From f919f9dcbb54a08e6e26d6c229ce92fb784fa1b2 Mon Sep 17 00:00:00 2001
From: Tom Rondeau <trondeau@vt.edu>
Date: Fri, 13 Apr 2012 18:36:53 -0400
Subject: Removed whitespace and added dtools/bin/remove-whitespace as a tool
to do this in the future.
The sed script was provided by Moritz Fischer.
---
gr-vocoder/lib/codec2/CMakeLists.txt | 2 +-
gr-vocoder/lib/codec2/_kiss_fft_guts.h | 8 +-
gr-vocoder/lib/codec2/c2dec.c | 2 +-
gr-vocoder/lib/codec2/c2demo.c | 4 +-
gr-vocoder/lib/codec2/c2enc.c | 2 +-
gr-vocoder/lib/codec2/c2sim.c | 60 ++++-----
gr-vocoder/lib/codec2/codebook/lsp8910.txt | 128 +++++++++---------
gr-vocoder/lib/codec2/codec2.c | 74 +++++------
gr-vocoder/lib/codec2/codec2.h | 2 +-
gr-vocoder/lib/codec2/codec2_internal.h | 4 +-
gr-vocoder/lib/codec2/comp.h | 8 +-
gr-vocoder/lib/codec2/defines.h | 24 ++--
gr-vocoder/lib/codec2/dump.c | 54 ++++----
gr-vocoder/lib/codec2/dump.h | 8 +-
gr-vocoder/lib/codec2/fft.c | 32 ++---
gr-vocoder/lib/codec2/fft.h | 2 +-
gr-vocoder/lib/codec2/fq20.sh | 2 +-
gr-vocoder/lib/codec2/generate_codebook.c | 10 +-
gr-vocoder/lib/codec2/globals.c | 12 +-
gr-vocoder/lib/codec2/globals.h | 10 +-
gr-vocoder/lib/codec2/interp.c | 86 ++++++-------
gr-vocoder/lib/codec2/interp.h | 2 +-
gr-vocoder/lib/codec2/kiss_fft.c | 20 +--
gr-vocoder/lib/codec2/kiss_fft.h | 28 ++--
gr-vocoder/lib/codec2/lpc.c | 136 ++++++++++----------
gr-vocoder/lib/codec2/lpc.h | 14 +-
gr-vocoder/lib/codec2/lsp.c | 4 +-
gr-vocoder/lib/codec2/nlp.c | 62 ++++-----
gr-vocoder/lib/codec2/nlp.h | 16 +--
gr-vocoder/lib/codec2/pack.c | 4 +-
gr-vocoder/lib/codec2/phase.c | 32 ++---
gr-vocoder/lib/codec2/phase.h | 14 +-
gr-vocoder/lib/codec2/postfilter.c | 14 +-
gr-vocoder/lib/codec2/postfilter.h | 8 +-
gr-vocoder/lib/codec2/quantise.c | 194 ++++++++++++++--------------
gr-vocoder/lib/codec2/quantise.h | 24 ++--
gr-vocoder/lib/codec2/sine.c | 200 ++++++++++++++---------------
gr-vocoder/lib/codec2/sine.h | 10 +-
38 files changed, 658 insertions(+), 658 deletions(-)
(limited to 'gr-vocoder/lib/codec2')
diff --git a/gr-vocoder/lib/codec2/CMakeLists.txt b/gr-vocoder/lib/codec2/CMakeLists.txt
index c9839bd965..95e9d256be 100644
--- a/gr-vocoder/lib/codec2/CMakeLists.txt
+++ b/gr-vocoder/lib/codec2/CMakeLists.txt
@@ -80,7 +80,7 @@ set(CODEBOOKSDVQ
${CMAKE_CURRENT_SOURCE_DIR}/codebook/dlsp2.txt
${CMAKE_CURRENT_SOURCE_DIR}/codebook/dlsp3.txt
${CMAKE_CURRENT_SOURCE_DIR}/codebook/dlsp4.txt
- ${CMAKE_CURRENT_SOURCE_DIR}/codebook/dlsp5.txt
+ ${CMAKE_CURRENT_SOURCE_DIR}/codebook/dlsp5.txt
)
add_custom_command(
diff --git a/gr-vocoder/lib/codec2/_kiss_fft_guts.h b/gr-vocoder/lib/codec2/_kiss_fft_guts.h
index ba66144403..f008a7b500 100644
--- a/gr-vocoder/lib/codec2/_kiss_fft_guts.h
+++ b/gr-vocoder/lib/codec2/_kiss_fft_guts.h
@@ -20,7 +20,7 @@ THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" AND
#include <limits.h>
#define MAXFACTORS 32
-/* e.g. an fft of length 128 has 4 factors
+/* e.g. an fft of length 128 has 4 factors
as far as kissfft is concerned
4*4*4*2
*/
@@ -48,7 +48,7 @@ struct kiss_fft_state{
#define SAMP_MAX 2147483647
#else
# define FRACBITS 15
-# define SAMPPROD int32_t
+# define SAMPPROD int32_t
#define SAMP_MAX 32767
#endif
@@ -152,12 +152,12 @@ struct kiss_fft_state{
#ifdef KISS_FFT_USE_ALLOCA
// define this to allow use of alloca instead of malloc for temporary buffers
-// Temporary buffers are used in two case:
+// Temporary buffers are used in two case:
// 1. FFT sizes that have "bad" factors. i.e. not 2,3 and 5
// 2. "in-place" FFTs. Notice the quotes, since kissfft does not really do an in-place transform.
#include <alloca.h>
#define KISS_FFT_TMP_ALLOC(nbytes) alloca(nbytes)
-#define KISS_FFT_TMP_FREE(ptr)
+#define KISS_FFT_TMP_FREE(ptr)
#else
#define KISS_FFT_TMP_ALLOC(nbytes) KISS_FFT_MALLOC(nbytes)
#define KISS_FFT_TMP_FREE(ptr) KISS_FFT_FREE(ptr)
diff --git a/gr-vocoder/lib/codec2/c2dec.c b/gr-vocoder/lib/codec2/c2dec.c
index b866d04d60..3123e96fbc 100644
--- a/gr-vocoder/lib/codec2/c2dec.c
+++ b/gr-vocoder/lib/codec2/c2dec.c
@@ -70,7 +70,7 @@ int main(int argc, char *argv[])
//buffering to occur
if (fout == stdout) fflush(stdout);
if (fin == stdin) fflush(stdin);
-
+
}
codec2_destroy(codec2);
diff --git a/gr-vocoder/lib/codec2/c2demo.c b/gr-vocoder/lib/codec2/c2demo.c
index efa8d6449a..b9e17a78eb 100644
--- a/gr-vocoder/lib/codec2/c2demo.c
+++ b/gr-vocoder/lib/codec2/c2demo.c
@@ -52,7 +52,7 @@ int main(int argc, char *argv[])
printf("usage: %s InputRawSpeechFile OutputRawSpeechFile\n", argv[0]);
exit(1);
}
-
+
if ( (fin = fopen(argv[1],"rb")) == NULL ) {
fprintf(stderr, "Error opening input speech file: %s: %s.\n",
argv[1], strerror(errno));
@@ -70,7 +70,7 @@ int main(int argc, char *argv[])
codec2 = codec2_create();
- while(fread(buf, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fin) ==
+ while(fread(buf, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fin) ==
CODEC2_SAMPLES_PER_FRAME) {
codec2_encode(codec2, bits, buf);
codec2_decode(codec2, buf, bits);
diff --git a/gr-vocoder/lib/codec2/c2enc.c b/gr-vocoder/lib/codec2/c2enc.c
index 4d1d019dfc..0e5b26cfdc 100644
--- a/gr-vocoder/lib/codec2/c2enc.c
+++ b/gr-vocoder/lib/codec2/c2enc.c
@@ -47,7 +47,7 @@ int main(int argc, char *argv[])
printf("usage: %s InputRawspeechFile OutputBitFile\n", argv[0]);
exit(1);
}
-
+
if (strcmp(argv[1], "-") == 0) fin = stdin;
else if ( (fin = fopen(argv[1],"rb")) == NULL ) {
fprintf(stderr, "Error opening input bit file: %s: %s.\n",
diff --git a/gr-vocoder/lib/codec2/c2sim.c b/gr-vocoder/lib/codec2/c2sim.c
index bb49c78995..e335078b64 100644
--- a/gr-vocoder/lib/codec2/c2sim.c
+++ b/gr-vocoder/lib/codec2/c2sim.c
@@ -4,8 +4,8 @@
AUTHOR......: David Rowe
DATE CREATED: 20/8/2010
- Codec2 simulation. Combines encoder and decoder and allows switching in
- out various algorithms and quantisation steps.
+ Codec2 simulation. Combines encoder and decoder and allows switching in
+ out various algorithms and quantisation steps.
\*---------------------------------------------------------------------------*/
@@ -45,13 +45,13 @@
#include "interp.h"
/*---------------------------------------------------------------------------*\
-
- switch_present()
-
- Searches the command line arguments for a "switch". If the switch is
- found, returns the command line argument where it ws found, else returns
- NULL.
-
+
+ switch_present()
+
+ Searches the command line arguments for a "switch". If the switch is
+ found, returns the command line argument where it ws found, else returns
+ NULL.
