc3971 100644 --- a/codecs/ilbc/constants.c +++ b/codecs/ilbc/constants.c @@ -767,4 +767,3 @@ (float)1.786499, (float)2.041748, (float)2.290405, (float)2.650757, (float)1.938232, (float)2.264404, (float)2.529053, (float)2.796143 }; - diff --git a/codecs/ilbc/constants.h b/codecs/ilbc/constants.h index 3487d529d..da84a9d47 100644 --- a/codecs/ilbc/constants.h +++ b/codecs/ilbc/constants.h @@ -77,4 +77,3 @@ #endif - diff --git a/codecs/ilbc/createCB.c b/codecs/ilbc/createCB.c index aefacf200..31d6eab19 100644 --- a/codecs/ilbc/createCB.c +++ b/codecs/ilbc/createCB.c @@ -228,4 +228,3 @@ } - diff --git a/codecs/ilbc/createCB.h b/codecs/ilbc/createCB.h index eda547c66..6689cc27d 100644 --- a/codecs/ilbc/createCB.h +++ b/codecs/ilbc/createCB.h @@ -55,4 +55,3 @@ ); #endif - diff --git a/codecs/ilbc/doCPLC.c b/codecs/ilbc/doCPLC.c index 9fa4584fe..c655c6df9 100644 --- a/codecs/ilbc/doCPLC.c +++ b/codecs/ilbc/doCPLC.c @@ -267,4 +267,3 @@ memcpy(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl*sizeof(float)); } - diff --git a/codecs/ilbc/doCPLC.h b/codecs/ilbc/doCPLC.h index 2bda7a144..9bbb519ee 100644 --- a/codecs/ilbc/doCPLC.h +++ b/codecs/ilbc/doCPLC.h @@ -26,4 +26,3 @@ ); #endif - diff --git a/codecs/ilbc/enhancer.c b/codecs/ilbc/enhancer.c index 1770207dd..681ec2a63 100644 --- a/codecs/ilbc/enhancer.c +++ b/codecs/ilbc/enhancer.c @@ -698,4 +698,3 @@ return (lag*2); } - diff --git a/codecs/ilbc/enhancer.h b/codecs/ilbc/enhancer.h index 8a081fb1b..80a494e09 100644 --- a/codecs/ilbc/enhancer.h +++ b/codecs/ilbc/enhancer.h @@ -33,4 +33,3 @@ ); #endif - diff --git a/codecs/ilbc/extract-cfile.awk b/codecs/ilbc/extract-cfile.awk index e4b07bc08..54fd2bbf7 100644 --- a/codecs/ilbc/extract-cfile.awk +++ b/codecs/ilbc/extract-cfile.awk @@ -5,7 +5,7 @@ BEGIN { srcname = "nothing"; } srcname = $2; printf("creating source file %s\n", srcname); }else if (srcname != "nothing") { - if (/Andersen,* *et* *al\./) + if (/Andersen,* *et* *al\./) printf("skipping %s\n", $0); else if (/ /) printf("skipping2 %s\n", $0); diff --git a/codecs/ilbc/filter.c b/codecs/ilbc/filter.c index 6565c2bee..322a9612f 100644 --- a/codecs/ilbc/filter.c +++ b/codecs/ilbc/filter.c @@ -172,4 +172,3 @@ *Out_ptr++ = o; } } - diff --git a/codecs/ilbc/filter.h b/codecs/ilbc/filter.h index 4c512cd93..9c13ec66b 100644 --- a/codecs/ilbc/filter.h +++ b/codecs/ilbc/filter.h @@ -77,4 +77,3 @@ ); #endif - diff --git a/codecs/ilbc/gainquant.c b/codecs/ilbc/gainquant.c index 0e74ff827..7b2052f6c 100644 --- a/codecs/ilbc/gainquant.c +++ b/codecs/ilbc/gainquant.c @@ -108,9 +108,3 @@ return 0.0; } - - - - - - diff --git a/codecs/ilbc/gainquant.h b/codecs/ilbc/gainquant.h index 0d024246d..543e7e324 100644 --- a/codecs/ilbc/gainquant.h +++ b/codecs/ilbc/gainquant.h @@ -27,4 +27,3 @@ ); #endif - diff --git a/codecs/ilbc/getCBvec.c b/codecs/ilbc/getCBvec.c index 4f2d29141..c0faf5f91 100644 --- a/codecs/ilbc/getCBvec.c +++ b/codecs/ilbc/getCBvec.c @@ -190,4 +190,3 @@ } } } - diff --git a/codecs/ilbc/getCBvec.h b/codecs/ilbc/getCBvec.h index 0080907c0..6ac63d743 100644 --- a/codecs/ilbc/getCBvec.h +++ b/codecs/ilbc/getCBvec.h @@ -22,4 +22,3 @@ ); #endif - diff --git a/codecs/ilbc/helpfun.c b/codecs/ilbc/helpfun.c index b79ac4f39..15189c3a1 100644 --- a/codecs/ilbc/helpfun.c +++ b/codecs/ilbc/helpfun.c @@ -321,4 +321,3 @@ return change; } - diff --git a/codecs/ilbc/helpfun.h b/codecs/ilbc/helpfun.h index 931ca68a6..21ff9c023 100644 --- a/codecs/ilbc/helpfun.h +++ b/codecs/ilbc/helpfun.h @@ -103,4 +103,3 @@ ); #endif - diff --git a/codecs/ilbc/hpInput.c b/codecs/ilbc/hpInput.c index 7ceee0964..93986d653 100644 --- a/codecs/ilbc/hpInput.c +++ b/codecs/ilbc/hpInput.c @@ -62,4 +62,3 @@ } } - diff --git a/codecs/ilbc/hpInput.h b/codecs/ilbc/hpInput.h index 3b020d11d..70fa35392 100644 --- a/codecs/ilbc/hpInput.h +++ b/codecs/ilbc/hpInput.h @@ -21,4 +21,3 @@ ); #endif - diff --git a/codecs/ilbc/hpOutput.c b/codecs/ilbc/hpOutput.c index 756160a65..4d2128ee0 100644 --- a/codecs/ilbc/hpOutput.c +++ b/codecs/ilbc/hpOutput.c @@ -58,4 +58,3 @@ po++; } } - diff --git a/codecs/ilbc/hpOutput.h b/codecs/ilbc/hpOutput.h index b213a1934..b192e3696 100644 --- a/codecs/ilbc/hpOutput.h +++ b/codecs/ilbc/hpOutput.h @@ -21,4 +21,3 @@ ); #endif - diff --git a/codecs/ilbc/iCBConstruct.c b/codecs/ilbc/iCBConstruct.c index ee9a73ca5..4d4330ff7 100644 --- a/codecs/ilbc/iCBConstruct.c +++ b/codecs/ilbc/iCBConstruct.c @@ -109,4 +109,3 @@ } } } - diff --git a/codecs/ilbc/iCBConstruct.h b/codecs/ilbc/iCBConstruct.h index 143501ed6..8ff82cb61 100644 --- a/codecs/ilbc/iCBConstruct.h +++ b/codecs/ilbc/iCBConstruct.h @@ -37,4 +37,3 @@ ); #endif - diff --git a/codecs/ilbc/iCBSearch.c b/codecs/ilbc/iCBSearch.c index 78d67df17..85c9ceca2 100644 --- a/codecs/ilbc/iCBSearch.c +++ b/codecs/ilbc/iCBSearch.c @@ -501,12 +501,3 @@ } gain_index[0]=j; } - - - - - - - - - diff --git a/codecs/ilbc/iCBSearch.h b/codecs/ilbc/iCBSearch.h index a61db1540..b8bdf68b0 100644 --- a/codecs/ilbc/iCBSearch.h +++ b/codecs/ilbc/iCBSearch.h @@ -34,4 +34,3 @@ ); #endif - diff --git a/codecs/ilbc/iLBC_decode.c b/codecs/ilbc/iLBC_decode.c index e7bda1164..b607f41d0 100644 --- a/codecs/ilbc/iLBC_decode.c +++ b/codecs/ilbc/iLBC_decode.c @@ -648,4 +648,3 @@ iLBCdec_inst->prev_enh_pl=1; } } - diff --git a/codecs/ilbc/iLBC_decode.h b/codecs/ilbc/iLBC_decode.h index 5f4384dcd..7af2fd76f 100644 --- a/codecs/ilbc/iLBC_decode.h +++ b/codecs/ilbc/iLBC_decode.h @@ -39,4 +39,3 @@ ); #endif - diff --git a/codecs/ilbc/iLBC_define.h b/codecs/ilbc/iLBC_define.h index 480c834b9..09b120e98 100644 --- a/codecs/ilbc/iLBC_define.h +++ b/codecs/ilbc/iLBC_define.h @@ -214,4 +214,3 @@ } iLBC_Dec_Inst_t; #endif - diff --git a/codecs/ilbc/iLBC_encode.c b/codecs/ilbc/iLBC_encode.c index 4c2e6f714..ff3205d0b 100644 --- a/codecs/ilbc/iLBC_encode.c +++ b/codecs/ilbc/iLBC_encode.c @@ -540,4 +540,3 @@ will treat it as a lost frame) */ dopack( &pbytes, 0, 1, &pos); } - diff --git a/codecs/ilbc/iLBC_encode.h b/codecs/ilbc/iLBC_encode.h index a3ab55f9d..2c2cd3fea 100644 --- a/codecs/ilbc/iLBC_encode.h +++ b/codecs/ilbc/iLBC_encode.h @@ -31,9 +31,3 @@ ); #endif - - - - - - diff --git a/codecs/ilbc/iLBC_test.c b/codecs/ilbc/iLBC_test.c index 92d6c0ddc..6ca7afa7e 100644 --- a/codecs/ilbc/iLBC_test.c +++ b/codecs/ilbc/iLBC_test.c @@ -311,4 +311,3 @@ } return(0); } - diff --git a/codecs/ilbc/lsf.c b/codecs/ilbc/lsf.c index b4fe0eda3..861e4f93b 100644 --- a/codecs/ilbc/lsf.c +++ b/codecs/ilbc/lsf.c @@ -274,10 +274,3 @@ a_coef[0] = 1.0; } - - - - - - - diff --git a/codecs/ilbc/lsf.h b/codecs/ilbc/lsf.h index caff77ec0..27d8dfb4c 100644 --- a/codecs/ilbc/lsf.h +++ b/codecs/ilbc/lsf.h @@ -24,4 +24,3 @@ ); #endif - diff --git a/codecs/ilbc/packing.c b/codecs/ilbc/packing.c index b7496a48e..10f1c5068 100644 --- a/codecs/ilbc/packing.c +++ b/codecs/ilbc/packing.c @@ -179,4 +179,3 @@ } } } - diff --git a/codecs/ilbc/packing.h b/codecs/ilbc/packing.h index cbb9f82df..96d390e19 100644 --- a/codecs/ilbc/packing.h +++ b/codecs/ilbc/packing.h @@ -65,4 +65,3 @@ ); #endif - diff --git a/codecs/ilbc/rfc3951.txt b/codecs/ilbc/rfc3951.txt index d4fba08e4..320651b7b 100644 --- a/codecs/ilbc/rfc3951.txt +++ b/codecs/ilbc/rfc3951.txt @@ -56,7 +56,7 @@ Abstract Andersen, et al. Experimental [Page 1] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -112,7 +112,7 @@ Table of Contents Andersen, et al. Experimental [Page 2] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -168,7 +168,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 3] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -224,7 +224,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 4] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -280,7 +280,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 5] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -336,7 +336,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 6] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -392,7 +392,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 7] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -448,7 +448,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 8] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -504,7 +504,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 9] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -560,7 +560,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 10] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -616,7 +616,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 11] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -672,7 +672,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 12] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -728,7 +728,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 13] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -784,7 +784,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 14] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -840,7 +840,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 15] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -896,7 +896,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 16] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -952,7 +952,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 17] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1008,7 +1008,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 18] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1064,7 +1064,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 19] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1120,7 +1120,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 20] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1176,7 +1176,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 21] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1232,7 +1232,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 22] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1288,7 +1288,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 23] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1344,7 +1344,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 24] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1400,7 +1400,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 25] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1456,7 +1456,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 26] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1512,7 +1512,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 27] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1568,7 +1568,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 28] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1624,7 +1624,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 29] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1680,7 +1680,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 30] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1736,7 +1736,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 31] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1792,7 +1792,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 32] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1848,7 +1848,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 33] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1904,7 +1904,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 34] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -1960,7 +1960,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 35] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2016,7 +2016,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 36] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2072,7 +2072,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 37] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2128,7 +2128,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 38] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2184,7 +2184,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 39] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2240,7 +2240,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 40] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2296,7 +2296,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 41] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2352,7 +2352,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 42] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2408,7 +2408,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 43] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2464,7 +2464,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 44] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2520,7 +2520,7 @@ APPENDIX A. Reference Implementation Andersen, et al. Experimental [Page 45] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2576,7 +2576,7 @@ A.1. iLBC_test.c Andersen, et al. Experimental [Page 46] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2632,7 +2632,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 47] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2688,7 +2688,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 48] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2744,7 +2744,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 49] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2800,7 +2800,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 50] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2856,7 +2856,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 51] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2912,7 +2912,7 @@ A.2. iLBC_encode.h Andersen, et al. Experimental [Page 52] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -2968,7 +2968,7 @@ A.3. iLBC_encode.c Andersen, et al. Experimental [Page 53] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3024,7 +3024,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 54] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3080,7 +3080,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 55] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3136,7 +3136,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 56] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3192,7 +3192,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 57] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3248,7 +3248,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 58] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3304,7 +3304,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 59] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3360,7 +3360,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 60] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3416,7 +3416,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 61] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3472,7 +3472,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 62] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3528,7 +3528,7 @@ A.4. iLBC_decode.h Andersen, et al. Experimental [Page 63] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3584,7 +3584,7 @@ A.5. iLBC_decode.c Andersen, et al. Experimental [Page 64] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3640,7 +3640,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 65] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3696,7 +3696,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 66] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3752,7 +3752,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 67] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3808,7 +3808,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 68] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3864,7 +3864,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 69] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3920,7 +3920,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 70] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -3976,7 +3976,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 71] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4032,7 +4032,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 72] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4088,7 +4088,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 73] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4144,7 +4144,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 74] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4200,7 +4200,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 75] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4256,7 +4256,7 @@ A.6. iLBC_define.h Andersen, et al. Experimental [Page 76] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4312,7 +4312,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 77] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4368,7 +4368,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 78] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4424,7 +4424,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 79] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4480,7 +4480,7 @@ A.7. constants.h Andersen, et al. Experimental [Page 80] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4536,7 +4536,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 81] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4592,7 +4592,7 @@ A.8. constants.c Andersen, et al. Experimental [Page 82] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4648,7 +4648,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 83] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4704,7 +4704,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 84] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4760,7 +4760,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 85] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4816,7 +4816,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 86] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4872,7 +4872,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 87] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4928,7 +4928,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 88] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -4984,7 +4984,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 89] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5040,7 +5040,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 90] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5096,7 +5096,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 91] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5152,7 +5152,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 92] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5208,7 +5208,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 93] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5264,7 +5264,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 94] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5320,7 +5320,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 95] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5376,7 +5376,7 @@ A.9. anaFilter.h Andersen, et al. Experimental [Page 96] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5432,7 +5432,7 @@ A.10. anaFilter.c Andersen, et al. Experimental [Page 97] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5488,7 +5488,7 @@ A.11. createCB.h Andersen, et al. Experimental [Page 98] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5544,7 +5544,7 @@ A.12. createCB.c Andersen, et al. Experimental [Page 99] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5600,7 +5600,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 100] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5656,7 +5656,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 101] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5712,7 +5712,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 102] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5768,7 +5768,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 103] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5824,7 +5824,7 @@ A.14. doCPLC.c Andersen, et al. Experimental [Page 104] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5880,7 +5880,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 105] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5936,7 +5936,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 106] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -5992,7 +5992,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 107] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6048,7 +6048,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 108] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6104,7 +6104,7 @@ A.15. enhancer.h Andersen, et al. Experimental [Page 109] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6160,7 +6160,7 @@ A.16. enhancer.c Andersen, et al. Experimental [Page 110] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6216,7 +6216,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 111] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6272,7 +6272,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 112] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6328,7 +6328,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 113] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6384,7 +6384,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 114] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6440,7 +6440,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 115] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6496,7 +6496,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 116] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6552,7 +6552,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 117] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6608,7 +6608,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 118] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6664,7 +6664,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 119] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6720,7 +6720,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 120] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6776,7 +6776,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 121] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6832,7 +6832,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 122] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6888,7 +6888,7 @@ A.17. filter.h Andersen, et al. Experimental [Page 123] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -6944,7 +6944,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 124] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7000,7 +7000,7 @@ A.18. filter.c Andersen, et al. Experimental [Page 125] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7056,7 +7056,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 126] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7112,7 +7112,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 127] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7168,7 +7168,7 @@ A.19. FrameClassify.h Andersen, et al. Experimental [Page 128] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7224,7 +7224,7 @@ A.20. FrameClassify.c Andersen, et al. Experimental [Page 129] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7280,7 +7280,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 130] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7336,7 +7336,7 @@ A.22. gainquant.c Andersen, et al. Experimental [Page 131] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7392,7 +7392,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 132] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7448,7 +7448,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 133] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7504,7 +7504,7 @@ A.24. getCBvec.c Andersen, et al. Experimental [Page 134] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7560,7 +7560,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 135] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7616,7 +7616,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 136] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7672,7 +7672,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 137] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7728,7 +7728,7 @@ A.25. helpfun.h Andersen, et al. Experimental [Page 138] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7784,7 +7784,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 139] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7840,7 +7840,7 @@ A.26. helpfun.c Andersen, et al. Experimental [Page 140] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7896,7 +7896,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 141] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -7952,7 +7952,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 142] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8008,7 +8008,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 143] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8064,7 +8064,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 144] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8120,7 +8120,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 145] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8176,7 +8176,7 @@ A.28. hpInput.c Andersen, et al. Experimental [Page 146] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8232,7 +8232,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 147] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8288,7 +8288,7 @@ A.30. hpOutput.c Andersen, et al. Experimental [Page 148] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8344,7 +8344,7 @@ A.31. iCBConstruct.h Andersen, et al. Experimental [Page 149] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8400,7 +8400,7 @@ A.32. iCBConstruct.c Andersen, et al. Experimental [Page 150] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8456,7 +8456,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 151] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8512,7 +8512,7 @@ A.33. iCBSearch.h Andersen, et al. Experimental [Page 152] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8568,7 +8568,7 @@ A.34. iCBSearch.c Andersen, et al. Experimental [Page 153] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8624,7 +8624,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 154] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8680,7 +8680,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 155] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8736,7 +8736,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 156] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8792,7 +8792,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 157] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8848,7 +8848,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 158] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8904,7 +8904,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 159] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -8960,7 +8960,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 160] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9016,7 +9016,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 161] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9072,7 +9072,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 162] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9128,7 +9128,7 @@ A.35. LPCdecode.h Andersen, et al. Experimental [Page 163] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9184,7 +9184,7 @@ A.36. LPCdecode.c Andersen, et al. Experimental [Page 164] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9240,7 +9240,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 165] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9296,7 +9296,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 166] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9352,7 +9352,7 @@ A.38. LPCencode.c Andersen, et al. Experimental [Page 167] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9408,7 +9408,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 168] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9464,7 +9464,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 169] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9520,7 +9520,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 170] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9576,7 +9576,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 171] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9632,7 +9632,7 @@ A.40. lsf.c Andersen, et al. Experimental [Page 172] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9688,7 +9688,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 173] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9744,7 +9744,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 174] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9800,7 +9800,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 175] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9856,7 +9856,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 176] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9912,7 +9912,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 177] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -9968,7 +9968,7 @@ A.41. packing.h Andersen, et al. Experimental [Page 178] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10024,7 +10024,7 @@ A.42. packing.c Andersen, et al. Experimental [Page 179] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10080,7 +10080,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 180] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10136,7 +10136,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 181] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10192,7 +10192,7 @@ A.43. StateConstructW.h Andersen, et al. Experimental [Page 182] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10248,7 +10248,7 @@ A.44. StateConstructW.c Andersen, et al. Experimental [Page 183] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10304,7 +10304,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 184] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10360,7 +10360,7 @@ A.45. StateSearchW.h Andersen, et al. Experimental [Page 185] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10416,7 +10416,7 @@ A.46. StateSearchW.c Andersen, et al. Experimental [Page 186] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10472,7 +10472,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 187] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10528,7 +10528,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 188] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10584,7 +10584,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 189] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10640,7 +10640,7 @@ A.48. syntFilter.c Andersen, et al. Experimental [Page 190] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10696,7 +10696,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 191] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10752,7 +10752,7 @@ Authors' Addresses Andersen, et al. Experimental [Page 192] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10808,7 +10808,7 @@ RFC 3951 Internet Low Bit Rate Codec December 2004 Andersen, et al. Experimental [Page 193] - + RFC 3951 Internet Low Bit Rate Codec December 2004 @@ -10864,4 +10864,3 @@ Acknowledgement Andersen, et al. Experimental [Page 194] - diff --git a/codecs/ilbc/syntFilter.c b/codecs/ilbc/syntFilter.c index 190eb009d..db5bb6673 100644 --- a/codecs/ilbc/syntFilter.c +++ b/codecs/ilbc/syntFilter.c @@ -64,17 +64,3 @@ memcpy(mem, &Out[len-LPC_FILTERORDER], LPC_FILTERORDER*sizeof(float)); } - - - - - - - - - - - - - - diff --git a/codecs/ilbc/syntFilter.h b/codecs/ilbc/syntFilter.h index 8865b8917..586ab2bd0 100644 --- a/codecs/ilbc/syntFilter.h +++ b/codecs/ilbc/syntFilter.h @@ -21,4 +21,3 @@ ); #endif - diff --git a/codecs/log2comp.h b/codecs/log2comp.h index 6dc59c307..aa43103cb 100644 --- a/codecs/log2comp.h +++ b/codecs/log2comp.h @@ -1,6 +1,6 @@ -/*! \file +/*! \file * \brief log2comp.h - various base 2 log computation versions - * + * * Asterisk -- An open source telephony toolkit. * * \author Alex Volkov <codepro@usa.net> @@ -53,8 +53,8 @@ static inline int ilog2(int val) static inline int ilog2(int val) { int a; - __asm__ ("cntlzw %0,%1" - : "=r" (a) + __asm__ ("cntlzw %0,%1" + : "=r" (a) : "r" (val) ); return 31-a; diff --git a/codecs/lpc10/Makefile b/codecs/lpc10/Makefile index b414fd77d..04b106b66 100644 --- a/codecs/lpc10/Makefile +++ b/codecs/lpc10/Makefile @@ -1,25 +1,25 @@ -# +# # Makefile for LPC-10 speech coder library (unix) -# +# # default C compiler CC?= gcc -# +# # These definitions for CFLAGS and LIB_TARGET_DIR are used when one # runs make in the lpc10 directory, without environment variables that # override them. When make is run in this directory from a makefile # for an application that uses the LPC10 coder, there are environment # variables set for CFLAGS and LIB_TARGET_DIR that override these # definitions. -# +# LIB_TARGET_DIR = . -# +# # -I$(LIB_TARGET_DIR) option needed so that #include "machine.h" # directives can find the machine.h file. -# +# CFLAGS+= -fPIC -Wno-comment diff --git a/codecs/lpc10/analys.c b/codecs/lpc10/analys.c index 50e95703d..5ea0a2d39 100644 --- a/codecs/lpc10/analys.c +++ b/codecs/lpc10/analys.c @@ -203,7 +203,7 @@ static integer c__1 = 1; /* This entry has no local state. It accesses a "constant" array */ /* declared in ANALYS. */ -/* Subroutine */ int analys_(real *speech, integer *voice, integer +/* Subroutine */ int analys_(real *speech, integer *voice, integer *pitch, real *rms, real *rc, struct lpc10_encoder_state *st) { /* Initialized data */ @@ -243,8 +243,8 @@ static integer c__1 = 1; extern int dcbias_(integer *, real *, real *); integer ipitch; integer *obound; - extern /* Subroutine */ int preemp_(real *, real *, integer *, real *, - real *), voicin_(integer *, real *, real *, integer *, integer *, + extern /* Subroutine */ int preemp_(real *, real *, integer *, real *, + real *), voicin_(integer *, real *, real *, integer *, integer *, real *, real *, integer *, real *, integer *, integer *, integer *, struct lpc10_encoder_state *); integer *voibuf; @@ -252,10 +252,10 @@ static integer c__1 = 1; real *rmsbuf; extern /* Subroutine */ int lpfilt_(real *, real *, integer *, integer *), ivfilt_(real *, real *, integer *, integer *, real *), energy_( - integer *, real *, real *), invert_(integer *, real *, real *, + integer *, real *, real *), invert_(integer *, real *, real *, real *); integer minptr, maxptr; - extern /* Subroutine */ int dyptrk_(real *, integer *, integer *, integer + extern /* Subroutine */ int dyptrk_(real *, integer *, integer *, integer *, integer *, integer *, struct lpc10_encoder_state *); real phi[100] /* was [10][10] */, psi[10]; @@ -355,13 +355,13 @@ static integer c__1 = 1; /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -371,35 +371,35 @@ static integer c__1 = 1; /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -426,7 +426,7 @@ static integer c__1 = 1; /* The tables TAU and BUFLIM, and the variable PRECOEF, are not */ /* Fortran PARAMETER's, but they are initialized with DATA */ /* statements, and never modified. Thus, they need not have SAVE */ -/* statements for them to keep their values from one invocation to +/* statements for them to keep their values from one invocation to */ /* the next. */ /* Local variables that need not be saved */ @@ -512,13 +512,13 @@ static integer c__1 = 1; vwin[(i__ << 1) - 1] = vwin[((i__ + 1) << 1) - 1] - contrl_1.lframe; awin[(i__ << 1) - 2] = awin[((i__ + 1) << 1) - 2] - contrl_1.lframe; awin[(i__ << 1) - 1] = awin[((i__ + 1) << 1) - 1] - contrl_1.lframe; -/* EWIN(*,J) is unused for J .NE. AF, so the following shift is +/* EWIN(*,J) is unused for J .NE. AF, so the following shift is */ /* unnecessary. It also causes error messages when the C versio n */ -/* of the code created from this by f2c is run with Purify. It +/* of the code created from this by f2c is run with Purify. It */ -/* correctly complains that uninitialized memory is being read. +/* correctly complains that uninitialized memory is being read. */ /* EWIN(1,I) = EWIN(1,I+1) - LFRAME */ /* EWIN(2,I) = EWIN(2,I+1) - LFRAME */ @@ -533,17 +533,17 @@ n */ } /* Copy input speech, scale to sign+12 bit integers */ /* Remove long term DC bias. */ -/* If the average value in the frame was over 1/4096 (after current +/* If the average value in the frame was over 1/4096 (after current */ /* BIAS correction), then subtract that much more from samples in */ /* next frame. If the average value in the frame was under */ -/* -1/4096, add 1/4096 more to samples in next frame. In all other +/* -1/4096, add 1/4096 more to samples in next frame. In all other */ /* cases, keep BIAS the same. */ temp = 0.f; i__1 = contrl_1.lframe; for (i__ = 1; i__ <= i__1; ++i__) { - inbuf[720 - contrl_1.lframe + i__ - 181] = speech[i__] * 4096.f - + inbuf[720 - contrl_1.lframe + i__ - 181] = speech[i__] * 4096.f - (*bias); temp += inbuf[720 - contrl_1.lframe + i__ - 181]; } @@ -555,7 +555,7 @@ n */ } /* Place Voicing Window */ i__ = 721 - contrl_1.lframe; - preemp_(&inbuf[i__ - 181], &pebuf[i__ - 181], &contrl_1.lframe, &precoef, + preemp_(&inbuf[i__ - 181], &pebuf[i__ - 181], &contrl_1.lframe, &precoef, zpre); onset_(pebuf, osbuf, osptr, &c__10, &c__181, &c__720, &contrl_1.lframe, st); @@ -563,39 +563,39 @@ n */ /* MAXOSP = MAX( MAXOSP, OSPTR ) */ - placev_(osbuf, osptr, &c__10, &obound[2], vwin, &c__3, &contrl_1.lframe, + placev_(osbuf, osptr, &c__10, &obound[2], vwin, &c__3, &contrl_1.lframe, &c__90, &c__156, &c__307, &c__462); -/* The Pitch Extraction algorithm estimates the pitch for a frame +/* The Pitch Extraction algorithm estimates the pitch for a frame */ /* of speech by locating the minimum of the average magnitude difference */ /* function (AMDF). The AMDF operates on low-pass, inverse filtered */ -/* speech. (The low-pass filter is an 800 Hz, 19 tap, equiripple, FIR +/* speech. (The low-pass filter is an 800 Hz, 19 tap, equiripple, FIR */ -/* filter and the inverse filter is a 2nd-order LPC filter.) The pitch +/* filter and the inverse filter is a 2nd-order LPC filter.) The pitch */ -/* estimate is later refined by dynamic programming (DYPTRK). However, +/* estimate is later refined by dynamic programming (DYPTRK). However, */ /* since some of DYPTRK's parameters are a function of the voicing */ /* decisions, a voicing decision must precede the final pitch estimation. */ /* See subroutines LPFILT, IVFILT, and TBDM. */ /* LPFILT reads indices LBUFH-LFRAME-29 = 511 through LBUFH = 720 */ -/* of INBUF, and writes indices LBUFH+1-LFRAME = 541 through LBUFH +/* of INBUF, and writes indices LBUFH+1-LFRAME = 541 through LBUFH */ /* = 720 of LPBUF. */ lpfilt_(&inbuf[228], &lpbuf[384], &c__312, &contrl_1.lframe); -/* IVFILT reads indices (PWINH-LFRAME-7) = 353 through PWINH = 540 +/* IVFILT reads indices (PWINH-LFRAME-7) = 353 through PWINH = 540 */ /* of LPBUF, and writes indices (PWINH-LFRAME+1) = 361 through */ /* PWINH = 540 of IVBUF. */ ivfilt_(&lpbuf[204], ivbuf, &c__312, &contrl_1.lframe, ivrc); /* TBDM reads indices PWINL = 229 through */ -/* (PWINL-1)+MAXWIN+(TAU(LTAU)-TAU(1))/2 = 452 of IVBUF, and writes +/* (PWINL-1)+MAXWIN+(TAU(LTAU)-TAU(1))/2 = 452 of IVBUF, and writes */ /* indices 1 through LTAU = 60 of AMDF. */ tbdm_(ivbuf, &c__156, tau, &c__60, amdf, &minptr, &maxptr, &mintau); -/* Voicing decisions are made for each half frame of input speech. +/* Voicing decisions are made for each half frame of input speech. */ /* An initial voicing classification is made for each half of the */ /* analysis frame, and the voicing decisions for the present frame */ @@ -605,9 +605,9 @@ n */ /* maximum-to-minimum ratio, the zero crossing rate, energy measures, */ /* reflection coefficients, and prediction gains. */ /* The pitch and voicing rules apply smoothing and isolated */ -/* corrections to the pitch and voicing estimates and, in the process, +/* corrections to the pitch and voicing estimates and, in the process, */ -/* introduce two frames of delay into the corrected pitch estimates and +/* introduce two frames of delay into the corrected pitch estimates and */ /* voicing decisions. */ for (half = 1; half <= 2; ++half) { diff --git a/codecs/lpc10/bsynz.c b/codecs/lpc10/bsynz.c index daf9105d6..de43b43d6 100644 --- a/codecs/lpc10/bsynz.c +++ b/codecs/lpc10/bsynz.c @@ -119,7 +119,7 @@ extern struct { /* reinitialize its state for any other reason, call the ENTRY */ /* INITBSYNZ. */ -/* Subroutine */ int bsynz_(real *coef, integer *ip, integer *iv, +/* Subroutine */ int bsynz_(real *coef, integer *ip, integer *iv, real *sout, real *rms, real *ratio, real *g2pass, struct lpc10_decoder_state *st) { @@ -245,13 +245,13 @@ extern struct { /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -261,35 +261,35 @@ extern struct { /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -301,11 +301,11 @@ extern struct { /* common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */ /* Function return value definitions */ /* Parameters/constants */ -/* KEXC is not a Fortran PARAMETER, but it is an array initialized +/* KEXC is not a Fortran PARAMETER, but it is an array initialized */ /* with a DATA statement that is never modified. */ /* Local variables that need not be saved */ -/* NOISE is declared with range (1:MAXPIT+MAXORD), but only indices +/* NOISE is declared with range (1:MAXPIT+MAXORD), but only indices */ /* ORDER+1 through ORDER+IP are ever used, and I think that IP */ /* .LE. MAXPIT. Why not declare it to be in the range (1:MAXPIT) */ @@ -357,7 +357,7 @@ extern struct { /* Impulse doublet excitation for plosives */ /* (RANDOM()+32768) is in the range 0 to 2**16-1. Therefore the */ -/* following expression should be evaluated using integers with +/* following expression should be evaluated using integers with at */ /* least 32 bits (16 isn't enough), and PX should be in the rang e */ diff --git a/codecs/lpc10/chanwr.c b/codecs/lpc10/chanwr.c index b7a9c722e..280cd6893 100644 --- a/codecs/lpc10/chanwr.c +++ b/codecs/lpc10/chanwr.c @@ -129,7 +129,7 @@ Some OSS fixes and a few lpc changes to make it actually work /* R5-1, R6-1, R7-2, R9-0, P-5, */ /* R5-2, R6-2,R10-1, R8-2, P-6, R9-1, */ /* R5-3, R6-3, R7-3, R9-2, R8-3, SYNC */ -/* Subroutine */ int chanwr_0_(int n__, integer *order, integer *ipitv, +/* Subroutine */ int chanwr_0_(int n__, integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits, struct lpc10_encoder_state *st) { @@ -150,7 +150,7 @@ Some OSS fixes and a few lpc changes to make it actually work /* Arguments */ /* Parameters/constants */ /* These arrays are not Fortran PARAMETER's, but they are defined */ -/* by DATA statements below, and their contents are never altered. +/* by DATA statements below, and their contents are never altered. */ /* Local variables that need not be saved */ /* Local state */ @@ -220,13 +220,13 @@ L_chanrd: return 0; } /* chanwr_ */ -/* Subroutine */ int chanwr_(integer *order, integer *ipitv, integer *irms, +/* Subroutine */ int chanwr_(integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits, struct lpc10_encoder_state *st) { return chanwr_0_(0, order, ipitv, irms, irc, ibits, st); } -/* Subroutine */ int chanrd_(integer *order, integer *ipitv, integer *irms, +/* Subroutine */ int chanrd_(integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits) { return chanwr_0_(1, order, ipitv, irms, irc, ibits, NULL); diff --git a/codecs/lpc10/dcbias.c b/codecs/lpc10/dcbias.c index d5a7d644f..2fb07cf2b 100644 --- a/codecs/lpc10/dcbias.c +++ b/codecs/lpc10/dcbias.c @@ -104,4 +104,3 @@ extern int dcbias_(integer *len, real *speech, real *sigout); } return 0; } /* dcbias_ */ - diff --git a/codecs/lpc10/decode.c b/codecs/lpc10/decode.c index 08b8b9192..d58cd0874 100644 --- a/codecs/lpc10/decode.c +++ b/codecs/lpc10/decode.c @@ -67,7 +67,7 @@ static integer c__2 = 2; * * Revision 1.15 2003/09/19 01:20:22 markster * Code cleanups (bug #66) - * + * * Revision 1.2 2003/09/19 01:20:22 markster * Code cleanups (bug #66) * @@ -144,7 +144,7 @@ static integer c__2 = 2; /* reinitialize its state for any other reason, call the ENTRY */ /* INITDECODE. */ -/* Subroutine */ int decode_(integer *ipitv, integer *irms, +/* Subroutine */ int decode_(integer *ipitv, integer *irms, integer *irc, integer *voice, integer *pitch, real *rms, real *rc, struct lpc10_decoder_state *st) { @@ -301,13 +301,13 @@ static integer c__2 = 2; /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -317,35 +317,35 @@ static integer c__2 = 2; /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -379,9 +379,9 @@ static integer c__2 = 2; /* The following are used regardless of CORRP's value */ /* The following are used only if CORRP is .TRUE. */ /* I am guessing the initial values for IVP2H, IOVOIC, DRC, DPIT, */ -/* and DRMS. They should be checked to see if they are reasonable. +/* and DRMS. They should be checked to see if they are reasonable. */ -/* I'm also guessing for ERATE, but I think 0 is the right initial +/* I'm also guessing for ERATE, but I think 0 is the right initial */ /* value. */ /* Parameter adjustments */ @@ -474,7 +474,7 @@ static integer c__2 = 2; if (*first) { *first = FALSE_; /* Assign PITCH a "default" value on the first call, since */ -/* otherwise it would be left uninitialized. The two lines +/* otherwise it would be left uninitialized. The two lines */ /* below were copied from above, since it seemed like a */ /* reasonable thing to do for the first call. */ @@ -535,7 +535,7 @@ static integer c__2 = 2; *pitch = dpit[1]; /* If bit 2 of ICORF is set then smooth RMS and RC's, */ if ((icorf & bit[1]) != 0) { - if ((i__1 = drms[1] - drms[0], (real) abs(i__1)) >= corth[ixcor + 3] + if ((i__1 = drms[1] - drms[0], (real) abs(i__1)) >= corth[ixcor + 3] && (i__2 = drms[1] - drms[2], (real) abs(i__2)) >= corth[ ixcor + 3]) { *irms = median_(&drms[2], &drms[1], drms); @@ -552,7 +552,7 @@ static integer c__2 = 2; } /* If bit 3 of ICORF is set then smooth pitch */ if ((icorf & bit[2]) != 0) { - if ((i__1 = dpit[1] - dpit[0], (real) abs(i__1)) >= corth[ixcor - 1] + if ((i__1 = dpit[1] - dpit[0], (real) abs(i__1)) >= corth[ixcor - 1] && (i__2 = dpit[1] - dpit[2], (real) abs(i__2)) >= corth[ ixcor - 1]) { *pitch = median_(&dpit[2], &dpit[1], dpit); diff --git a/codecs/lpc10/difmag.c b/codecs/lpc10/difmag.c index ab59e8c9a..5132f96e9 100644 --- a/codecs/lpc10/difmag.c +++ b/codecs/lpc10/difmag.c @@ -87,7 +87,7 @@ extern int difmag_(real *speech, integer *lpita, integer *tau, integer *ltau, in /* This subroutine has no local state. */ -/* Subroutine */ int difmag_(real *speech, integer *lpita, integer *tau, +/* Subroutine */ int difmag_(real *speech, integer *lpita, integer *tau, integer *ltau, integer *maxlag, real *amdf, integer *minptr, integer * maxptr) { @@ -130,4 +130,3 @@ extern int difmag_(real *speech, integer *lpita, integer *tau, integer *ltau, in } return 0; } /* difmag_ */ - diff --git a/codecs/lpc10/dyptrk.c b/codecs/lpc10/dyptrk.c index 45fb5eb08..ca195b601 100644 --- a/codecs/lpc10/dyptrk.c +++ b/codecs/lpc10/dyptrk.c @@ -200,13 +200,13 @@ extern struct { /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -216,35 +216,35 @@ extern struct { /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -260,7 +260,7 @@ extern struct { /* removed. */ /* Local state */ /* It would be a bit more "general" to define S(LTAU), if Fortran */ -/* allows the argument of a function to be used as the dimension of +/* allows the argument of a function to be used as the dimension of */ /* a local array variable. */ /* IPOINT is always in the range 0 to DEPTH-1. */ @@ -272,16 +272,16 @@ extern struct { /* initial values (all indices of P with second index equal to */ /* IPTR are all written before being read in this subroutine). */ -/* From examining the code carefully, it appears that all of these +/* From examining the code carefully, it appears that all of these */ /* should be saved from one invocation to the next. */ /* I've run lpcsim with the "-l 6" option to see all of the */ /* debugging information that is printed out by this subroutine */ /* below, and it appears that S, P, IPOINT, and ALPHAX are all */ -/* initialized to 0 (these initial values would likely be different +/* initialized to 0 (these initial values would likely be different */ -/* on different platforms, compilers, etc.). Given that the output +/* on different platforms, compilers, etc.). Given that the output */ /* of the coder sounds reasonable, I'm going to initialize these */ /* variables to 0 explicitly. */ @@ -299,9 +299,9 @@ extern struct { /* Function Body */ -/* Calculate the confidence factor ALPHA, used as a threshold slope in +/* Calculate the confidence factor ALPHA, used as a threshold slope in */ -/* SEESAW. If unvoiced, set high slope so that every point in P array +/* SEESAW. If unvoiced, set high slope so that every point in P array */ /* is marked as a potential pitch frequency. A scaled up version (ALPHAX )*/ @@ -391,11 +391,11 @@ n*/ *pitch = p[*pitch + j * 60 - 61]; } -/* The following statement subtracts one from IPOINT, mod DEPTH. I +/* The following statement subtracts one from IPOINT, mod DEPTH. I */ -/* think the author chose to add DEPTH-1, instead of subtracting 1, +/* think the author chose to add DEPTH-1, instead of subtracting 1, */ -/* because then it will work even if MOD doesn't work as desired on +/* because then it will work even if MOD doesn't work as desired on */ /* negative arguments. */ diff --git a/codecs/lpc10/encode.c b/codecs/lpc10/encode.c index b81799f6b..f7b47762d 100644 --- a/codecs/lpc10/encode.c +++ b/codecs/lpc10/encode.c @@ -213,13 +213,13 @@ static integer c__2 = 2; /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -229,35 +229,35 @@ static integer c__2 = 2; /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -269,7 +269,7 @@ static integer c__2 = 2; /* common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */ /* Parameters/constants */ /* These arrays are not Fortran PARAMETER's, but they are defined */ -/* by DATA statements below, and their contents are never altered. +/* by DATA statements below, and their contents are never altered. */ /* Local variables that need not be saved */ /* Parameter adjustments */ @@ -370,4 +370,3 @@ static integer c__2 = 2; /* 801 FORMAT(1X,'<<ENCODE OUT>>',T32,2I3,I6,I5,T50,10I8) */ return 0; } /* encode_ */ - diff --git a/codecs/lpc10/energy.c b/codecs/lpc10/energy.c index eada04bef..eba35d89f 100644 --- a/codecs/lpc10/energy.c +++ b/codecs/lpc10/energy.c @@ -100,4 +100,3 @@ extern int energy_(integer *len, real *speech, real *rms); *rms = (real)sqrt(*rms / *len); return 0; } /* energy_ */ - diff --git a/codecs/lpc10/f2c.h b/codecs/lpc10/f2c.h index e50d642e0..09bc75727 100644 --- a/codecs/lpc10/f2c.h +++ b/codecs/lpc10/f2c.h @@ -267,18 +267,18 @@ extern integer i_nint(real *x); #endif #ifdef P_R_O_T_O_T_Y_P_E_S -extern int bsynz_(real *coef, integer *ip, integer *iv, +extern int bsynz_(real *coef, integer *ip, integer *iv, real *sout, real *rms, real *ratio, real *g2pass, struct lpc10_decoder_state *st); extern int chanwr_(integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits, struct lpc10_encoder_state *st); extern int chanrd_(integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits); -extern int chanwr_0_(int n__, integer *order, integer *ipitv, +extern int chanwr_0_(int n__, integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits, struct lpc10_encoder_state *st); extern int dcbias_(integer *len, real *speech, real *sigout); -extern int decode_(integer *ipitv, integer *irms, integer *irc, +extern int decode_(integer *ipitv, integer *irms, integer *irc, integer *voice, integer *pitch, real *rms, real *rc, struct lpc10_decoder_state *st); extern int deemp_(real *x, integer *n, struct lpc10_decoder_state *st); diff --git a/codecs/lpc10/ham84.c b/codecs/lpc10/ham84.c index fddd8f3c0..6fb1301cb 100644 --- a/codecs/lpc10/ham84.c +++ b/codecs/lpc10/ham84.c @@ -123,4 +123,3 @@ extern int ham84_(integer *input, integer *output, integer *errcnt); } return 0; } /* ham84_ */ - diff --git a/codecs/lpc10/invert.c b/codecs/lpc10/invert.c index 03c27e20d..786e4bf97 100644 --- a/codecs/lpc10/invert.c +++ b/codecs/lpc10/invert.c @@ -191,4 +191,3 @@ L100: /* END DO */ return 0; } /* invert_ */ - diff --git a/codecs/lpc10/irc2pc.c b/codecs/lpc10/irc2pc.c index b96ff0d66..dc3358b42 100644 --- a/codecs/lpc10/irc2pc.c +++ b/codecs/lpc10/irc2pc.c @@ -148,4 +148,3 @@ extern int irc2pc_(real *rc, real *pc, integer *order, real *gprime, real *g2pas } return 0; } /* irc2pc_ */ - diff --git a/codecs/lpc10/ivfilt.c b/codecs/lpc10/ivfilt.c index 784de2571..7a0bf6036 100644 --- a/codecs/lpc10/ivfilt.c +++ b/codecs/lpc10/ivfilt.c @@ -133,4 +133,3 @@ extern int ivfilt_(real *lpbuf, real *ivbuf, integer *len, integer *nsamp, real } return 0; } /* ivfilt_ */ - diff --git a/codecs/lpc10/lpc10.h b/codecs/lpc10/lpc10.h index a57f84f3f..82e7b2cba 100644 --- a/codecs/lpc10/lpc10.h +++ b/codecs/lpc10/lpc10.h @@ -87,7 +87,7 @@ struct lpc10_encoder_state { real z21; real z12; real z22; - + /* State used by function analys */ real inbuf[540], pebuf[540]; real lpbuf[696], ivbuf[312]; @@ -242,7 +242,7 @@ struct lpc10_decoder_state { (indices 0 through (LPC10_BITS_IN_COMPRESSED_FRAME-1)), and the array speech[] is written (indices 0 through (LPC10_SAMPLES_PER_FRAME-1)). - + */ struct lpc10_encoder_state * create_lpc10_encoder_state (void); diff --git a/codecs/lpc10/lpcdec.c b/codecs/lpc10/lpcdec.c index dd859ffce..55f8892b6 100644 --- a/codecs/lpc10/lpcdec.c +++ b/codecs/lpc10/lpcdec.c @@ -114,11 +114,11 @@ static integer c__10 = 10; struct lpc10_decoder_state *st) { integer irms, voice[2], pitch, ipitv; - extern /* Subroutine */ int decode_(integer *, integer *, integer *, + extern /* Subroutine */ int decode_(integer *, integer *, integer *, integer *, integer *, real *, real *, struct lpc10_decoder_state *); real rc[10]; - extern /* Subroutine */ int chanrd_(integer *, integer *, integer *, - integer *, integer *), synths_(integer *, + extern /* Subroutine */ int chanrd_(integer *, integer *, integer *, + integer *, integer *), synths_(integer *, integer *, real *, real *, real *, integer *, struct lpc10_decoder_state *); integer irc[10], len; @@ -220,13 +220,13 @@ static integer c__10 = 10; /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -236,35 +236,35 @@ static integer c__10 = 10; /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ diff --git a/codecs/lpc10/lpcenc.c b/codecs/lpc10/lpcenc.c index 989a2defd..990e73178 100644 --- a/codecs/lpc10/lpcenc.c +++ b/codecs/lpc10/lpcenc.c @@ -110,10 +110,10 @@ static integer c__10 = 10; { integer irms, voice[2], pitch, ipitv; real rc[10]; - extern /* Subroutine */ int encode_(integer *, integer *, real *, real *, - integer *, integer *, integer *), chanwr_(integer *, integer *, + extern /* Subroutine */ int encode_(integer *, integer *, real *, real *, + integer *, integer *, integer *), chanwr_(integer *, integer *, integer *, integer *, integer *, struct lpc10_encoder_state *), - analys_(real *, integer *, + analys_(real *, integer *, integer *, real *, real *, struct lpc10_encoder_state *), prepro_(real *, integer *, struct lpc10_encoder_state *); integer irc[10]; @@ -126,7 +126,7 @@ static integer c__10 = 10; * * Revision 1.14 2003/02/12 13:59:15 matteo * mer feb 12 14:56:57 CET 2003 - * + * * Revision 1.1.1.1 2003/02/12 13:59:15 matteo * mer feb 12 14:56:57 CET 2003 * diff --git a/codecs/lpc10/lpcini.c b/codecs/lpc10/lpcini.c index ea68176e3..ced34980f 100644 --- a/codecs/lpc10/lpcini.c +++ b/codecs/lpc10/lpcini.c @@ -187,13 +187,13 @@ struct { /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -203,35 +203,35 @@ struct { /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -280,7 +280,7 @@ void init_lpc10_encoder_state(struct lpc10_encoder_state *st) st->z21 = 0.0f; st->z12 = 0.0f; st->z22 = 0.0f; - + /* State used by function analys */ for (i = 0; i < 540; i++) { st->inbuf[i] = 0.0f; diff --git a/codecs/lpc10/lpfilt.c b/codecs/lpc10/lpfilt.c index 375528921..159b2d7d9 100644 --- a/codecs/lpc10/lpfilt.c +++ b/codecs/lpc10/lpfilt.c @@ -122,4 +122,3 @@ extern int lpfilt_(real *inbuf, real *lpbuf, integer *len, integer *nsamp); } return 0; } /* lpfilt_ */ - diff --git a/codecs/lpc10/median.c b/codecs/lpc10/median.c index 383c46e89..52c4a78d0 100644 --- a/codecs/lpc10/median.c +++ b/codecs/lpc10/median.c @@ -86,4 +86,3 @@ integer median_(integer *d1, integer *d2, integer *d3) } return ret_val; } /* median_ */ - diff --git a/codecs/lpc10/mload.c b/codecs/lpc10/mload.c index 1cdb0647c..e384f0d77 100644 --- a/codecs/lpc10/mload.c +++ b/codecs/lpc10/mload.c @@ -96,7 +96,7 @@ extern int mload_(integer *order, integer *awins, integer *awinf, real *speech, /* This subroutine has no local state. */ -/* Subroutine */ int mload_(integer *order, integer *awins, integer *awinf, +/* Subroutine */ int mload_(integer *order, integer *awins, integer *awinf, real *speech, real *phi, real *psi) { /* System generated locals */ @@ -136,21 +136,21 @@ extern int mload_(integer *order, integer *awins, integer *awinf, real *speech, for (r__ = 2; r__ <= i__1; ++r__) { i__2 = r__; for (c__ = 2; c__ <= i__2; ++c__) { - phi[r__ + c__ * phi_dim1] = phi[r__ - 1 + (c__ - 1) * phi_dim1] - - speech[*awinf + 1 - r__] * speech[*awinf + 1 - c__] + + phi[r__ + c__ * phi_dim1] = phi[r__ - 1 + (c__ - 1) * phi_dim1] - + speech[*awinf + 1 - r__] * speech[*awinf + 1 - c__] + speech[start - r__] * speech[start - c__]; } } /* End correct to get additional elements of PSI */ i__1 = *order - 1; for (c__ = 1; c__ <= i__1; ++c__) { - psi[c__] = phi[c__ + 1 + phi_dim1] - speech[start - 1] * speech[start + psi[c__] = phi[c__ + 1 + phi_dim1] - speech[start - 1] * speech[start - 1 - c__] + speech[*awinf] * speech[*awinf - c__]; } /* Copy lower triangular section into upper (why bother?) */ -/* I'm commenting this out, since the upper triangular half of PHI +/* I'm commenting this out, since the upper triangular half of PHI */ -/* is never used by later code, unless a sufficiently high level of +/* is never used by later code, unless a sufficiently high level of */ /* tracing is turned on. */ /* DO R = 1,ORDER */ @@ -160,4 +160,3 @@ extern int mload_(integer *order, integer *awins, integer *awinf, real *speech, /* END DO */ return 0; } /* mload_ */ - diff --git a/codecs/lpc10/onset.c b/codecs/lpc10/onset.c index ddca3b477..b4469c16a 100644 --- a/codecs/lpc10/onset.c +++ b/codecs/lpc10/onset.c @@ -203,35 +203,35 @@ static real c_b2 = 1.f; /* N, D Numerator and denominator of prediction filters */ /* FPC Current prediction coefs */ /* L2BUF, L2SUM1, L2SUM2 State of slope filter */ -/* The only "significant" change I've made is to change L2SUM2 out +/* The only "significant" change I've made is to change L2SUM2 out */ /* of the list of local variables that need to be saved, since it */ /* didn't need to be. */ -/* L2SUM1 need not be, but avoiding saving it would require a small +/* L2SUM1 need not be, but avoiding saving it would require a small */ /* change to the body of the code. See comments below for an */ -/* example of how the code could be changed to avoid saving L2SUM1. +/* example of how the code could be changed to avoid saving L2SUM1. */ /* FPC and LASTI are saved from one invocation to the next, but */ -/* they are not given initial values. This is acceptable, because +/* they are not given initial values. This is acceptable, because */ /* FPC will be assigned a value the first time that this function */ /* is called after D is initialized to 1, since the formula to */ /* change D will not change it to 0 in one step, and the IF (D */ -/* .NE. 0) statement will execute its THEN part, initializing FPC. +/* .NE. 0) statement will execute its THEN part, initializing FPC. */ /* LASTI's value will not be used until HYST is .TRUE., and */ /* whenever HYST is changed from its initial value of .FALSE., */ /* LASTI is assigned a value. */ /* In a C version of this coder, it would be nice if all of these */ -/* saved things, in this and all other subroutines, could be stored +/* saved things, in this and all other subroutines, could be stored */ -/* in a single struct lpc10_coder_state_t, initialized with a call +/* in a single struct lpc10_coder_state_t, initialized with a call */ -/* to a function like lpc10_init(&lpc10_coder_state). In this way, +/* to a function like lpc10_init(&lpc10_coder_state). In this way, */ -/* a program that used these functions could conveniently alternate +/* a program that used these functions could conveniently alternate */ /* coding more than one distinct audio stream. */ @@ -264,7 +264,7 @@ static real c_b2 = 1.f; } i__1 = *sbufh; for (i__ = *sbufh - *lframe + 1; i__ <= i__1; ++i__) { -/* Compute FPC; Use old FPC on divide by zero; Clamp FPC to +/- 1. +/* Compute FPC; Use old FPC on divide by zero; Clamp FPC to +/- 1. */ *n = (pebuf[i__] * pebuf[i__ - 1] + (*n) * 63.f) / 64.f; /* Computing 2nd power */ @@ -278,11 +278,11 @@ static real c_b2 = 1.f; } } /* Filter FPC */ -/* In order to allow L2SUM1 not to be saved from one invocation +/* In order to allow L2SUM1 not to be saved from one invocation of */ /* this subroutine to the next, one could change the sequence of */ -/* assignments below, up to the IF statement, to the following. +/* assignments below, up to the IF statement, to the following. In */ /* addition, the initial value of L2PTR2 should be changed to */ /* L2WID/2 instead of L2WID/2+1. */ diff --git a/codecs/lpc10/pitsyn.c b/codecs/lpc10/pitsyn.c index 36f6f2a7a..e345fba10 100644 --- a/codecs/lpc10/pitsyn.c +++ b/codecs/lpc10/pitsyn.c @@ -129,8 +129,8 @@ extern int pitsyn_(integer *order, integer *voice, integer *pitch, real *rms, re /* RATIO - Previous to present energy ratio */ /* Always assigned a value. */ -/* Subroutine */ int pitsyn_(integer *order, integer *voice, - integer *pitch, real *rms, real *rc, integer *lframe, integer *ivuv, +/* Subroutine */ int pitsyn_(integer *order, integer *voice, + integer *pitch, real *rms, real *rc, integer *lframe, integer *ivuv, integer *ipiti, real *rmsi, real *rci, integer *nout, real *ratio, struct lpc10_decoder_state *st) { @@ -201,7 +201,7 @@ extern int pitsyn_(integer *order, integer *voice, integer *pitch, real *rms, re /* Frame size, Prediction order, Pitch period */ /* Local variables that need not be saved */ /* LSAMP is initialized in the IF (FIRST) THEN clause, but it is */ -/* not used the first time through, and it is given a value before +/* not used the first time through, and it is given a value before */ /* use whenever FIRST is .FALSE., so it appears unnecessary to */ /* assign it a value when FIRST is .TRUE. */ @@ -214,12 +214,12 @@ extern int pitsyn_(integer *order, integer *voice, integer *pitch, real *rms, re /* JSAMP - If this routine is called N times with identical values of */ /* LFRAME, then the total length of all pitch periods returned */ -/* is always N*LFRAME-JSAMP, and JSAMP is always in the range 0 +/* is always N*LFRAME-JSAMP, and JSAMP is always in the range 0 */ /* to MAXPIT-1 (see below for why this is so). Thus JSAMP is */ /* the number of samples "left over" from the previous call to */ /* PITSYN, that haven't been "used" in a pitch period returned */ -/* from this subroutine. Every time this subroutine is called, +/* from this subroutine. Every time this subroutine is called, */ /* it returns pitch periods with a total length of at most */ /* LFRAME+JSAMP. */ @@ -277,7 +277,7 @@ extern int pitsyn_(integer *order, integer *voice, integer *pitch, real *rms, re *nout = *lframe / *pitch; *jsamp = *lframe - *nout * *pitch; -/* SYNTHS only calls this subroutine with PITCH in the range +/* SYNTHS only calls this subroutine with PITCH in the range 20 */ /* to 156. LFRAME = MAXFRM = 180, so NOUT is somewhere in th e */ @@ -361,7 +361,7 @@ e */ vflag = 1; } } -/* Here is the value of most variables that are used below, depending +/* Here is the value of most variables that are used below, depending on */ /* the values of IVOICO, VOICE(1), and VOICE(2). VOICE(1) and VOICE(2 ) */ @@ -369,11 +369,11 @@ on */ /* previous call (see notes for the IF (NOUT .NE. 0) statement near th e */ /* end). Each of these three values is either 0 or 1. These three */ -/* values below are given as 3-bit long strings, in the order IVOICO, +/* values below are given as 3-bit long strings, in the order IVOICO, */ /* VOICE(1), and VOICE(2). It appears that the code above assumes tha t */ -/* the bit sequences 010 and 101 never occur, but I wonder whether a +/* the bit sequences 010 and 101 never occur, but I wonder whether a */ /* large enough number of bit errors in the channel could cause such a */ @@ -390,7 +390,7 @@ t */ 4, */ /* and the 45 for NL-JSAMP is actually LFRAME-3*LFRAME/4. */ -/* Note that LSAMP-JSAMP is given as the variable. This was just for +/* Note that LSAMP-JSAMP is given as the variable. This was just for */ /* brevity, to avoid adding "+JSAMP" to all of the column entries. */ /* Similarly for NL-JSAMP. */ @@ -429,13 +429,13 @@ t */ /* The only possible non-0 value of SLOPE (in column 111) is */ /* (PITCH-IPITO)/FLOAT(LSAMP) */ -/* Column 101 is identical to 100. Any good properties we can prove +/* Column 101 is identical to 100. Any good properties we can prove */ /* for 100 will also hold for 101. Similarly for 010 and 011. */ -/* SYNTHS calls this subroutine with PITCH restricted to the range 20 +/* SYNTHS calls this subroutine with PITCH restricted to the range 20 to */ -/* 156. IPITO is similarly restricted to this range, after the first +/* 156. IPITO is similarly restricted to this range, after the first */ /* call. IP below is also restricted to this range, given the */ /* definitions of IPITO, SLOPE, UVPIT, and that I is in the range ISTA @@ -456,7 +456,7 @@ ugh */ /* (I - MAXPIT) .LE. JUSED .LE. (I-1) */ -/* Note that the final value of I is LSAMP+1, so that +/* Note that the final value of I is LSAMP+1, so that after */ /* the DO loop is complete, we know: */ @@ -474,10 +474,10 @@ after */ /* The following check is no longer nece ssary, now that */ -/* we can prove that NOUT will never go +/* we can prove that NOUT will never go over 16. */ -/* IF (NOUT .GT. 16) STOP 'PITSYN: too many epochs' +/* IF (NOUT .GT. 16) STOP 'PITSYN: too many epochs' */ ipiti[*nout] = ip; @@ -501,7 +501,7 @@ over 16. */ goto L100; } -/* I want to prove what range UVPIT must lie in after +/* I want to prove what range UVPIT must lie in after the */ /* assignments to it below. To do this, I must determ ine */ @@ -521,7 +521,7 @@ at: */ /* ISTART is one more than this. */ -/* Let newLSAMP be the value assigned to LSAMP below. +/* Let newLSAMP be the value assigned to LSAMP below. This */ /* is 180+JSAMP. Thus (newLSAMP-oldLSAMP) is either 4 5 or */ @@ -566,7 +566,7 @@ ge. */ L100: *jsamp = lsamp - jused; } -/* Given that the maximum pitch period MAXPIT .LT. LFRAME (this is +/* Given that the maximum pitch period MAXPIT .LT. LFRAME (this is */ /* currently true on every call, since SYNTHS always sets */ /* LFRAME=180), NOUT will always be .GE. 1 at this point. */ diff --git a/codecs/lpc10/placea.c b/codecs/lpc10/placea.c index dacb50e7a..7a485c418 100644 --- a/codecs/lpc10/placea.c +++ b/codecs/lpc10/placea.c @@ -114,7 +114,7 @@ extern int placea_(integer *ipitch, integer *voibuf, integer *obound, integer *a /* This subroutine has no local state. */ /* Subroutine */ int placea_(integer *ipitch, integer *voibuf, integer * - obound, integer *af, integer *vwin, integer *awin, integer *ewin, + obound, integer *af, integer *vwin, integer *awin, integer *ewin, integer *lframe, integer *maxwin) { /* System generated locals */ @@ -146,13 +146,13 @@ extern int placea_(integer *ipitch, integer *voibuf, integer *obound, integer *a /* Case 1: Sustained Voiced Speech */ /* If the five most recent