+
\*---------------------------------------------------------------------------*/
int switch_present(sw,argc,argv)
@@ -71,9 +71,9 @@ register char *argv[]; /* array of command line arguments in string form */
void synth_one_frame(short buf[], MODEL *model, float Sn_[], float Pn[]);
/*---------------------------------------------------------------------------*\
-
- MAIN
-
+
+ MAIN
+
\*---------------------------------------------------------------------------*/
int main(int argc, char *argv[])
@@ -103,10 +103,10 @@ int main(int argc, char *argv[])
int lsp, lspd, lspdvq, lsp_quantiser;
float ak[LPC_MAX];
COMP Sw_[FFT_ENC];
- COMP Ew[FFT_ENC];
-
+ COMP Ew[FFT_ENC];
+
int dump;
-
+
int phase0;
float ex_phase[MAX_AMP+1];
@@ -129,7 +129,7 @@ int main(int argc, char *argv[])
float AresdB_prev[MAX_AMP];
for(i=0; i<MAX_AMP; i++)
- AresdB_prev[i] = 0.0;
+ AresdB_prev[i] = 0.0;
for(i=0; i<M; i++)
Sn[i] = 1.0;
@@ -201,12 +201,12 @@ int main(int argc, char *argv[])
if ((order < 4) || (order > 20)) {
fprintf(stderr, "Error in lpc order: %d\n", order);
exit(1);
- }
+ }
}
dump = switch_present("--dump",argc,argv);
#ifdef DUMP
- if (dump)
+ if (dump)
dump_on(argv[dump+1]);
#endif
@@ -264,7 +264,7 @@ int main(int argc, char *argv[])
//Sn[i+M-N] = hpf((float)buf[i], hpf_states);
Sn[i+M-N] = (float)buf[i];
}
-
+
/* Estimate pitch */
nlp(nlp_states,Sn,N,M,P_MIN,P_MAX,&pitch,Sw,&prev_Wo);
@@ -272,10 +272,10 @@ int main(int argc, char *argv[])
/* estimate model parameters */
- dft_speech(Sw, Sn, w);
+ dft_speech(Sw, Sn, w);
two_stage_pitch_refinement(&model, Sw);
estimate_amplitudes(&model, Sw, W);
-#ifdef DUMP
+#ifdef DUMP
dump_Sn(Sn); dump_Sw(Sw); dump_model(&model);
#endif
@@ -284,7 +284,7 @@ int main(int argc, char *argv[])
if (phase0) {
float Wn[M]; /* windowed speech samples */
float Rk[LPC_MAX+1]; /* autocorrelation coeffs */
-
+
#ifdef DUMP
dump_phase(&model.phi[0], model.L);
#endif
@@ -299,7 +299,7 @@ int main(int argc, char *argv[])
#ifdef DUMP
dump_ak(ak, LPC_ORD);
#endif
-
+
/* determine voicing */
snr = est_voicing_mbe(&model, Sw, W, Sw_, Ew, prev_Wo);
@@ -313,7 +313,7 @@ int main(int argc, char *argv[])
for(i=0; i<MAX_AMP; i++)
model.phi[i] = 0;
-
+
if (hand_voicing) {
fscanf(fvoicing,"%d\n",&model.voiced);
}
@@ -350,7 +350,7 @@ int main(int argc, char *argv[])
e = decode_energy(encode_energy(e));
model.Wo = decode_Wo(encode_Wo(model.Wo));
- aks_to_M2(ak, order, &model, e, &snr, 1);
+ aks_to_M2(ak, order, &model, e, &snr, 1);
apply_lpc_correction(&model);
sum_snr += snr;
#ifdef DUMP
@@ -371,7 +371,7 @@ int main(int argc, char *argv[])
/* option decimation to 20ms rate, which enables interpolation
routine to synthesise in between frame */
-
+
if (decimate) {
if (!phase0) {
printf("needs --phase0 to resample phase for interpolated Wo\n");
@@ -395,17 +395,17 @@ int main(int argc, char *argv[])
prev_lsps, prev_e, lsps, e, ak_interp);
apply_lpc_correction(&interp_model);
#endif
-
+
if (phase0)
phase_synth_zero_order(&interp_model, ak_interp, ex_phase,
- order);
+ order);
if (postfilt)
postfilter(&interp_model, &bg_est);
synth_one_frame(buf, &interp_model, Sn_, Pn);
if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
if (phase0)
- phase_synth_zero_order(&model, ak, ex_phase, order);
+ phase_synth_zero_order(&model, ak, ex_phase, order);
if (postfilt)
postfilter(&model, &bg_est);
synth_one_frame(buf, &model, Sn_, Pn);
@@ -422,7 +422,7 @@ int main(int argc, char *argv[])
}
else {
if (phase0)
- phase_synth_zero_order(&model, ak, ex_phase, order);
+ phase_synth_zero_order(&model, ak, ex_phase, order);
if (postfilt)
postfilter(&model, &bg_est);
synth_one_frame(buf, &model, Sn_, Pn);
diff --git a/gr-vocoder/lib/codec2/codebook/lsp8910.txt b/gr-vocoder/lib/codec2/codebook/lsp8910.txt
index 93cfdd81dd..759136bfec 100644
--- a/gr-vocoder/lib/codec2/codebook/lsp8910.txt
+++ b/gr-vocoder/lib/codec2/codebook/lsp8910.txt
@@ -1,65 +1,65 @@
3 64
-2.048073 2.534502 2.645915
-2.019670 2.269744 2.605462
-1.961101 2.329646 2.562857
-1.968573 2.532712 2.616918
-2.183480 2.514381 2.629582
-2.259379 2.516615 2.620410
-2.172791 2.462460 2.567064
-2.097666 2.303933 2.421685
-2.052990 2.353242 2.546992
-2.043642 2.232362 2.499262
-2.106151 2.393131 2.488401
-2.099167 2.437862 2.558655
-2.013877 2.422875 2.530071
-2.033848 2.483776 2.584598
-2.114474 2.516856 2.602372
-2.229214 2.584056 2.678855
-2.131151 2.584299 2.674845
-1.472721 2.477091 2.630241
-2.010907 2.598415 2.682989
-2.353653 2.524066 2.619773
-2.419897 2.623938 2.699605
-2.319080 2.602148 2.689044
-1.860342 2.503881 2.616576
-1.910517 2.386693 2.610126
-1.748689 2.371809 2.496542
-1.618495 2.403425 2.554956
-1.844073 2.437026 2.533443
-1.924810 2.388543 2.502698
-1.937227 2.258363 2.501697
-1.687554 2.209123 2.545239
-1.851950 2.278628 2.565632
-1.868154 2.330150 2.444883
-1.874180 2.213118 2.351940
-1.757311 2.030626 2.433836
-1.650306 2.152371 2.243421
-1.612794 1.884686 2.339313
-1.745431 2.278895 2.389449
-1.590923 2.304155 2.408510
-1.475982 2.275548 2.509897
-1.508695 2.045463 2.455520
-1.872054 2.061777 2.246202
-1.983947 2.159155 2.445535
-1.745180 2.483765 2.593698
-1.900116 2.079600 2.407479
-1.841672 2.167042 2.486827
-1.932912 2.148464 2.569850
-2.134174 2.363673 2.584252
-2.106094 2.450645 2.638417
-1.954135 2.460313 2.666512
-1.907634 2.573801 2.674025
-1.625579 2.539569 2.656363
-1.785866 2.572616 2.676082
-1.798447 2.376454 2.624298
-2.020033 2.397244 2.619868
-1.946581 2.468791 2.564185
-2.008920 2.342400 2.469132
-1.983846 2.271044 2.395408
-1.988039 2.154150 2.317920
-2.077197 2.216622 2.389101
-2.117255 2.283907 2.512242
-2.177233 2.334622 2.458268
-2.214655 2.425510 2.620013
-2.199931 2.390272 2.520731
-2.271755 2.448682 2.552649
+2.048073 2.534502 2.645915
+2.019670 2.269744 2.605462
+1.961101 2.329646 2.562857
+1.968573 2.532712 2.616918
+2.183480 2.514381 2.629582
+2.259379 2.516615 2.620410
+2.172791 2.462460 2.567064
+2.097666 2.303933 2.421685
+2.052990 2.353242 2.546992
+2.043642 2.232362 2.499262
+2.106151 2.393131 2.488401
+2.099167 2.437862 2.558655
+2.013877 2.422875 2.530071
+2.033848 2.483776 2.584598
+2.114474 2.516856 2.602372
+2.229214 2.584056 2.678855
+2.131151 2.584299 2.674845
+1.472721 2.477091 2.630241
+2.010907 2.598415 2.682989
+2.353653 2.524066 2.619773
+2.419897 2.623938 2.699605
+2.319080 2.602148 2.689044
+1.860342 2.503881 2.616576
+1.910517 2.386693 2.610126
+1.748689 2.371809 2.496542
+1.618495 2.403425 2.554956
+1.844073 2.437026 2.533443
+1.924810 2.388543 2.502698
+1.937227 2.258363 2.501697
+1.687554 2.209123 2.545239
+1.851950 2.278628 2.565632
+1.868154 2.330150 2.444883
+1.874180 2.213118 2.351940
+1.757311 2.030626 2.433836
+1.650306 2.152371 2.243421
+1.612794 1.884686 2.339313
+1.745431 2.278895 2.389449
+1.590923 2.304155 2.408510
+1.475982 2.275548 2.509897
+1.508695 2.045463 2.455520
+1.872054 2.061777 2.246202
+1.983947 2.159155 2.445535
+1.745180 2.483765 2.593698
+1.900116 2.079600 2.407479
+1.841672 2.167042 2.486827
+1.932912 2.148464 2.569850
+2.134174 2.363673 2.584252
+2.106094 2.450645 2.638417
+1.954135 2.460313 2.666512
+1.907634 2.573801 2.674025
+1.625579 2.539569 2.656363
+1.785866 2.572616 2.676082
+1.798447 2.376454 2.624298
+2.020033 2.397244 2.619868
+1.946581 2.468791 2.564185
+2.008920 2.342400 2.469132
+1.983846 2.271044 2.395408
+1.988039 2.154150 2.317920
+2.077197 2.216622 2.389101
+2.117255 2.283907 2.512242
+2.177233 2.334622 2.458268
+2.214655 2.425510 2.620013
+2.199931 2.390272 2.520731
+2.271755 2.448682 2.552649
diff --git a/gr-vocoder/lib/codec2/codec2.c b/gr-vocoder/lib/codec2/codec2.c
index 92708ee322..93ea9208c1 100644
--- a/gr-vocoder/lib/codec2/codec2.c
+++ b/gr-vocoder/lib/codec2/codec2.c
@@ -4,7 +4,7 @@
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
- Codec2 fully quantised encoder and decoder functions. If you want use
+ Codec2 fully quantised encoder and decoder functions. If you want use
codec2, the codec2_xxx functions are for you.
\*---------------------------------------------------------------------------*/
@@ -45,16 +45,16 @@
#include "codec2_internal.h"
/*---------------------------------------------------------------------------*\
-
+
FUNCTIONS
\*---------------------------------------------------------------------------*/
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: codec2_create
- AUTHOR......: David Rowe
- DATE CREATED: 21/8/2010
+
+ FUNCTION....: codec2_create
+ AUTHOR......: David Rowe
+ DATE CREATED: 21/8/2010
Create and initialise an instance of the codec. Returns a pointer
to the codec states or NULL on failure. One set of states is
@@ -106,10 +106,10 @@ void *codec2_create()
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: codec2_create
- AUTHOR......: David Rowe
- DATE CREATED: 21/8/2010
+
+ FUNCTION....: codec2_create
+ AUTHOR......: David Rowe
+ DATE CREATED: 21/8/2010
Destroy an instance of the codec.
@@ -118,7 +118,7 @@ void *codec2_create()
void codec2_destroy(void *codec2_state)
{
CODEC2 *c2;
-
+
assert(codec2_state != NULL);
c2 = (CODEC2*)codec2_state;
nlp_destroy(c2->nlp);
@@ -126,12 +126,12 @@ void codec2_destroy(void *codec2_state)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: codec2_encode
- AUTHOR......: David Rowe
- DATE CREATED: 21/8/2010
- Encodes 160 speech samples (20ms of speech) into 51 bits.
+ FUNCTION....: codec2_encode
+ AUTHOR......: David Rowe
+ DATE CREATED: 21/8/2010
+
+ Encodes 160 speech samples (20ms of speech) into 51 bits.
The codec2 algorithm actually operates internally on 10ms (80
sample) frames, so we run the encoding algorithm twice. On the
@@ -148,7 +148,7 @@ void codec2_destroy(void *codec2_state)
Wo (fundamental frequnecy) 7
Voicing (10ms update) 2
TOTAL 51
-
+
\*---------------------------------------------------------------------------*/
void codec2_encode(void *codec2_state, unsigned char * bits, short speech[])
@@ -174,13 +174,13 @@ void codec2_encode(void *codec2_state, unsigned char * bits, short speech[])
analyse_one_frame(c2, &model, &speech[N]);
voiced2 = model.voiced;
-
+
Wo_index = encode_Wo(model.Wo);
- encode_amplitudes(lsp_indexes,
+ encode_amplitudes(lsp_indexes,
&energy_index,
- &model,
- c2->Sn,
- c2->w);
+ &model,
+ c2->Sn,
+ c2->w);
memset(bits, '\0', ((CODEC2_BITS_PER_FRAME + 7) / 8));
pack(bits, &nbit, Wo_index, WO_BITS);
for(i=0; i<LPC_ORD; i++) {
@@ -189,15 +189,15 @@ void codec2_encode(void *codec2_state, unsigned char * bits, short speech[])
pack(bits, &nbit, energy_index, E_BITS);
pack(bits, &nbit, voiced1, 1);
pack(bits, &nbit, voiced2, 1);
-
+
assert(nbit == CODEC2_BITS_PER_FRAME);
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: codec2_decode
- AUTHOR......: David Rowe
- DATE CREATED: 21/8/2010
+
+ FUNCTION....: codec2_decode
+ AUTHOR......: David Rowe
+ DATE CREATED: 21/8/2010
Decodes frames of 51 bits into 160 samples (20ms) of speech.
@@ -239,7 +239,7 @@ void codec2_decode(void *codec2_state, short speech[],
model.Wo = decode_Wo(Wo_index);
model.L = PI/model.Wo;
memset(&model.A, 0, (model.L+1)*sizeof(model.A[0]));
- decode_amplitudes(&model,
+ decode_amplitudes(&model,
ak,
lsp_indexes,
energy_index,
@@ -270,10 +270,10 @@ void codec2_decode(void *codec2_state, short speech[],
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: synthesise_one_frame()
- AUTHOR......: David Rowe
- DATE CREATED: 23/8/2010
+
+ FUNCTION....: synthesise_one_frame()
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/8/2010
Synthesise 80 speech samples (10ms) from model parameters.
@@ -299,14 +299,14 @@ void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model, float ak[])
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: analyse_one_frame()
- AUTHOR......: David Rowe
- DATE CREATED: 23/8/2010
+
+ FUNCTION....: analyse_one_frame()
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/8/2010
Extract sinusoidal model parameters from 80 speech samples (10ms of
speech).
-
+
\*---------------------------------------------------------------------------*/
void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[])
diff --git a/gr-vocoder/lib/codec2/codec2.h b/gr-vocoder/lib/codec2/codec2.h
index 946dedca5e..88e68814fa 100644
--- a/gr-vocoder/lib/codec2/codec2.h
+++ b/gr-vocoder/lib/codec2/codec2.h
@@ -4,7 +4,7 @@
AUTHOR......: David Rowe
DATE CREATED: 21/8/2010
- Codec2 fully quantised encoder and decoder functions. If you want use
+ Codec2 fully quantised encoder and decoder functions. If you want use
codec2, these are the functions you need to call.
\*---------------------------------------------------------------------------*/
diff --git a/gr-vocoder/lib/codec2/codec2_internal.h b/gr-vocoder/lib/codec2/codec2_internal.h
index 3943ac29dd..3aa5300b10 100644
--- a/gr-vocoder/lib/codec2/codec2_internal.h
+++ b/gr-vocoder/lib/codec2/codec2_internal.h
@@ -30,7 +30,7 @@
#define __CODEC2_INTERNAL__
/*---------------------------------------------------------------------------*\
-
+
STATES
\*---------------------------------------------------------------------------*/
@@ -52,7 +52,7 @@ typedef struct {
} CODEC2;
/*---------------------------------------------------------------------------*\
-
+
FUNCTION HEADERS
\*---------------------------------------------------------------------------*/
diff --git a/gr-vocoder/lib/codec2/comp.h b/gr-vocoder/lib/codec2/comp.h
index cedcab37f2..ffc20c163d 100644
--- a/gr-vocoder/lib/codec2/comp.h
+++ b/gr-vocoder/lib/codec2/comp.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: comp.h
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 24/08/09
-
+
Complex number definition.
-
+
\*---------------------------------------------------------------------------*/
/*
diff --git a/gr-vocoder/lib/codec2/defines.h b/gr-vocoder/lib/codec2/defines.h
index 2dcd527d37..75064fae14 100644
--- a/gr-vocoder/lib/codec2/defines.h
+++ b/gr-vocoder/lib/codec2/defines.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
- FILE........: defines.h
- AUTHOR......: David Rowe
- DATE CREATED: 23/4/93
-
- Defines and structures used throughout the codec.
-
+ FILE........: defines.h
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/4/93
+
+ Defines and structures used throughout the codec.
+
\*---------------------------------------------------------------------------*/
/*
@@ -29,9 +29,9 @@
#define __DEFINES__
/*---------------------------------------------------------------------------*\
-
- DEFINES
-
+
+ DEFINES
+
\*---------------------------------------------------------------------------*/
/* General defines */
@@ -58,9 +58,9 @@
#define P_MAX 160 /* maximum pitch */
/*---------------------------------------------------------------------------*\
-
- TYPEDEFS
-
+
+ TYPEDEFS
+
\*---------------------------------------------------------------------------*/
/* Structure to hold model parameters for one frame */
diff --git a/gr-vocoder/lib/codec2/dump.c b/gr-vocoder/lib/codec2/dump.c
index 73a378e239..1dc34c858d 100644
--- a/gr-vocoder/lib/codec2/dump.c
+++ b/gr-vocoder/lib/codec2/dump.c
@@ -1,9 +1,9 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: dump.c
- AUTHOR......: David Rowe
- DATE CREATED: 25/8/09
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 25/8/09
+
Routines to dump data to text files for Octave analysis.