voicing decisions are */ /* voiced, then the window is placed phase-synchronously with the */ -/* previous window, as close to the present voicing window if possible. +/* previous window, as close to the present voicing window if possible. */ /* If onsets bound the voicing window, then preference is given to */ /* a phase-synchronous placement which does not overlap these onsets. */ /* Case 2: Voiced Transition */ -/* If at least one voicing decision in AF is voicied, and there are no +/* If at least one voicing decision in AF is voicied, and there are no */ /* onsets, then the window is placed as in case 1. */ @@ -207,7 +207,7 @@ rd */ awin[(*af << 1) + 1] -= *ipitch; awin[(*af << 1) + 2] -= *ipitch; } -/* Similarly if the placement puts the analysis window below LRANGE. +/* Similarly if the placement puts the analysis window below LRANGE. */ while(awin[(*af << 1) + 1] < lrange) { awin[(*af << 1) + 1] += *ipitch; @@ -239,4 +239,3 @@ e*/ } return 0; } /* placea_ */ - diff --git a/codecs/lpc10/placev.c b/codecs/lpc10/placev.c index 56e72c4b8..9d1548c46 100644 --- a/codecs/lpc10/placev.c +++ b/codecs/lpc10/placev.c @@ -109,8 +109,8 @@ extern int placev_(integer *osbuf, integer *osptr, integer *oslen, integer *obou /* This subroutine has no local state. */ -/* Subroutine */ int placev_(integer *osbuf, integer *osptr, integer *oslen, - integer *obound, integer *vwin, integer *af, integer *lframe, integer +/* Subroutine */ int placev_(integer *osbuf, integer *osptr, integer *oslen, + integer *obound, integer *vwin, integer *af, integer *lframe, integer *minwin, integer *maxwin, integer *dvwinl, integer *dvwinh) { /* System generated locals */ @@ -146,15 +146,15 @@ extern int placev_(integer *osbuf, integer *osptr, integer *oslen, integer *obou /* given in the LPC-10e phase 1 report. */ /* 1. If there are no onsets in this range, then the voicing window */ -/* is centered in the pitch window. If such a placement is not within +/* is centered in the pitch window. If such a placement is not within */ -/* the window's placement range, then the window is placed in the left- +/* the window's placement range, then the window is placed in the left- */ /* most portion of the placement range. Its length is always MAXWIN. */ /* 2. If the first onset is in 2F and there is sufficient room to place */ -/* the window immediately before this onset, then the window is placed +/* the window immediately before this onset, then the window is placed */ /* there, and its length is set to the maximum possible under these */ /* constraints. */ @@ -177,7 +177,7 @@ ing*/ /* Note that the values of MINWIN and LFRAME must be chosen such */ /* that case 2 = false implies case 3 = true. This means that */ /* MINWIN <= LFRAME/2. If this were not the case, then a fourth case */ -/* would have to be added for when the window cannot fit either before +/* would have to be added for when the window cannot fit either before */ /* or after the onset. */ @@ -185,11 +185,11 @@ ing*/ */ /* time, due to the filter delays in computing onsets. The result is tha t*/ -/* occasionally a voicing window will overlap that onset. The only way +/* occasionally a voicing window will overlap that onset. The only way */ -/* to circumvent this problem is to add more delay in processing input +/* to circumvent this problem is to add more delay in processing input */ -/* speech. In the trade-off between delay and window-placement, window +/* speech. In the trade-off between delay and window-placement, window */ /* placement lost. */ /* Compute the placement range */ @@ -272,4 +272,3 @@ L120: } return 0; } /* placev_ */ - diff --git a/codecs/lpc10/preemp.c b/codecs/lpc10/preemp.c index 645428c3c..61316dd5b 100644 --- a/codecs/lpc10/preemp.c +++ b/codecs/lpc10/preemp.c @@ -110,12 +110,12 @@ extern int preemp_(real *inbuf, real *pebuf, integer *nsamp, real *coef, real *z /* Could it be more efficient to apply multiple filters */ /* simultaneously, by combining them into one equivalent filter? */ -/* Are there ever cases when "factoring" one high-order filter into +/* Are there ever cases when "factoring" one high-order filter into */ /* multiple smaller-order filter actually reduces the number of */ /* arithmetic operations needed to perform them? */ /* When I first read this subroutine, I didn't understand why the */ -/* variable temp was used. It seemed that the statements in the do +/* variable temp was used. It seemed that the statements in the do */ /* loop could be replaced with the following: */ @@ -141,4 +141,3 @@ extern int preemp_(real *inbuf, real *pebuf, integer *nsamp, real *coef, real *z } return 0; } /* preemp_ */ - diff --git a/codecs/lpc10/random.c b/codecs/lpc10/random.c index 0f8e9b209..dc4da2a91 100644 --- a/codecs/lpc10/random.c +++ b/codecs/lpc10/random.c @@ -122,4 +122,3 @@ integer random_(struct lpc10_decoder_state *st) } return ret_val; } /* random_ */ - diff --git a/codecs/lpc10/rcchk.c b/codecs/lpc10/rcchk.c index 6cb76ef7d..4cc84f381 100644 --- a/codecs/lpc10/rcchk.c +++ b/codecs/lpc10/rcchk.c @@ -102,7 +102,7 @@ extern int rcchk_(integer *order, real *rc1f, real *rc2f); } } return 0; -/* Note: In version embedded in other software, all calls to ERROR +/* Note: In version embedded in other software, all calls to ERROR */ /* should probably be removed. */ L10: @@ -116,4 +116,3 @@ L10: } return 0; } /* rcchk_ */ - diff --git a/codecs/lpc10/synths.c b/codecs/lpc10/synths.c index 4c5a70fac..2a3179b91 100644 --- a/codecs/lpc10/synths.c +++ b/codecs/lpc10/synths.c @@ -187,8 +187,8 @@ static real c_b2 = .7f; integer ipiti[16]; real g2pass; real pc[10]; - extern /* Subroutine */ int pitsyn_(integer *, integer *, integer *, real - *, real *, integer *, integer *, integer *, real *, real *, + extern /* Subroutine */ int pitsyn_(integer *, integer *, integer *, real + *, real *, integer *, integer *, integer *, real *, real *, integer *, real *, struct lpc10_decoder_state *); real rci[160] /* was [10][16] */; @@ -288,13 +288,13 @@ static real c_b2 = .7f; /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -304,35 +304,35 @@ static real c_b2 = .7f; /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -345,21 +345,21 @@ static real c_b2 = .7f; /* Parameters/constants */ /* Local variables that need not be saved */ /* Local state */ -/* BUF is a buffer of speech samples that would have been returned +/* BUF is a buffer of speech samples that would have been returned */ /* by the older version of SYNTHS, but the newer version doesn't, */ /* so that the newer version can always return MAXFRM samples on */ /* every call. This has the effect of delaying the return of */ /* samples for one additional frame time. */ -/* Indices 1 through BUFLEN contain samples that are left over from +/* Indices 1 through BUFLEN contain samples that are left over from */ /* the last call to SYNTHS. Given the way that PITSYN works, */ /* BUFLEN should always be in the range MAXFRM-MAXPIT+1 through */ /* MAXFRM, inclusive, after a call to SYNTHS is complete. */ /* On the first call to SYNTHS (or the first call after */ -/* reinitializing with the entry INITSYNTHS), BUFLEN is MAXFRM, and +/* reinitializing with the entry INITSYNTHS), BUFLEN is MAXFRM, and */ /* a frame of silence is always returned. */ /* Parameter adjustments */ @@ -388,7 +388,7 @@ static real c_b2 = .7f; r__1 = min(r__2,.99f); rc[i__] = max(r__1,-.99f); } - pitsyn_(&contrl_1.order, &voice[1], pitch, rms, &rc[1], &contrl_1.lframe, + pitsyn_(&contrl_1.order, &voice[1], pitch, rms, &rc[1], &contrl_1.lframe, ivuv, ipiti, rmsi, rci, &nout, &ratio, st); if (nout > 0) { i__1 = nout; @@ -405,7 +405,7 @@ d of */ *buflen += ipiti[j - 1]; } -/* Copy first MAXFRM samples from BUF to output array SPEECH +/* Copy first MAXFRM samples from BUF to output array SPEECH */ /* (scaling them), and then remove them from the beginning of */ diff --git a/codecs/lpc10/tbdm.c b/codecs/lpc10/tbdm.c index 2f6f3d692..4ca4d73a3 100644 --- a/codecs/lpc10/tbdm.c +++ b/codecs/lpc10/tbdm.c @@ -91,7 +91,7 @@ extern int tbdm_(real *speech, integer *lpita, integer *tau, integer *ltau, real /* This subroutine has no local state. */ -/* Subroutine */ int tbdm_(real *speech, integer *lpita, integer *tau, +/* Subroutine */ int tbdm_(real *speech, integer *lpita, integer *tau, integer *ltau, real *amdf, integer *minptr, integer *maxptr, integer * mintau) { @@ -101,7 +101,7 @@ extern int tbdm_(real *speech, integer *lpita, integer *tau, integer *ltau, real /* Local variables */ real amdf2[6]; integer minp2, ltau2, maxp2, i__; - extern /* Subroutine */ int difmag_(real *, integer *, integer *, integer + extern /* Subroutine */ int difmag_(real *, integer *, integer *, integer *, integer *, real *, integer *, integer *); integer minamd, ptr, tau2[6]; @@ -118,7 +118,7 @@ extern int tbdm_(real *speech, integer *lpita, integer *tau, integer *ltau, real --tau; /* Function Body */ - difmag_(&speech[1], lpita, &tau[1], ltau, &tau[*ltau], &amdf[1], minptr, + difmag_(&speech[1], lpita, &tau[1], ltau, &tau[*ltau], &amdf[1], minptr, maxptr); *mintau = tau[*minptr]; minamd = (integer)amdf[*minptr]; @@ -185,4 +185,3 @@ extern int tbdm_(real *speech, integer *lpita, integer *tau, integer *ltau, real } return 0; } /* tbdm_ */ - diff --git a/codecs/lpc10/voicin.c b/codecs/lpc10/voicin.c index 3605d2f2e..2d36e2c99 100644 --- a/codecs/lpc10/voicin.c +++ b/codecs/lpc10/voicin.c @@ -256,8 +256,8 @@ s*/ /* INITVOICIN. */ /* Subroutine */ int voicin_(integer *vwin, real *inbuf, real * - lpbuf, integer *buflim, integer *half, real *minamd, real *maxamd, - integer *mintau, real *ivrc, integer *obound, integer *voibuf, + lpbuf, integer *buflim, integer *half, real *minamd, real *maxamd, + integer *mintau, real *ivrc, integer *obound, integer *voibuf, integer *af, struct lpc10_encoder_state *st) { /* Initialized data */ @@ -298,8 +298,8 @@ s*/ real *maxmin; integer vstate; real rc1; - extern /* Subroutine */ int vparms_(integer *, real *, real *, integer *, - integer *, real *, integer *, integer *, integer *, integer *, + extern /* Subroutine */ int vparms_(integer *, real *, real *, integer *, + integer *, real *, integer *, integer *, integer *, integer *, real *, real *, real *, real *); integer fbe, lbe; real *snr; @@ -363,13 +363,13 @@ s*/ /* Error correction */ /* Subroutine SETUP is the only place where order is assigned a value, */ /* and that value is 10. It could increase efficiency 1% or so to */ -/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as +/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as */ /* a variable in a COMMON block, since it is used in many places in the */ -/* core of the coding and decoding routines. Actually, I take that back. +/* core of the coding and decoding routines. Actually, I take that back. */ /* At least when compiling with f2c, the upper bound of DO loops is */ -/* stored in a local variable before the DO loop begins, and then that is +/* stored in a local variable before the DO loop begins, and then that is */ /* compared against on each iteration. */ /* Similarly for lframe, which is given a value of MAXFRM in SETUP. */ @@ -379,35 +379,35 @@ s*/ /* nbits is similar to quant, and is given a value of 54 in SETUP. */ /* corrp is given a value of .TRUE. in SETUP, and is only used in the */ /* subroutines ENCODE and DECODE. It doesn't affect the speed of the */ -/* coder significantly whether it is .TRUE. or .FALSE., or whether it is +/* coder significantly whether it is .TRUE. or .FALSE., or whether it is */ /* a constant or a variable, since it is only examined once per frame. */ /* Leaving it as a variable that is set to .TRUE. seems like a good */ /* idea, since it does enable some error-correction capability for */ -/* unvoiced frames, with no change in the coding rate, and no noticeable +/* unvoiced frames, with no change in the coding rate, and no noticeable */ /* quality difference in the decoded speech. */ /* integer quant, nbits */ -/* *** Read/write: variables for debugging, not needed for LPC algorithm +/* *** Read/write: variables for debugging, not needed for LPC algorithm */ -/* Current frame, Unstable frames, Output clip count, Max onset buffer, +/* Current frame, Unstable frames, Output clip count, Max onset buffer, */ /* Debug listing detail level, Line count on listing page */ /* nframe is not needed for an embedded LPC10 at all. */ /* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */ /* ERROR, which is only called from RCCHK. When LPC10 is embedded into */ -/* an application, I would recommend removing the call to ERROR in RCCHK, +/* an application, I would recommend removing the call to ERROR in RCCHK, */ /* and remove ERROR and nunsfm completely. */ -/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in +/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in */ /* sread.f. When LPC10 is embedded into an application, one might want */ /* to cause it to be incremented in a routine that takes the output of */ /* SYNTHS and sends it to an audio device. It could be optionally */ /* displayed, for those that might want to know what it is. */ -/* maxosp is never initialized to 0 in SETUP, although it probably should +/* maxosp is never initialized to 0 in SETUP, although it probably should */ /* be, and it is updated in subroutine ANALYS. I doubt that its value */ /* would be of much interest to an application in which LPC10 is */ @@ -418,7 +418,7 @@ s*/ /* common /contrl/ quant, nbits */ /* common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */ /* Parameters/constants */ -/* Voicing coefficient and Linear Discriminant Analysis