\*---------------------------------------------------------------------------*/
@@ -123,10 +123,10 @@ void dump_Sn(float Sn[]) {
for(i=0; i<M/2; i++)
fprintf(fsn,"%f\t",Sn[i]);
- fprintf(fsn,"\n");
+ fprintf(fsn,"\n");
for(i=M/2; i<M; i++)
fprintf(fsn,"%f\t",Sn[i]);
- fprintf(fsn,"\n");
+ fprintf(fsn,"\n");
}
void dump_Sw(COMP Sw[]) {
@@ -144,7 +144,7 @@ void dump_Sw(COMP Sw[]) {
for(i=0; i<FFT_ENC/2; i++)
fprintf(fsw,"%f\t",
10.0*log10(Sw[i].real*Sw[i].real + Sw[i].imag*Sw[i].imag));
- fprintf(fsw,"\n");
+ fprintf(fsw,"\n");
}
void dump_Sw_(COMP Sw_[]) {
@@ -162,7 +162,7 @@ void dump_Sw_(COMP Sw_[]) {
for(i=0; i<FFT_ENC/2; i++)
fprintf(fsw_,"%f\t",
10.0*log10(Sw_[i].real*Sw_[i].real + Sw_[i].imag*Sw_[i].imag));
- fprintf(fsw_,"\n");
+ fprintf(fsw_,"\n");
}
void dump_Ew(COMP Ew[]) {
@@ -180,7 +180,7 @@ void dump_Ew(COMP Ew[]) {
for(i=0; i<FFT_ENC/2; i++)
fprintf(few,"%f\t",
10.0*log10(Ew[i].real*Ew[i].real + Ew[i].imag*Ew[i].imag));
- fprintf(few,"\n");
+ fprintf(few,"\n");
}
void dump_model(MODEL *model) {
@@ -195,13 +195,13 @@ void dump_model(MODEL *model) {
assert(fmodel != NULL);
}
- fprintf(fmodel,"%f\t%d\t", model->Wo, model->L);
+ fprintf(fmodel,"%f\t%d\t", model->Wo, model->L);
for(l=1; l<=model->L; l++)
fprintf(fmodel,"%f\t",model->A[l]);
for(l=model->L+1; l<MAX_AMP; l++)
fprintf(fmodel,"0.0\t");
fprintf(fmodel,"%d\t",model->voiced);
- fprintf(fmodel,"\n");
+ fprintf(fmodel,"\n");
}
void dump_quantised_model(MODEL *model) {
@@ -216,12 +216,12 @@ void dump_quantised_model(MODEL *model) {
assert(fqmodel != NULL);
}
- fprintf(fqmodel,"%f\t%d\t", model->Wo, model->L);
+ fprintf(fqmodel,"%f\t%d\t", model->Wo, model->L);
for(l=1; l<=model->L; l++)
fprintf(fqmodel,"%f\t",model->A[l]);
for(l=model->L+1; l<MAX_AMP; l++)
fprintf(fqmodel,"0.0\t");
- fprintf(fqmodel,"\n");
+ fprintf(fqmodel,"\n");
}
void dump_resample(float w[], float A[], int n) {
@@ -241,7 +241,7 @@ void dump_resample(float w[], float A[], int n) {
fprintf(fres,"%f\t",w[l]);
for(l=0; l<n; l++)
fprintf(fres,"%f\t",A[l]);
- fprintf(fres,"\n");
+ fprintf(fres,"\n");
}
void dump_phase(float phase[], int L) {
@@ -260,7 +260,7 @@ void dump_phase(float phase[], int L) {
fprintf(fphase,"%f\t",phase[l]);
for(l=L+1; l<MAX_AMP; l++)
fprintf(fphase,"%f\t",0.0);
- fprintf(fphase,"\n");
+ fprintf(fphase,"\n");
}
void dump_phase_(float phase_[], int L) {
@@ -279,7 +279,7 @@ void dump_phase_(float phase_[], int L) {
fprintf(fphase_,"%f\t",phase_[l]);
for(l=L+1; l<MAX_AMP; l++)
fprintf(fphase_,"%f\t",0.0);
- fprintf(fphase_,"\n");
+ fprintf(fphase_,"\n");
}
void dump_snr(float snr) {
@@ -310,7 +310,7 @@ void dump_Pw(COMP Pw[]) {
for(i=0; i<FFT_DEC/2; i++)
fprintf(fpw,"%f\t",Pw[i].real);
- fprintf(fpw,"\n");
+ fprintf(fpw,"\n");
}
void dump_lsp(float lsp[]) {
@@ -327,7 +327,7 @@ void dump_lsp(float lsp[]) {
for(i=0; i<10; i++)
fprintf(flsp,"%f\t",lsp[i]);
- fprintf(flsp,"\n");
+ fprintf(flsp,"\n");
}
void dump_ak(float ak[], int order) {
@@ -344,7 +344,7 @@ void dump_ak(float ak[], int order) {
for(i=0; i<=order; i++)
fprintf(fak,"%f\t",ak[i]);
- fprintf(fak,"\n");
+ fprintf(fak,"\n");
}
void dump_Fw(COMP Fw[]) {
@@ -361,7 +361,7 @@ void dump_Fw(COMP Fw[]) {
for(i=0; i<256; i++)
fprintf(ffw,"%f\t",Fw[i].real);
- fprintf(ffw,"\n");
+ fprintf(ffw,"\n");
}
void dump_e(float e_hz[]) {
@@ -378,10 +378,10 @@ void dump_e(float e_hz[]) {
for(i=0; i<500/2; i++)
fprintf(fe,"%f\t",e_hz[i]);
- fprintf(fe,"\n");
+ fprintf(fe,"\n");
for(i=500/2; i<500; i++)
fprintf(fe,"%f\t",e_hz[i]);
- fprintf(fe,"\n");
+ fprintf(fe,"\n");
}
void dump_sq(float sq[]) {
@@ -398,10 +398,10 @@ void dump_sq(float sq[]) {
for(i=0; i<M/2; i++)
fprintf(fsq,"%f\t",sq[i]);
- fprintf(fsq,"\n");
+ fprintf(fsq,"\n");
for(i=M/2; i<M; i++)
fprintf(fsq,"%f\t",sq[i]);
- fprintf(fsq,"\n");
+ fprintf(fsq,"\n");
}
void dump_dec(COMP Fw[]) {
@@ -418,7 +418,7 @@ void dump_dec(COMP Fw[]) {
for(i=0; i<320/5; i++)
fprintf(fdec,"%f\t",Fw[i].real);
- fprintf(fdec,"\n");
+ fprintf(fdec,"\n");
}
void dump_bg(float e, float bg_est, float percent_uv) {
@@ -432,7 +432,7 @@ void dump_bg(float e, float bg_est, float percent_uv) {
assert(fbg != NULL);
}
- fprintf(fbg,"%f\t%f\t%f\n", e, bg_est, percent_uv);
+ fprintf(fbg,"%f\t%f\t%f\n", e, bg_est, percent_uv);
}
void dump_E(float E) {
@@ -463,7 +463,7 @@ void dump_Rk(float Rk[]) {
for(i=0; i<P_MAX; i++)
fprintf(frk,"%f\t",Rk[i]);
- fprintf(frk,"\n");
+ fprintf(frk,"\n");
}
#endif
diff --git a/gr-vocoder/lib/codec2/dump.h b/gr-vocoder/lib/codec2/dump.h
index eeddd34065..4b92d009d9 100644
--- a/gr-vocoder/lib/codec2/dump.h
+++ b/gr-vocoder/lib/codec2/dump.h
@@ -1,9 +1,9 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: dump.h
- AUTHOR......: David Rowe
- DATE CREATED: 25/8/09
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 25/8/09
+
Routines to dump data to text files for Octave analysis.
\*---------------------------------------------------------------------------*/
diff --git a/gr-vocoder/lib/codec2/fft.c b/gr-vocoder/lib/codec2/fft.c
index a3027e4cd5..19f3141a80 100644
--- a/gr-vocoder/lib/codec2/fft.c
+++ b/gr-vocoder/lib/codec2/fft.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: fft.c
- AUTHOR......: Bruce Robertson
- DATE CREATED: 20/11/2010
-
- Bridging function to the kiss_fft package.
-
+
+ FILE........: fft.c
+ AUTHOR......: Bruce Robertson
+ DATE CREATED: 20/11/2010
+
+ Bridging function to the kiss_fft package.
+
\*---------------------------------------------------------------------------*/
/*
@@ -29,9 +29,9 @@
#include "kiss_fft.h"
/*---------------------------------------------------------------------------*\
-
- GLOBALS
-
+
+ GLOBALS
+
\*---------------------------------------------------------------------------*/
kiss_fft_cpx *fin;
@@ -40,10 +40,10 @@ kiss_fft_cfg cfg_forward;
kiss_fft_cfg cfg_reverse;
/*---------------------------------------------------------------------------*\
-
- initialize_fft(int n)
-
- Initialisation function for kiss_fft. This assumes that all calls to fft()
+
+ initialize_fft(int n)
+
+ Initialisation function for kiss_fft. This assumes that all calls to fft()
use the same datatypes and are one arrays of the same size.
\*---------------------------------------------------------------------------*/
@@ -62,8 +62,8 @@ initialize_fft (int n)
}
/*---------------------------------------------------------------------------*\
-
- fft(float x[], int n, int isign)
+
+ fft(float x[], int n, int isign)
Function that calls kiss_fft with the signature of four1 from NRC.
\*---------------------------------------------------------------------------*/
diff --git a/gr-vocoder/lib/codec2/fft.h b/gr-vocoder/lib/codec2/fft.h
index 84c6737bd8..d5f83045e1 100644
--- a/gr-vocoder/lib/codec2/fft.h
+++ b/gr-vocoder/lib/codec2/fft.h
@@ -1,5 +1,5 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: fft.h
AUTHOR......: Bruce Robertson
DATE CREATED: 29/11/2010
diff --git a/gr-vocoder/lib/codec2/fq20.sh b/gr-vocoder/lib/codec2/fq20.sh
index b83784b437..9ccf739dc2 100755
--- a/gr-vocoder/lib/codec2/fq20.sh
+++ b/gr-vocoder/lib/codec2/fq20.sh
@@ -1,7 +1,7 @@
#!/bin/sh
# fq20.shsh
# David Rowe 27 July 2010
-#
+#
# Decode a file with fully quantised codec at 20ms frame rate
../src/sinedec ../raw/$1.raw $1.mdl -o $1_phase0_lsp_20_EWo2.raw --phase 0 --lpc 10 --lsp --postfilter --dec
diff --git a/gr-vocoder/lib/codec2/generate_codebook.c b/gr-vocoder/lib/codec2/generate_codebook.c
index 0bea80d854..705f29d382 100644
--- a/gr-vocoder/lib/codec2/generate_codebook.c
+++ b/gr-vocoder/lib/codec2/generate_codebook.c
@@ -1,9 +1,9 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: generate_codebook.c
- AUTHOR......: Bruce Perens
- DATE CREATED: 29 Sep 2010
-
+ AUTHOR......: Bruce Perens
+ DATE CREATED: 29 Sep 2010
+
Generate header files containing LSP quantisers, runs at compile time.
\*---------------------------------------------------------------------------*/
@@ -92,7 +92,7 @@ get_float(FILE * in, const char * name, char * * cursor, char * buffer,
while ( (c = *s) != '\0' && !isdigit(c) && c != '-' && c != '.' )
s++;
-
+
/* Comments start with "#" and continue to the end of the line. */
if ( c != '\0' && c != '#' ) {
char * end = 0;
diff --git a/gr-vocoder/lib/codec2/globals.c b/gr-vocoder/lib/codec2/globals.c
index f2182f79a5..da2faf7229 100644
--- a/gr-vocoder/lib/codec2/globals.c
+++ b/gr-vocoder/lib/codec2/globals.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: globals.c
- AUTHOR......: David Rowe
- DATE CREATED: 11/5/94
-
- Globals for sinusoidal speech coder.
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 11/5/94
+
+ Globals for sinusoidal speech coder.
+
\*---------------------------------------------------------------------------*/
/*
diff --git a/gr-vocoder/lib/codec2/globals.h b/gr-vocoder/lib/codec2/globals.h
index cef720344c..d01e7b4e92 100644
--- a/gr-vocoder/lib/codec2/globals.h
+++ b/gr-vocoder/lib/codec2/globals.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: globals.h
AUTHOR......: David Rowe
- DATE CREATED: 1/11/94
-
- Globals for sinusoidal speech coder.
-
+ DATE CREATED: 1/11/94
+
+ Globals for sinusoidal speech coder.
+
\*---------------------------------------------------------------------------*/
/*
diff --git a/gr-vocoder/lib/codec2/interp.c b/gr-vocoder/lib/codec2/interp.c
index 257c4a81c2..64372b050f 100644
--- a/gr-vocoder/lib/codec2/interp.c
+++ b/gr-vocoder/lib/codec2/interp.c
@@ -41,10 +41,10 @@ float sample_log_amp(MODEL *model, float w);
/*---------------------------------------------------------------------------*\
- FUNCTION....: interp()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/10
-
+ FUNCTION....: interp()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/10
+
Given two frames decribed by model parameters 20ms apart, determines
the model parameters of the 10ms frame between them. Assumes
voicing is available for middle (interpolated) frame. Outputs are
@@ -58,7 +58,7 @@ float sample_log_amp(MODEL *model, float w);
When this function is used (--dec mode) bg noise appears to be
amplitude modulated, and gets louder. The interp_lsp() function
below seems to do a better job.
-
+
\*---------------------------------------------------------------------------*/
void interpolate(
@@ -97,13 +97,13 @@ void interpolate(
/*---------------------------------------------------------------------------*\
FUNCTION....: sample_log_amp()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/10
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/10
+
Samples the amplitude envelope at an arbitrary frequency w. Uses
linear interpolation in the log domain to sample between harmonic
amplitudes.
-
+
\*---------------------------------------------------------------------------*/
float sample_log_amp(MODEL *model, float w)
@@ -125,9 +125,9 @@ float sample_log_amp(MODEL *model, float w)
log_amp = (1.0-f)*log10(model->A[model->L] + 1E-6);
}
else {
- log_amp = (1.0-f)*log10(model->A[m] + 1E-6) +
+ log_amp = (1.0-f)*log10(model->A[m] + 1E-6) +
f*log10(model->A[m+1] + 1E-6);
- //printf("m=%d A[m] %f A[m+1] %f x %f %f %f\n", m, model->A[m],
+ //printf("m=%d A[m] %f A[m+1] %f x %f %f %f\n", m, model->A[m],
// model->A[m+1], pow(10.0, log_amp),
// (1-f), f);
}
@@ -138,13 +138,13 @@ float sample_log_amp(MODEL *model, float w)
/*---------------------------------------------------------------------------*\
FUNCTION....: sample_log_amp_quad()
- AUTHOR......: David Rowe
- DATE CREATED: 9 March 2011
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 9 March 2011
+
Samples the amplitude envelope at an arbitrary frequency w. Uses
quadratic interpolation in the log domain to sample between harmonic
amplitudes.
-
+
y(x) = ax*x + bx + c
We assume three points are x=-1, x=0, x=1, which we map to m-1,m,m+1
@@ -172,7 +172,7 @@ float sample_log_amp_quad(MODEL *model, float w)
log_amp = a*x*x + b*x + c;
//printf("m=%d A[m-1] %f A[m] %f A[m+1] %f w %f x %f log_amp %f\n", m,
- // model->A[m-1],
+ // model->A[m-1],
// model->A[m], model->A[m+1], w, x, pow(10.0, log_amp));
return log_amp;
}
@@ -180,14 +180,14 @@ float sample_log_amp_quad(MODEL *model, float w)
/*---------------------------------------------------------------------------*\
FUNCTION....: sample_log_amp_quad_nl()
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 10 March 2011
-
+
Samples the amplitude envelope at an arbitrary frequency w. Uses
quadratic interpolation in the log domain to sample between harmonic
amplitudes. This version can handle non-linear steps along a freq
axis defined by arbitrary steps.
-
+
y(x) = ax*x + bx + c
We assume three points are (x_1,y_1), (0,y0) and (x1,y1).
@@ -215,9 +215,9 @@ float sample_log_amp_quad_nl(
for (i=0; i<np; i++)
if (fabs(w[i] - w_sample) < best_dist) {
best_dist = fabs(w[i] - w_sample);
- m = i;
+ m = i;
}
-
+
/* stay one point away from edge of array */
if (m < 1) m = 1;
@@ -234,13 +234,13 @@ float sample_log_amp_quad_nl(
a = (y_1*x1 - y1*x_1 + c*x_1 - c*x1)/(x_1*x_1*x1 - x1*x1*x_1);
b = (y1 -a*x1*x1 - c)/x1;
x = w_sample - w[m];
-
+
//printf("%f %f %f\n", w[0], w[1], w[2]);
//printf("%f %f %f %f %f %f\n", x_1, y_1, 0.0, y0, x1, y1);
log_amp = a*x*x + b*x + c;
//printf("a %f b %f c %f\n", a, b, c);
//printf("m=%d A[m-1] %f A[m] %f A[m+1] %f w_sample %f w[m] %f x %f log_amp %f\n", m,
- // A[m-1],
+ // A[m-1],
// A[m], A[m+1], w_sample, w[m], x, log_amp);
//exit(0);
return log_amp;
@@ -254,20 +254,20 @@ float fres[] = {100, 200, 300, 400, 500, 600, 700, 800, 900, 1000,
/*---------------------------------------------------------------------------*\
FUNCTION....: resample_amp_nl()
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 7 March 2011
-
- Converts the current model with L {Am} samples spaced Wo apart to
+
+ Converts the current model with L {Am} samples spaced Wo apart to
RES_POINTS samples spaced Wo/RES_POINTS apart. Then subtracts
from the previous frames samples to get the delta.
\*---------------------------------------------------------------------------*/
-void resample_amp_fixed(MODEL *model,
+void resample_amp_fixed(MODEL *model,
float w[], float A[],
float wres[], float Ares[],
- float AresdB_prev[],
- float AresdB[],
+ float AresdB_prev[],
+ float AresdB[],
float deltat[])
{
int i;
@@ -280,7 +280,7 @@ void resample_amp_fixed(MODEL *model,
for(i=0; i<RES_POINTS; i++) {
wres[i] = fres[i]*PI/4000.0;
}
-
+
for(i=0; i<RES_POINTS; i++) {
Ares[i] = pow(10.0,sample_log_amp_quad_nl(w, A, model->L, wres[i]));
}
@@ -297,13 +297,13 @@ void resample_amp_fixed(MODEL *model,
/*---------------------------------------------------------------------------*\
FUNCTION....: resample_amp_nl()
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 7 March 2011
-
+
Converts the current model with L {Am} samples spaced Wo apart to M
samples spaced Wo/M apart. Then converts back to L {Am} samples.
used to prototype constant rate Amplitude encoding ideas.
-
+
Returns the SNR in dB.
\*---------------------------------------------------------------------------*/
@@ -346,7 +346,7 @@ float resample_amp_nl(MODEL *model, int m, float AresdB_prev[])
#endif
signal = noise = 0.0;
-
+
for(i=1; i<model->L; i++) {
new_A = pow(10.0,sample_log_amp_quad_nl(wres, Ares, RES_POINTS, model->Wo*i));
signal += pow(model->A[i], 2.0);
@@ -364,13 +364,13 @@ float resample_amp_nl(MODEL *model, int m, float AresdB_prev[])
/*---------------------------------------------------------------------------*\
FUNCTION....: resample_amp()
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 10 March 2011
-
+
Converts the current model with L {Am} samples spaced Wo apart to M
samples with a non-linear spacing. Then converts back to L {Am}
samples. used to prototype constant rate Amplitude encoding ideas.
-
+
Returns the SNR in dB.
\*---------------------------------------------------------------------------*/
@@ -394,7 +394,7 @@ float resample_amp(MODEL *model, int m)
//dump_resample(&model_m);
signal = noise = 0.0;
-
+
for(i=1; i<model->L/4; i++) {
new_A = pow(10,sample_log_amp_quad(&model_m, i*model->Wo));
signal += pow(model->A[i], 2.0);
@@ -411,10 +411,10 @@ float resample_amp(MODEL *model, int m)
/*---------------------------------------------------------------------------*\
- FUNCTION....: interp_lsp()
- AUTHOR......: David Rowe
+ FUNCTION....: interp_lsp()
+ AUTHOR......: David Rowe
DATE CREATED: 10 Nov 2010
-
+
Given two frames decribed by model parameters 20ms apart, determines
the model parameters of the 10ms frame between them. Assumes
voicing is available for middle (interpolated) frame. Outputs are
@@ -422,7 +422,7 @@ float resample_amp(MODEL *model, int m)
This version uses interpolation of LSPs, seems to do a better job
with bg noise.