variables: +/* Voicing coefficient and Linear Discriminant Analysis variables: */ /* Max number of VDC's and VDC levels */ /* The following are not Fortran PARAMETER's, but they are */ @@ -428,15 +428,15 @@ s*/ /* Note: */ /* VALUE(1) through VALUE(8) are assigned values, but VALUE(9) */ -/* never is. Yet VALUE(9) is read in the loop that begins "DO I = +/* never is. Yet VALUE(9) is read in the loop that begins "DO I = */ -/* 1, 9" below. I believe that this doesn't cause any problems in +/* 1, 9" below. I believe that this doesn't cause any problems in */ -/* this subroutine, because all VDC(9,*) array elements are 0, and +/* this subroutine, because all VDC(9,*) array elements are 0, and */ -/* this is what is multiplied by VALUE(9) in all cases. Still, it +/* this is what is multiplied by VALUE(9) in all cases. Still, it */ -/* would save a multiplication to change the loop to "DO I = 1, 8". +/* would save a multiplication to change the loop to "DO I = 1, 8". */ /* Local state */ /* WARNING! */ @@ -450,16 +450,16 @@ s*/ /* For VOICE, note that it is "shifted" in the statement that */ /* begins "IF (HALF .EQ. 1) THEN" below. Also, uninitialized */ -/* values in the VOICE array can only affect entries in the VOIBUF +/* values in the VOICE array can only affect entries in the VOIBUF */ -/* array that are for the same frame, or for an older frame. Thus +/* array that are for the same frame, or for an older frame. Thus */ /* the effects of uninitialized values in VOICE cannot linger on */ /* for more than 2 or 3 frame times. */ /* For SFBUE and SLBUE, the effects of uninitialized values can */ /* linger on for many frame times, because their previous values */ -/* are exponentially decayed. Thus it is more important to choose +/* are exponentially decayed. Thus it is more important to choose */ /* initial values for these variables. I would guess that a */ /* reasonable initial value for SFBUE is REF/16, the same as used */ @@ -521,7 +521,7 @@ s*/ /* LBVE, LBUE, FBVE, FBUE, OFBUE, OLBUE */ /* MAXMIN is initialized on the first call, assuming that HALF */ -/* .EQ. 1 on first call. This is how ANALYS calls this subroutine. +/* .EQ. 1 on first call. This is how ANALYS calls this subroutine. */ /* Voicing Decision Parameter vector (* denotes zero coefficient): */ @@ -544,7 +544,7 @@ fic*/ /* The VOICE array contains the result of the linear discriminant functio n*/ -/* (analog values). The VOIBUF array contains the hard-limited binary +/* (analog values). The VOIBUF array contains the hard-limited binary */ /* voicing decisions. The VOICE and VOIBUF arrays, according to FORTRAN */ @@ -564,10 +564,10 @@ n*/ *maxmin = *maxamd / max(*minamd,1.f); } /* Calculate voicing parameters twice per frame: */ - vparms_(&vwin[1], &inbuf[inbuf_offset], &lpbuf[lpbuf_offset], &buflim[1], + vparms_(&vwin[1], &inbuf[inbuf_offset], &lpbuf[lpbuf_offset], &buflim[1], half, dither, mintau, &zc, &lbe, &fbe, &qs, &rc1, &ar_b__, & ar_f__); -/* Estimate signal-to-noise ratio to select the appropriate VDC vector. +/* Estimate signal-to-noise ratio to select the appropriate VDC vector. */ /* The SNR is estimated as the running average of the ratio of the */ /* running average full-band voiced energy to the running average */ @@ -607,10 +607,10 @@ L69: voibuf[*half + 6] = 0; } /* Skip voicing decision smoothing in first half-frame: */ -/* Give a value to VSTATE, so that trace statements below will print +/* Give a value to VSTATE, so that trace statements below will print */ /* a consistent value from one call to the next when HALF .EQ. 1. */ -/* The value of VSTATE is not used for any other purpose when this is +/* The value of VSTATE is not used for any other purpose when this is */ /* true. */ vstate = -1; @@ -631,7 +631,7 @@ L69: .*/ /* Voicing override of transitions at onsets: */ -/* If a V/UV or UV/V voicing decision transition occurs within one-half +/* If a V/UV or UV/V voicing decision transition occurs within one-half */ /* frame of an onset bounding a voicing window, then the transition is */ /* moved to occur at the onset. */ diff --git a/codecs/lpc10/vparms.c b/codecs/lpc10/vparms.c index c75b1b17d..f802e6ffd 100644 --- a/codecs/lpc10/vparms.c +++ b/codecs/lpc10/vparms.c @@ -131,8 +131,8 @@ static real c_b2 = 1.f; /* This subroutine has no local state. */ -/* Subroutine */ int vparms_(integer *vwin, real *inbuf, real *lpbuf, integer - *buflim, integer *half, real *dither, integer *mintau, integer *zc, +/* Subroutine */ int vparms_(integer *vwin, real *inbuf, real *lpbuf, integer + *buflim, integer *half, real *dither, integer *mintau, integer *zc, integer *lbe, integer *fbe, real *qs, real *rc1, real *ar_b__, real * ar_f__) { @@ -252,4 +252,3 @@ is)*/ *fbe = min(i__1,32767); return 0; } /* vparms_ */ - diff --git a/codecs/speex/arch.h b/codecs/speex/arch.h index af42e645d..435befcef 100644 --- a/codecs/speex/arch.h +++ b/codecs/speex/arch.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -219,11 +219,11 @@ typedef float spx_word32_t; #if defined (CONFIG_TI_C54X) || defined (CONFIG_TI_C55X) /* 2 on TI C5x DSP */ -#define BYTES_PER_CHAR 2 +#define BYTES_PER_CHAR 2 #define BITS_PER_CHAR 16 #define LOG2_BITS_PER_CHAR 4 -#else +#else #define BYTES_PER_CHAR 1 #define BITS_PER_CHAR 8 diff --git a/codecs/speex/fixed_generic.h b/codecs/speex/fixed_generic.h index 3fb096ed9..0b219188d 100644 --- a/codecs/speex/fixed_generic.h +++ b/codecs/speex/fixed_generic.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR diff --git a/codecs/speex/resample.c b/codecs/speex/resample.c index 2b0395180..0797352ad 100644 --- a/codecs/speex/resample.c +++ b/codecs/speex/resample.c @@ -1,6 +1,6 @@ /* Copyright (C) 2007-2008 Jean-Marc Valin Copyright (C) 2008 Thorvald Natvig - + File: resample.c Arbitrary resampling code @@ -38,22 +38,22 @@ - Low memory requirement - Good *perceptual* quality (and not best SNR) - Warning: This resampler is relatively new. Although I think I got rid of + Warning: This resampler is relatively new. Although I think I got rid of all the major bugs and I don't expect the API to change anymore, there may be something I've missed. So use with caution. This algorithm is based on this original resampling algorithm: Smith, Julius O. Digital Audio Resampling Home Page - Center for Computer Research in Music and Acoustics (CCRMA), + Center for Computer Research in Music and Acoustics (CCRMA), Stanford University, 2007. Web published at http://www-ccrma.stanford.edu/~jos/resample/. - There is one main difference, though. This resampler uses cubic + There is one main difference, though. This resampler uses cubic interpolation instead of linear interpolation in the above paper. This makes the table much smaller and makes it possible to compute that table - on a per-stream basis. In turn, being able to tweak the table for each - stream makes it possible to both reduce complexity on simple ratios - (e.g. 2/3), and get rid of the rounding operations in the inner loop. + on a per-stream basis. In turn, being able to tweak the table for each + stream makes it possible to both reduce complexity on simple ratios + (e.g. 2/3), and get rid of the rounding operations in the inner loop. The latter both reduces CPU time and makes the algorithm more SIMD-friendly. */ @@ -106,7 +106,7 @@ struct SpeexResamplerState_ { spx_uint32_t out_rate; spx_uint32_t num_rate; spx_uint32_t den_rate; - + int quality; spx_uint32_t nb_channels; spx_uint32_t filt_len; @@ -118,17 +118,17 @@ struct SpeexResamplerState_ { spx_uint32_t oversample; int initialised; int started; - + /* These are per-channel */ spx_int32_t *last_sample; spx_uint32_t *samp_frac_num; spx_uint32_t *magic_samples; - + spx_word16_t *mem; spx_word16_t *sinc_table; spx_uint32_t sinc_table_length; resampler_basic_func resampler_ptr; - + int in_stride; int out_stride; } ; @@ -170,7 +170,7 @@ static double kaiser8_table[36] = { 0.32108304, 0.27619388, 0.23465776, 0.19672670, 0.16255380, 0.13219758, 0.10562887, 0.08273982, 0.06335451, 0.04724088, 0.03412321, 0.02369490, 0.01563093, 0.00959968, 0.00527363, 0.00233883, 0.00050000, 0.00000000}; - + static double kaiser6_table[36] = { 0.99733006, 1.00000000, 0.99733006, 0.98935595, 0.97618418, 0.95799003, 0.93501423, 0.90755855, 0.87598009, 0.84068475, 0.80211977, 0.76076565, @@ -183,7 +183,7 @@ struct FuncDef { double *table; int oversample; }; - + static struct FuncDef _KAISER12 = {kaiser12_table, 64}; #define KAISER12 (&_KAISER12) /*static struct FuncDef _KAISER12 = {kaiser12_table, 32}; @@ -205,7 +205,7 @@ struct QualityMapping { /* This table maps conversion quality to internal parameters. There are two - reasons that explain why the up-sampling bandwidth is larger than the + reasons that explain why the up-sampling bandwidth is larger than the down-sampling bandwidth: 1) When up-sampling, we can assume that the spectrum is already attenuated close to the Nyquist rate (from an A/D or a previous resampling filter) @@ -231,7 +231,7 @@ static double compute_func(float x, struct FuncDef *func) { float y, frac; double interp[4]; - int ind; + int ind; y = x*func->oversample; ind = (int)floor(y); frac = (y-ind); @@ -242,7 +242,7 @@ static double compute_func(float x, struct FuncDef *func) interp[0] = -0.3333333333*frac + 0.5*(frac*frac) - 0.1666666667*(frac*frac*frac); /* Just to make sure we don't have rounding problems */ interp[1] = 1.f-interp[3]-interp[2]-interp[0]; - + /*sum = frac*accum[1] + (1-frac)*accum[2];*/ return interp[0]*func->table[ind] + interp[1]*func->table[ind+1] + interp[2]*func->table[ind+2] + interp[3]*func->table[ind+3]; } @@ -461,7 +461,7 @@ static int resampler_basic_interpolate_single(SpeexResamplerState *st, spx_uint3 cubic_coef(frac, interp); sum = interpolate_product_single(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp); #endif - + out[out_stride * out_sample++] = PSHR32(sum,15); last_sample += int_advance; samp_frac_num += frac_advance; @@ -523,7 +523,7 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3 cubic_coef(frac, interp); sum = interpolate_product_double(iptr, st->sinc_table + st->oversample + 4 - offset - 2, N, st->oversample, interp); #endif - + out[out_stride * out_sample++] = PSHR32(sum,15); last_sample += int_advance; samp_frac_num += frac_advance; @@ -543,11 +543,11 @@ static int resampler_basic_interpolate_double(SpeexResamplerState *st, spx_uint3 static void update_filter(SpeexResamplerState *st) { spx_uint32_t old_length; - + old_length = st->filt_len; st->oversample = quality_map[st->quality].oversample; st->filt_len = quality_map[st->quality].base_length; - + if (st->num_rate > st->den_rate) { /* down-sampling */ @@ -570,7 +570,7 @@ static void update_filter(SpeexResamplerState *st) /* up-sampling */ st->cutoff = quality_map[st->quality].upsample_bandwidth; } - + /* Choose the resampling type that requires the least amount of memory */ if (st->den_rate <= st->oversample) { @@ -623,7 +623,7 @@ static void update_filter(SpeexResamplerState *st) st->int_advance = st->num_rate/st->den_rate; st->frac_advance = st->num_rate%st->den_rate; - + /* Here's the place where we update the filter memory to take into account the change in filter length. It's probably the messiest part of the code due to handling of lots of corner cases. */ @@ -661,7 +661,7 @@ static void update_filter(SpeexResamplerState *st) /*if (st->magic_samples[i])*/ { /* Try and remove the magic samples as if nothing had happened */ - + /* FIXME: This is wrong but for now we need it to avoid going over the array bounds */ olen = old_length + 2*st->magic_samples[i]; for (j=old_length-2+st->magic_samples[i];j>=0;j--) @@ -736,18 +736,18 @@ static void update_filter(SpeexResamplerState *st) st->filt_len = 0; st->mem = 0; st->resampler_ptr = 0; - + st->cutoff = 1.f; st->nb_channels = nb_channels; st->in_stride = 1; st->out_stride = 1; - + #ifdef FIXED_POINT st->buffer_size = 160; #else st->buffer_size = 160; #endif - + /* Per channel data */ st->last_sample = (spx_int32_t*)speex_alloc(nb_channels*sizeof(int)); st->magic_samples = (spx_uint32_t*)speex_alloc(nb_channels*sizeof(int)); @@ -762,9 +762,9 @@ static void update_filter(SpeexResamplerState *st) speex_resampler_set_quality(st, quality); speex_resampler_set_rate_frac(st, ratio_num, ratio_den, in_rate, out_rate); - + update_filter(st); - + st->initialised = 1; if (err) *err = RESAMPLER_ERR_SUCCESS; @@ -789,17 +789,17 @@ static int speex_resampler_process_native(SpeexResamplerState *st, spx_uint32_t int out_sample = 0; spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size; spx_uint32_t ilen; - + st->started = 1; - + /* Call the right resampler through the function ptr */ out_sample = st->resampler_ptr(st, channel_index, mem, in_len, out, out_len); - + if (st->last_sample[channel_index] < (spx_int32_t)*in_len) *in_len = st->last_sample[channel_index]; *out_len = out_sample; st->last_sample[channel_index] -= *in_len; - + ilen = *in_len; for(j=0;j<N-1;++j) @@ -812,11 +812,11 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i spx_uint32_t tmp_in_len = st->magic_samples[channel_index]; spx_word16_t *mem = st->mem + channel_index * st->mem_alloc_size; const int N = st->filt_len; - + speex_resampler_process_native(st, channel_index, &tmp_in_len, *out, &out_len); st->magic_samples[channel_index] -= tmp_in_len; - + /* If we couldn't process all "magic" input samples, save the rest for next time */ if (st->magic_samples[channel_index]) { @@ -842,13 +842,13 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i const spx_uint32_t xlen = st->mem_alloc_size - filt_offs; const int istride = st->in_stride; - if (st->magic_samples[channel_index]) + if (st->magic_samples[channel_index]) olen -= speex_resampler_magic(st, channel_index, &out, olen); if (! st->magic_samples[channel_index]) { while (ilen && olen) { spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen; spx_uint32_t ochunk = olen; - + if (in) { for(j=0;j<ichunk;++j) x[j+filt_offs]=in[j*istride]; @@ -892,7 +892,7 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i #endif st->out_stride = 1; - + while (ilen && olen) { spx_word16_t *y = ystack; spx_uint32_t ichunk = (ilen > xlen) ? xlen : ilen; @@ -929,7 +929,7 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i #else out[j*ostride_save] = WORD2INT(ystack[j]); #endif - + ilen -= ichunk; olen -= ochunk; out += (ochunk+omagic) * ostride_save; @@ -963,7 +963,7 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i st->out_stride = ostride_save; return RESAMPLER_ERR_SUCCESS; } - + int speex_resampler_process_interleaved_int(SpeexResamplerState *st, const spx_int16_t *in, spx_uint32_t *in_len, spx_int16_t *out, spx_uint32_t *out_len) { spx_uint32_t i; @@ -1003,7 +1003,7 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i spx_uint32_t i; if (st->in_rate == in_rate && st->out_rate == out_rate && st->num_rate == ratio_num && st->den_rate == ratio_den) return RESAMPLER_ERR_SUCCESS; - + old_den = st->den_rate; st->in_rate = in_rate; st->out_rate = out_rate; @@ -1018,7 +1018,7 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i st->den_rate /= fact; } } - + if (old_den > 0) { for (i=0;i<st->nb_channels;i++) @@ -1029,7 +1029,7 @@ static int speex_resampler_magic(SpeexResamplerState *st, spx_uint32_t channel_i st->samp_frac_num[i] = st->den_rate-1; } } - + if (st->initialised) update_filter(st); return RESAMPLER_ERR_SUCCESS; diff --git a/codecs/speex/resample_sse.h b/codecs/speex/resample_sse.h index 4bd35a2d0..d85898067 100644 --- a/codecs/speex/resample_sse.h +++ b/codecs/speex/resample_sse.h @@ -9,18 +9,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR diff --git a/codecs/speex/speex_resampler.h b/codecs/speex/speex_resampler.h index 5dead0e79..d02022d2e 100644 --- a/codecs/speex/speex_resampler.h +++ b/codecs/speex/speex_resampler.h @@ -1,8 +1,8 @@ /* Copyright (C) 2007 Jean-Marc Valin - + File: speex_resampler.h Resampling code - + The design goals of this code are: - Very fast algorithm - Low memory requirement @@ -45,7 +45,7 @@ /********* WARNING: MENTAL SANITY ENDS HERE *************/ -/* If the resampler is defined outside of Speex, we change the symbol names so that +/* If the resampler is defined outside of Speex, we change the symbol names so that there won't be any clash if linking with Speex later on. */ #define RANDOM_PREFIX ast @@ -55,7 +55,7 @@ #define CAT_PREFIX2(a,b) a ## b #define CAT_PREFIX(a,b) CAT_PREFIX2(a, b) - + #define speex_resampler_init CAT_PREFIX(RANDOM_PREFIX,_resampler_init) #define speex_resampler_init_frac CAT_PREFIX(RANDOM_PREFIX,_resampler_init_frac) #define speex_resampler_destroy CAT_PREFIX(RANDOM_PREFIX,_resampler_destroy) @@ -83,7 +83,7 @@ #define spx_int32_t int #define spx_uint16_t unsigned short #define spx_uint32_t unsigned int - + #else /* OUTSIDE_SPEEX */ #include "speex/speex_types.h" @@ -106,7 +106,7 @@ enum { RESAMPLER_ERR_BAD_STATE = 2, RESAMPLER_ERR_INVALID_ARG = 3, RESAMPLER_ERR_PTR_OVERLAP = 4, - + RESAMPLER_ERR_MAX_ERROR }; @@ -123,14 +123,14 @@ typedef struct SpeexResamplerState_ SpeexResamplerState; * \return Newly created resampler state * \retval NULL Error: not enough memory */ -SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, - spx_uint32_t in_rate, - spx_uint32_t out_rate, +SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, + spx_uint32_t in_rate, + spx_uint32_t out_rate, int quality, int *err); -/** Create a new resampler with fractional input/output rates. The sampling - * rate ratio is an arbitrary rational number with both the numerator and +/** Create a new resampler with fractional input/output rates. The sampling + * rate ratio is an arbitrary rational number with both the numerator and * denominator being 32-bit integers. * @param nb_channels Number of channels to be processed * @param ratio_num Numerator of the sampling rate ratio @@ -143,11 +143,11 @@ SpeexResamplerState *speex_resampler_init(spx_uint32_t nb_channels, * @return Newly created resampler state * @retval NULL Error: not enough memory */ -SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, - spx_uint32_t ratio_num, - spx_uint32_t ratio_den, - spx_uint32_t in_rate, - spx_uint32_t out_rate, +SpeexResamplerState *speex_resampler_init_frac(spx_uint32_t nb_channels, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, + spx_uint32_t out_rate, int quality, int *err); @@ -158,24 +158,24 @@ void speex_resampler_destroy(SpeexResamplerState *st); /** Resample a float array. The input and output buffers must *not* overlap. * @param st Resampler state - * @param channel_index Index of the channel to process for the multi-channel + * @param channel_index Index of the channel to process for the multi-channel * base (0 otherwise) * @param in Input buffer - * @param in_len Number of input samples in the input buffer. Returns the + * @param in_len Number of input samples in the input buffer. Returns the * number of samples processed * @param out Output buffer * @param out_len Size of the output buffer. Returns the number of samples written */ -int speex_resampler_process_float(SpeexResamplerState *st, - spx_uint32_t channel_index, - const float *in, - spx_uint32_t *in_len, - float *out, +int speex_resampler_process_float(SpeexResamplerState *st, + spx_uint32_t channel_index, + const float *in, + spx_uint32_t *in_len, + float *out, spx_uint32_t *out_len); /** Resample an int array. The input and output buffers must *not* overlap. * @param st Resampler state - * @param channel_index Index of the channel to process for the multi-channel + * @param channel_index Index of the channel to process for the multi-channel * base (0 otherwise) * @param in Input buffer * @param in_len Number of input samples in the input buffer. Returns the number @@ -183,11 +183,11 @@ int speex_resampler_process_float(SpeexResamplerState *st, * @param out Output buffer * @param out_len Size of the output buffer. Returns the number of samples written */ -int speex_resampler_process_int(SpeexResamplerState *st, - spx_uint32_t channel_index, - const spx_int16_t *in, - spx_uint32_t *in_len, - spx_int16_t *out, +int speex_resampler_process_int(SpeexResamplerState *st, + spx_uint32_t channel_index, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, spx_uint32_t *out_len); /** Resample an interleaved float array. The input and output buffers must *not* overlap. @@ -199,10 +199,10 @@ int speex_resampler_process_int(SpeexResamplerState *st, * @param out_len Size of the output buffer. Returns the number of samples written. * This is all per-channel. */ -int speex_resampler_process_interleaved_float(SpeexResamplerState *st, - const float *in, - spx_uint32_t *in_len, - float *out, +int speex_resampler_process_interleaved_float(SpeexResamplerState *st, + const float *in, + spx_uint32_t *in_len, + float *out, spx_uint32_t *out_len); /** Resample an interleaved int array. The input and output buffers must *not* overlap. @@ -214,10 +214,10 @@ int speex_resampler_process_interleaved_float(SpeexResamplerState *st, * @param out_len Size of the output buffer. Returns the number of samples written. * This is all per-channel. */ -int speex_resampler_process_interleaved_int(SpeexResamplerState *st, - const spx_int16_t *in, - spx_uint32_t *in_len, - spx_int16_t *out, +int speex_resampler_process_interleaved_int(SpeexResamplerState *st, + const spx_int16_t *in, + spx_uint32_t *in_len, + spx_int16_t *out, spx_uint32_t *out_len); /** Set (change) the input/output sampling rates (integer value). @@ -225,8 +225,8 @@ int speex_resampler_process_interleaved_int(SpeexResamplerState *st, * @param in_rate Input sampling rate (integer number of Hz). * @param out_rate Output sampling rate (integer number of Hz). */ -int speex_resampler_set_rate(SpeexResamplerState *st, - spx_uint32_t in_rate, +int speex_resampler_set_rate(SpeexResamplerState *st, + spx_uint32_t in_rate, spx_uint32_t out_rate); /** Get the current input/output sampling rates (integer value). @@ -234,11 +234,11 @@ int speex_resampler_set_rate(SpeexResamplerState *st, * @param in_rate Input sampling rate (integer number of Hz) copied. * @param out_rate Output sampling rate (integer number of Hz) copied. */ -void speex_resampler_get_rate(SpeexResamplerState *st, - spx_uint32_t *in_rate, +void speex_resampler_get_rate(SpeexResamplerState *st, + spx_uint32_t *in_rate, spx_uint32_t *out_rate); -/** Set (change) the input/output sampling rates and resampling ratio +/** Set (change) the input/output sampling rates and resampling ratio * (fractional values in Hz supported). * @param st Resampler state * @param ratio_num Numerator of the sampling rate ratio @@ -246,10 +246,10 @@ void speex_resampler_get_rate(SpeexResamplerState *st, * @param in_rate Input sampling rate rounded to the nearest integer (in Hz). * @param out_rate Output sampling rate rounded to the nearest integer (in Hz). */ -int speex_resampler_set_rate_frac(SpeexResamplerState *st, - spx_uint32_t ratio_num, - spx_uint32_t ratio_den, - spx_uint32_t in_rate, +int speex_resampler_set_rate_frac(SpeexResamplerState *st, + spx_uint32_t ratio_num, + spx_uint32_t ratio_den, + spx_uint32_t in_rate, spx_uint32_t out_rate); /** Get the current resampling ratio. This will be reduced to the least @@ -258,52 +258,52 @@ int speex_resampler_set_rate_frac(SpeexResamplerState *st, * @param ratio_num Numerator of the sampling rate ratio copied * @param ratio_den Denominator of the sampling rate ratio copied */ -void speex_resampler_get_ratio(SpeexResamplerState *st, - spx_uint32_t *ratio_num, +void speex_resampler_get_ratio(SpeexResamplerState *st, + spx_uint32_t *ratio_num, spx_uint32_t *ratio_den); /** Set (change) the conversion quality. * @param st Resampler state - * @param quality Resampling quality between 0 and 10, where 0 has poor + * @param quality Resampling quality between 0 and 10, where 0 has poor * quality and 10 has very high quality. */ -int speex_resampler_set_quality(SpeexResamplerState *st, +int speex_resampler_set_quality(SpeexResamplerState *st, int quality); /** Get the conversion quality. * @param st Resampler state - * @param quality Resampling quality between 0 and 10, where 0 has poor + * @param quality Resampling quality between 0 and 10, where 0 has poor * quality and 10 has very high quality. */ -void speex_resampler_get_quality(SpeexResamplerState *st, +void speex_resampler_get_quality(SpeexResamplerState *st, int *quality); /** Set (change) the input stride. * @param st Resampler state * @param stride Input stride */ -void speex_resampler_set_input_stride(SpeexResamplerState *st, +void speex_resampler_set_input_stride(SpeexResamplerState *st, spx_uint32_t stride); /** Get the input stride. * @param st Resampler state * @param stride Input stride copied */ -void speex_resampler_get_input_stride(SpeexResamplerState *st, +void speex_resampler_get_input_stride(SpeexResamplerState *st, spx_uint32_t *stride); /** Set (change) the output stride. * @param st Resampler state * @param stride Output stride */ -void speex_resampler_set_output_stride(SpeexResamplerState *st, +void speex_resampler_set_output_stride(SpeexResamplerState *st, spx_uint32_t stride); /** Get the output stride. * @param st Resampler state copied * @param stride Output stride */ -void speex_resampler_get_output_stride(SpeexResamplerState *st, +void speex_resampler_get_output_stride(SpeexResamplerState *st, spx_uint32_t *stride); /** Get the latency in input samples introduced by the resampler. @@ -316,8 +316,8 @@ int speex_resampler_get_input_latency(SpeexResamplerState *st); */ int speex_resampler_get_output_latency(SpeexResamplerState *st); -/** Make sure that the first samples to go out of the resamplers don't have - * leading zeros. This is only useful before starting to use a newly created +/** Make sure that the first samples to go out of the resamplers don't have + * leading zeros. This is only useful before starting to use a newly created * resampler. It is recommended to use that when resampling an audio file, as * it will generate a file with the same length. For real-time processing, * it is probably easier not to use this call (so that the output duration diff --git a/codecs/speex/stack_alloc.h b/codecs/speex/stack_alloc.h index 5264e666b..6c56334f8 100644 --- a/codecs/speex/stack_alloc.h +++ b/codecs/speex/stack_alloc.h @@ -7,18 +7,18 @@ Redistribution and use in source and binary forms, with or without modification, are permitted provided that the following conditions are met: - + - Redistributions of source code must retain the above copyright notice, this list of conditions and the following disclaimer. - + - Redistributions in binary form must reproduce the above copyright notice, this list of conditions and the following disclaimer in the documentation and/or other materials provided with the distribution. - + - Neither the name of the Xiph.org Foundation nor the names of its contributors may be used to endorse or promote products derived from this software without specific prior written permission. - + THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR @@ -101,7 +101,7 @@ #endif #if defined(VAR_ARRAYS) -#define VARDECL(var) +#define VARDECL(var) #define ALLOC(var, size, type) type var[size] #elif defined(USE_ALLOCA) #define VARDECL(var) var |