-
+
\*---------------------------------------------------------------------------*/
void interpolate_lsp(
@@ -469,5 +469,5 @@ void interpolate_lsp(
/* convert back to amplitudes */
lsp_to_lpc(lsps, ak_interp, LPC_ORD);
- aks_to_M2(ak_interp, LPC_ORD, interp, e, &snr, 0);
+ aks_to_M2(ak_interp, LPC_ORD, interp, e, &snr, 0);
}
diff --git a/gr-vocoder/lib/codec2/interp.h b/gr-vocoder/lib/codec2/interp.h
index d41eac3f81..d509726140 100644
--- a/gr-vocoder/lib/codec2/interp.h
+++ b/gr-vocoder/lib/codec2/interp.h
@@ -31,7 +31,7 @@
#define RES_POINTS 20
void interpolate(MODEL *interp, MODEL *prev, MODEL *next);
-void interpolate_lsp(MODEL *interp, MODEL *prev, MODEL *next,
+void interpolate_lsp(MODEL *interp, MODEL *prev, MODEL *next,
float *prev_lsps, float prev_e,
float *next_lsps, float next_e,
float *ak_interp);
diff --git a/gr-vocoder/lib/codec2/kiss_fft.c b/gr-vocoder/lib/codec2/kiss_fft.c
index 465d6c97a0..17b4e72112 100644
--- a/gr-vocoder/lib/codec2/kiss_fft.c
+++ b/gr-vocoder/lib/codec2/kiss_fft.c
@@ -250,7 +250,7 @@ void kf_work(
const kiss_fft_cpx * Fout_end = Fout + p*m;
#ifdef _OPENMP
- // use openmp extensions at the
+ // use openmp extensions at the
// top-level (not recursive)
if (fstride==1 && p<=5)
{
@@ -258,15 +258,15 @@ void kf_work(
// execute the p different work units in different threads
# pragma omp parallel for
- for (k=0;k<p;++k)
+ for (k=0;k<p;++k)
kf_work( Fout +k*m, f+ fstride*in_stride*k,fstride*p,in_stride,factors,st);
// all threads have joined by this point
switch (p) {
case 2: kf_bfly2(Fout,fstride,st,m); break;
- case 3: kf_bfly3(Fout,fstride,st,m); break;
+ case 3: kf_bfly3(Fout,fstride,st,m); break;
case 4: kf_bfly4(Fout,fstride,st,m); break;
- case 5: kf_bfly5(Fout,fstride,st,m); break;
+ case 5: kf_bfly5(Fout,fstride,st,m); break;
default: kf_bfly_generic(Fout,fstride,st,m,p); break;
}
return;
@@ -282,7 +282,7 @@ void kf_work(
do{
// recursive call:
// DFT of size m*p performed by doing
- // p instances of smaller DFTs of size m,
+ // p instances of smaller DFTs of size m,
// each one takes a decimated version of the input
kf_work( Fout , f, fstride*p, in_stride, factors,st);
f += fstride*in_stride;
@@ -291,21 +291,21 @@ void kf_work(
Fout=Fout_beg;
- // recombine the p smaller DFTs
+ // recombine the p smaller DFTs
switch (p) {
case 2: kf_bfly2(Fout,fstride,st,m); break;
- case 3: kf_bfly3(Fout,fstride,st,m); break;
+ case 3: kf_bfly3(Fout,fstride,st,m); break;
case 4: kf_bfly4(Fout,fstride,st,m); break;
- case 5: kf_bfly5(Fout,fstride,st,m); break;
+ case 5: kf_bfly5(Fout,fstride,st,m); break;
default: kf_bfly_generic(Fout,fstride,st,m,p); break;
}
}
/* facbuf is populated by p1,m1,p2,m2, ...
- where
+ where
p[i] * m[i] = m[i-1]
m0 = n */
-static
+static
void kf_factor(int n,int * facbuf)
{
int p=4;
diff --git a/gr-vocoder/lib/codec2/kiss_fft.h b/gr-vocoder/lib/codec2/kiss_fft.h
index 64c50f4aae..c01722cad7 100644
--- a/gr-vocoder/lib/codec2/kiss_fft.h
+++ b/gr-vocoder/lib/codec2/kiss_fft.h
@@ -28,17 +28,17 @@ extern "C" {
# define kiss_fft_scalar __m128
#define KISS_FFT_MALLOC(nbytes) _mm_malloc(nbytes,16)
#define KISS_FFT_FREE _mm_free
-#else
+#else
#define KISS_FFT_MALLOC malloc
#define KISS_FFT_FREE free
-#endif
+#endif
#ifdef FIXED_POINT
-#include <sys/types.h>
+#include <sys/types.h>
# if (FIXED_POINT == 32)
# define kiss_fft_scalar int32_t
-# else
+# else
# define kiss_fft_scalar int16_t
# endif
#else
@@ -55,9 +55,9 @@ typedef struct {
typedef struct kiss_fft_state* kiss_fft_cfg;
-/*
+/*
* kiss_fft_alloc
- *
+ *
* Initialize a FFT (or IFFT) algorithm's cfg/state buffer.
*
* typical usage: kiss_fft_cfg mycfg=kiss_fft_alloc(1024,0,NULL,NULL);
@@ -67,18 +67,18 @@ typedef struct kiss_fft_state* kiss_fft_cfg;
*
* If lenmem is NULL, then kiss_fft_alloc will allocate a cfg buffer using malloc.
* The returned value should be free()d when done to avoid memory leaks.
- *
+ *
* The state can be placed in a user supplied buffer 'mem':
* If lenmem is not NULL and mem is not NULL and *lenmem is large enough,
* then the function places the cfg in mem and the size used in *lenmem
* and returns mem.
- *
+ *
* If lenmem is not NULL and ( mem is NULL or *lenmem is not large enough),
- * then the function returns NULL and places the minimum cfg
+ * then the function returns NULL and places the minimum cfg
* buffer size in *lenmem.
* */
-kiss_fft_cfg kiss_fft_alloc(int nfft,int inverse_fft,void * mem,size_t * lenmem);
+kiss_fft_cfg kiss_fft_alloc(int nfft,int inverse_fft,void * mem,size_t * lenmem);
/*
* kiss_fft(cfg,in_out_buf)
@@ -97,16 +97,16 @@ void kiss_fft(kiss_fft_cfg cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout);
* */
void kiss_fft_stride(kiss_fft_cfg cfg,const kiss_fft_cpx *fin,kiss_fft_cpx *fout,int fin_stride);
-/* If kiss_fft_alloc allocated a buffer, it is one contiguous
+/* If kiss_fft_alloc allocated a buffer, it is one contiguous
buffer and can be simply free()d when no longer needed*/
#define kiss_fft_free free
/*
- Cleans up some memory that gets managed internally. Not necessary to call, but it might clean up
+ Cleans up some memory that gets managed internally. Not necessary to call, but it might clean up
your compiler output to call this before you exit.
*/
void kiss_fft_cleanup(void);
-
+
/*
* Returns the smallest integer k, such that k>=n and k has only "fast" factors (2,3,5)
@@ -118,7 +118,7 @@ int kiss_fft_next_fast_size(int n);
(kiss_fft_next_fast_size( ((n)+1)>>1)<<1)
#ifdef __cplusplus
-}
+}
#endif
#endif
diff --git a/gr-vocoder/lib/codec2/lpc.c b/gr-vocoder/lib/codec2/lpc.c
index ba80113773..1784f75ca1 100644
--- a/gr-vocoder/lib/codec2/lpc.c
+++ b/gr-vocoder/lib/codec2/lpc.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: lpc.c
- AUTHOR......: David Rowe
- DATE CREATED: 30/9/90
-
- Linear Prediction functions written in C.
-
+
+ FILE........: lpc.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 30/9/90
+
+ Linear Prediction functions written in C.
+
\*---------------------------------------------------------------------------*/
/*
@@ -34,11 +34,11 @@
#include "lpc.h"
/*---------------------------------------------------------------------------*\
-
- hanning_window()
-
- Hanning windows a frame of speech samples.
-
+
+ hanning_window()
+
+ Hanning windows a frame of speech samples.
+
\*---------------------------------------------------------------------------*/
void hanning_window(
@@ -54,12 +54,12 @@ void hanning_window(
}
/*---------------------------------------------------------------------------*\
-
- autocorrelate()
-
- Finds the first P autocorrelation values of an array of windowed speech
- samples Sn[].
-
+
+ autocorrelate()
+
+ Finds the first P autocorrelation values of an array of windowed speech
+ samples Sn[].
+
\*---------------------------------------------------------------------------*/
void autocorrelate(
@@ -79,12 +79,12 @@ void autocorrelate(
}
/*---------------------------------------------------------------------------*\
-
- autocorrelate_freq()
-
+
+ autocorrelate_freq()
+
Finds the first P autocorrelation values from an array of frequency domain
- power samples.
-
+ power samples.
+
\*---------------------------------------------------------------------------*/
void autocorrelate_freq(
@@ -106,18 +106,18 @@ void autocorrelate_freq(
}
/*---------------------------------------------------------------------------*\
-
- levinson_durbin()
-
- Given P+1 autocorrelation coefficients, finds P Linear Prediction Coeff.
+
+ levinson_durbin()
+
+ Given P+1 autocorrelation coefficients, finds P Linear Prediction Coeff.
(LPCs) where P is the order of the LPC all-pole model. The Levinson-Durbin
- algorithm is used, and is described in:
-
- J. Makhoul
- "Linear prediction, a tutorial review"
- Proceedings of the IEEE
- Vol-63, No. 4, April 1975
-
+ algorithm is used, and is described in:
+
+ J. Makhoul
+ "Linear prediction, a tutorial review"
+ Proceedings of the IEEE
+ Vol-63, No. 4, April 1975
+
\*---------------------------------------------------------------------------*/
void levinson_durbin(
@@ -152,18 +152,18 @@ void levinson_durbin(
for(i=1; i<=order; i++)
lpcs[i] = a[order][i];
- lpcs[0] = 1.0;
+ lpcs[0] = 1.0;
}
/*---------------------------------------------------------------------------*\
-
- inverse_filter()
-
+
+ inverse_filter()
+
Inverse Filter, A(z). Produces an array of residual samples from an array
- of input samples and linear prediction coefficients.
-
+ of input samples and linear prediction coefficients.
+
The filter memory is stored in the first order samples of the input array.
-
+
\*---------------------------------------------------------------------------*/
void inverse_filter(
@@ -180,26 +180,26 @@ void inverse_filter(
res[i] = 0.0;
for(j=0; j<=order; j++)
res[i] += Sn[i-j]*a[j];
- }
+ }
}
/*---------------------------------------------------------------------------*\
-
- synthesis_filter()
-
- C version of the Speech Synthesis Filter, 1/A(z). Given an array of
+
+ synthesis_filter()
+
+ C version of the Speech Synthesis Filter, 1/A(z). Given an array of
residual or excitation samples, and the the LP filter coefficients, this
function will produce an array of speech samples. This filter structure is
- IIR.
-
- The synthesis filter has memory as well, this is treated in the same way
- as the memory for the inverse filter (see inverse_filter() notes above).
- The difference is that the memory for the synthesis filter is stored in
+ IIR.
+
+ The synthesis filter has memory as well, this is treated in the same way
+ as the memory for the inverse filter (see inverse_filter() notes above).
+ The difference is that the memory for the synthesis filter is stored in
the output array, wheras the memory of the inverse filter is stored in the
- input array.
-
- Note: the calling function must update the filter memory.
-
+ input array.
+
+ Note: the calling function must update the filter memory.
+
\*---------------------------------------------------------------------------*/
void synthesis_filter(
@@ -222,12 +222,12 @@ void synthesis_filter(
}
/*---------------------------------------------------------------------------*\
-
- find_aks()
-
- This function takes a frame of samples, and determines the linear
- prediction coefficients for that frame of samples.
-
+
+ find_aks()
+
+ This function takes a frame of samples, and determines the linear
+ prediction coefficients for that frame of samples.
+
\*---------------------------------------------------------------------------*/
void find_aks(
@@ -257,11 +257,11 @@ void find_aks(
}
/*---------------------------------------------------------------------------*\
-
- weight()
-
- Weights a vector of LPCs.
-
+
+ weight()
+
+ Weights a vector of LPCs.
+
\*---------------------------------------------------------------------------*/
void weight(
@@ -272,8 +272,8 @@ void weight(
)
{
int i;
-
+
for(i=1; i<=order; i++)
akw[i] = ak[i]*pow(gamma,(float)i);
}
-
+
diff --git a/gr-vocoder/lib/codec2/lpc.h b/gr-vocoder/lib/codec2/lpc.h
index ead05e1bad..9125189d1b 100644
--- a/gr-vocoder/lib/codec2/lpc.h
+++ b/gr-vocoder/lib/codec2/lpc.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: lpc.h
- AUTHOR......: David Rowe
- DATE CREATED: 24/8/09
-
- Linear Prediction functions written in C.
-
+
+ FILE........: lpc.h
+ AUTHOR......: David Rowe
+ DATE CREATED: 24/8/09
+
+ Linear Prediction functions written in C.
+
\*---------------------------------------------------------------------------*/
/*
diff --git a/gr-vocoder/lib/codec2/lsp.c b/gr-vocoder/lib/codec2/lsp.c
index 47001c1efd..b57507bb42 100644
--- a/gr-vocoder/lib/codec2/lsp.c
+++ b/gr-vocoder/lib/codec2/lsp.c
@@ -153,7 +153,7 @@ int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta)
float Q[LSP_MAX_ORDER + 1];
float P[LSP_MAX_ORDER + 1];
- flag = 1;
+ flag = 1;
m = lpcrdr/2; /* order of P'(z) & Q'(z) polynimials */
/* Allocate memory space for polynomials */
@@ -274,7 +274,7 @@ void lsp_to_lpc(float *lsp, float *ak, int lpcrdr)
int m = lpcrdr/2;
float freq[LSP_MAX_ORDER];
float Wp[(LSP_MAX_ORDER * 4) + 2];
-
+
/* convert from radians to the x=cos(w) domain */
for(i=0; i<lpcrdr; i++)
diff --git a/gr-vocoder/lib/codec2/nlp.c b/gr-vocoder/lib/codec2/nlp.c
index 42ae909196..0d5e530ce6 100644
--- a/gr-vocoder/lib/codec2/nlp.c
+++ b/gr-vocoder/lib/codec2/nlp.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: nlp.c
- AUTHOR......: David Rowe
- DATE CREATED: 23/3/93
-
- Non Linear Pitch (NLP) estimation functions.
-
+
+ FILE........: nlp.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/3/93
+
+ Non Linear Pitch (NLP) estimation functions.
+
\*---------------------------------------------------------------------------*/
/*
@@ -35,9 +35,9 @@
#include <stdlib.h>
/*---------------------------------------------------------------------------*\
-
- DEFINES
-
+
+ DEFINES
+
\*---------------------------------------------------------------------------*/
#define PMAX_M 600 /* maximum NLP analysis window size */
@@ -52,9 +52,9 @@
#define NLP_NTAP 48 /* Decimation LPF order */
/*---------------------------------------------------------------------------*\
-
- GLOBALS
-
+
+ GLOBALS
+
\*---------------------------------------------------------------------------*/
/* 48 tap 600Hz low pass FIR filter coefficients */
@@ -117,14 +117,14 @@ typedef struct {
} NLP;
float post_process_mbe(COMP Fw[], int pmin, int pmax, float gmax);
-float post_process_sub_multiples(COMP Fw[],
+float post_process_sub_multiples(COMP Fw[],
int pmin, int pmax, float gmax, int gmax_bin,
float *prev_Wo);
/*---------------------------------------------------------------------------*\
-
- nlp_create()
-
+
+ nlp_create()
+
Initialisation function for NLP pitch estimator.
\*---------------------------------------------------------------------------*/
@@ -149,9 +149,9 @@ void *nlp_create()
}
/*---------------------------------------------------------------------------*\
-
+
nlp_destory()
-
+
Initialisation function for NLP pitch estimator.
\*---------------------------------------------------------------------------*/
@@ -163,9 +163,9 @@ void nlp_destroy(void *nlp_state)
}
/*---------------------------------------------------------------------------*\
-
- nlp()
-
+
+ nlp()
+
Determines the pitch in samples using the Non Linear Pitch (NLP)
algorithm [1]. Returns the fundamental in Hz. Note that the actual
pitch estimate is for the centre of the M sample Sn[] vector, not
@@ -189,11 +189,11 @@ void nlp_destroy(void *nlp_state)
References:
[1] http://www.itr.unisa.edu.au/~steven/thesis/dgr.pdf Chapter 4
-
+
\*---------------------------------------------------------------------------*/
float nlp(
- void *nlp_state,
+ void *nlp_state,
float Sn[], /* input speech vector */
int n, /* frames shift (no. new samples in Sn[]) */
int m, /* analysis window size */
@@ -271,7 +271,7 @@ float nlp(
}
}
- best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin,
+ best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin,
prev_Wo);
/* Shift samples in buffer to make room for new samples */
@@ -282,13 +282,13 @@ float nlp(
/* return pitch and F0 estimate */
*pitch = (float)SAMPLE_RATE/best_f0;
- return(best_f0);
+ return(best_f0);
}
/*---------------------------------------------------------------------------*\
-
- post_process_sub_multiples()
-
+
+ post_process_sub_multiples()
+
Given the global maximma of Fw[] we search interger submultiples for
local maxima. If local maxima exist and they are above an
experimentally derived threshold (OK a magic number I pulled out of
@@ -306,7 +306,7 @@ float nlp(
\*---------------------------------------------------------------------------*/
-float post_process_sub_multiples(COMP Fw[],
+float post_process_sub_multiples(COMP Fw[],
int pmin, int pmax, float gmax, int gmax_bin,
float *prev_Wo)
{
@@ -361,4 +361,4 @@ float post_process_sub_multiples(COMP Fw[],
return best_f0;
}
-
+
diff --git a/gr-vocoder/lib/codec2/nlp.h b/gr-vocoder/lib/codec2/nlp.h
index 88a3733dc2..5e11f1186e 100644
--- a/gr-vocoder/lib/codec2/nlp.h
+++ b/gr-vocoder/lib/codec2/nlp.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: nlp.c
- AUTHOR......: David Rowe
- DATE CREATED: 23/3/93
-
- Non Linear Pitch (NLP) estimation functions.
-
+
+ FILE........: nlp.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/3/93
+
+ Non Linear Pitch (NLP) estimation functions.
+
\*---------------------------------------------------------------------------*/
/*
@@ -32,7 +32,7 @@
void *nlp_create();
void nlp_destroy(void *nlp_state);
-float nlp(void *nlp_state, float Sn[], int n, int m, int pmin, int pmax,
+float nlp(void *nlp_state, float Sn[], int n, int m, int pmin, int pmax,
float *pitch, COMP Sw[], float *prev_Wo);
float test_candidate_mbe(COMP Sw[], float f0, COMP Sw_[]);
diff --git a/gr-vocoder/lib/codec2/pack.c b/gr-vocoder/lib/codec2/pack.c
index 31551dfc43..e04c9378cf 100644
--- a/gr-vocoder/lib/codec2/pack.c
+++ b/gr-vocoder/lib/codec2/pack.c
@@ -65,7 +65,7 @@ pack(
bitArray[wordIndex] |=
((unsigned char)((field >> (fieldWidth - sliceWidth))
<< (bitsLeft - sliceWidth)));
-
+
*bitIndex = bI + sliceWidth;
fieldWidth -= sliceWidth;
} while ( fieldWidth != 0 );
@@ -91,7 +91,7 @@ unpack(
bitsLeft < fieldWidth ? bitsLeft : fieldWidth;
field |= (((bitArray[bI >> ShiftRight] >> (bitsLeft - sliceWidth)) & ((1 << sliceWidth) - 1)) << (fieldWidth - sliceWidth));
-
+
*bitIndex = bI + sliceWidth;
fieldWidth -= sliceWidth;
} while ( fieldWidth != 0 );
diff --git a/gr-vocoder/lib/codec2/phase.c b/gr-vocoder/lib/codec2/phase.c
index 0e1a14a601..69cc6697fc 100644
--- a/gr-vocoder/lib/codec2/phase.c
+++ b/gr-vocoder/lib/codec2/phase.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: phase.c
- AUTHOR......: David Rowe
- DATE CREATED: 1/2/09
-
+
+ FILE........: phase.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 1/2/09
+
Functions for modelling and synthesising phase.
-
+
\*---------------------------------------------------------------------------*/
/*
@@ -22,7 +22,7 @@
License for more details.
You should have received a copy of the GNU Lesser General Public License
- along with this program; if not,see <http://www.gnu.org/licenses/>.
+ along with this program; if not,see <http://www.gnu.org/licenses/>.
*/
#include "defines.h"
@@ -101,14 +101,14 @@ void aks_to_H(
phase_synth_zero_order()
- Synthesises phases based on SNR and a rule based approach. No phase
+ Synthesises phases based on SNR and a rule based approach. No phase
parameters are required apart from the SNR (which can be reduced to a
1 bit V/UV decision per frame).
The phase of each harmonic is modelled as the phase of a LPC
synthesis filter excited by an impulse. Unlike the first order
model the position of the impulse is not transmitted, so we create
- an excitation pulse train using a rule based approach.
+ an excitation pulse train using a rule based approach.
Consider a pulse train with a pulse starting time n=0, with pulses
repeated at a rate of Wo, the fundamental frequency. A pulse train
@@ -149,10 +149,10 @@ void aks_to_H(
This E[m] then gets passed through the LPC synthesis filter to
determine the final harmonic phase.
-
+
Comparing to speech synthesised using original phases:
- - Through headphones speech synthesised with this model is not as
+ - Through headphones speech synthesised with this model is not as
good. Through a loudspeaker it is very close to original phases.
- If there are voicing errors, the speech can sound clicky or
@@ -207,21 +207,21 @@ void phase_synth_zero_order(
G = 1.0;
aks_to_H(model, aks, G, H, order);
- /*
+ /*
Update excitation fundamental phase track, this sets the position
of each pitch pulse during voiced speech. After much experiment
I found that using just this frame's Wo improved quality for UV
sounds compared to interpolating two frames Wo like this:
-
+
ex_phase[0] += (*prev_Wo+mode->Wo)*N/2;
*/
-
+
ex_phase[0] += (model->Wo)*N;
ex_phase[0] -= TWO_PI*floor(ex_phase[0]/TWO_PI + 0.5);
r = TWO_PI/GLOTTAL_FFT_SIZE;
for(m=1; m<=model->L; m++) {
-
+
/* generate excitation */
if (model->voiced) {
@@ -254,7 +254,7 @@ void phase_synth_zero_order(
A_[m].imag = H[m].imag*Ex[m].real + H[m].real*Ex[m].imag;
/* modify sinusoidal phase */
-
+
new_phi = atan2(A_[m].imag, A_[m].real+1E-12);
model->phi[m] = new_phi;
}
diff --git a/gr-vocoder/lib/codec2/phase.h b/gr-vocoder/lib/codec2/phase.h
index 833bc7cdce..4f1a620899 100644
--- a/gr-vocoder/lib/codec2/phase.h
+++ b/gr-vocoder/lib/codec2/phase.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
- FILE........: phase.h
- AUTHOR......: David Rowe
- DATE CREATED: 1/2/09
-
+
+ FILE........: phase.h
+ AUTHOR......: David Rowe
+ DATE CREATED: 1/2/09
+
Functions for modelling phase.
-
+
\*---------------------------------------------------------------------------*/
/*
@@ -28,7 +28,7 @@
#ifndef __PHASE__
#define __PHASE__
-void phase_synth_zero_order(MODEL *model, float aks[], float *ex_phase,
+void phase_synth_zero_order(MODEL *model, float aks[], float *ex_phase,
int order);
#endif
diff --git a/gr-vocoder/lib/codec2/postfilter.c b/gr-vocoder/lib/codec2/postfilter.c
index 6e17eeb872..05d77c8450 100644
--- a/gr-vocoder/lib/codec2/postfilter.c
+++ b/gr-vocoder/lib/codec2/postfilter.c
@@ -1,13 +1,13 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: postfilter.c
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 13/09/09
-
+
Postfilter to improve sound quality for speech with high levels of
background noise. Unlike mixed-excitation models requires no bits
to be transmitted to handle background noise.
-
+
\*---------------------------------------------------------------------------*/
/*
@@ -71,7 +71,7 @@
This idea is rather experimental. Some potential problems that may
happen:
-
+
1/ If someone says "aaaaaaaahhhhhhhhh" will background estimator track
up to speech level? This would be a bad thing.
@@ -88,13 +88,13 @@
5/ Not sure what happens during long periods of voiced speech
e.g. "sshhhhhhh"
-
+
\*---------------------------------------------------------------------------*/
void postfilter(
MODEL *model,
float *bg_est
-)
+)
{
int m, uv;
float e;
diff --git a/gr-vocoder/lib/codec2/postfilter.h b/gr-vocoder/lib/codec2/postfilter.h
index bf080b1b65..156714e54f 100644
--- a/gr-vocoder/lib/codec2/postfilter.h
+++ b/gr-vocoder/lib/codec2/postfilter.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: postfilter.h
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 13/09/09
-
+
Postfilter header file.
-
+
\*---------------------------------------------------------------------------*/
/*
diff --git a/gr-vocoder/lib/codec2/quantise.c b/gr-vocoder/lib/codec2/quantise.c
index ff8d156b5b..c098030992 100644
--- a/gr-vocoder/lib/codec2/quantise.c
+++ b/gr-vocoder/lib/codec2/quantise.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: quantise.c
- AUTHOR......: David Rowe
- DATE CREATED: 31/5/92
-
- Quantisation functions for the sinusoidal coder.
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 31/5/92
+
+ Quantisation functions for the sinusoidal coder.
+
\*---------------------------------------------------------------------------*/
/*
@@ -41,16 +41,16 @@
#define LSP_DELTA1 0.01 /* grid spacing for LSP root searches */
/*---------------------------------------------------------------------------*\
-
+
FUNCTION HEADERS
\*---------------------------------------------------------------------------*/
-float speech_to_uq_lsps(float lsp[], float ak[], float Sn[], float w[],
+float speech_to_uq_lsps(float lsp[], float ak[], float Sn[], float w[],
int order);
/*---------------------------------------------------------------------------*\
-
+
FUNCTIONS
\*---------------------------------------------------------------------------*/
@@ -61,7 +61,7 @@ int lsp_bits(int i) {
#if VECTOR_QUANTISATION
/*---------------------------------------------------------------------------*\
-
+
quantise_uniform
Simulates uniform quantising of a float.
@@ -146,7 +146,7 @@ long quantise(const float * cb, float vec[], float w[], int k, int m, float *se)
}
/*---------------------------------------------------------------------------*\
-
+
lspd_quantise
Scalar lsp difference quantiser.
@@ -154,10 +154,10 @@ long quantise(const float * cb, float vec[], float w[], int k, int m, float *se)
\*---------------------------------------------------------------------------*/
void lspd_quantise(
- float lsp[],
+ float lsp[],
float lsp_[],
int order
-)
+)
{
int i,k,m;
float lsp_hz[LPC_MAX];
@@ -183,8 +183,8 @@ void lspd_quantise(
wt[0] = 1.0;
for(i=0; i<order; i++) {
- if (i)
- dlsp[i] = lsp_hz[i] - lsp__hz[i-1];
+ if (i)
+ dlsp[i] = lsp_hz[i] - lsp__hz[i-1];
else
dlsp[0] = lsp_hz[0];
@@ -194,14 +194,14 @@ void lspd_quantise(
indexes[i] = quantise(cb, &dlsp[i], wt, k, m, &se);
dlsp_[i] = cb[indexes[i]*k];
- if (i)
+ if (i)
lsp__hz[i] = lsp__hz[i-1] + dlsp_[i];
else
lsp__hz[0] = dlsp_[0];
}
for(; i<order; i++)
lsp__hz[i] = lsp__hz[i-1] + dlsp[i];
-
+
/* convert back to radians */
for(i=0; i<order; i++)
@@ -209,7 +209,7 @@ void lspd_quantise(
}
/*---------------------------------------------------------------------------*\
-
+
lspd_vq_quantise
Vector lsp difference quantiser.
@@ -217,10 +217,10 @@ void lspd_quantise(
\*---------------------------------------------------------------------------*/
void lspdvq_quantise(
- float lsp[],
+ float lsp[],
float lsp_[],
int order
-)
+)
{
int i,k,m,ncb, nlsp;
float dlsp[LPC_MAX];
@@ -243,8 +243,8 @@ void lspdvq_quantise(
/* scalar quantise dLSPs 1,2,3,4,5 */
for(i=0; i<5; i++) {
- if (i)
- dlsp[i] = (lsp[i] - lsp_[i-1])*4000.0/PI;
+ if (i)
+ dlsp[i] = (lsp[i] - lsp_[i-1])*4000.0/PI;
else
dlsp[0] = lsp[0]*4000.0/PI;
@@ -253,8 +253,8 @@ void lspdvq_quantise(
cb = lsp_cbdvq[i].cb;
index = quantise(cb, &dlsp[i], wt, k, m, &se);
dlsp_[i] = cb[index*k]*PI/4000.0;
-
- if (i)
+
+ if (i)
lsp_[i] = lsp_[i-1] + dlsp_[i];
else
lsp_[0] = dlsp_[0];
@@ -330,7 +330,7 @@ void force_min_lsp_dist(float lsp[], int lpc_order)
}
/*---------------------------------------------------------------------------*\
-
+
lpc_model_amplitudes
Derive a LPC model for amplitude samples then estimate amplitude samples
@@ -342,7 +342,7 @@ void force_min_lsp_dist(float lsp[], int lpc_order)
float lpc_model_amplitudes(
float Sn[], /* Input frame of speech samples */
- float w[],
+ float w[],
MODEL *model, /* sinusoidal model parameters */
int order, /* LPC model order */
int lsp_quant, /* optional LSP quantisation if non-zero */
@@ -353,7 +353,7 @@ float lpc_model_amplitudes(
float R[LPC_MAX+1];
float E;
int i,j;
- float snr;
+ float snr;
float lsp[LPC_MAX];
float lsp_hz[LPC_MAX];
float lsp_[LPC_MAX];
@@ -368,11 +368,11 @@ float lpc_model_amplitudes(
Wn[i] = Sn[i]*w[i];
autocorrelate(Wn,R,M,order);
levinson_durbin(R,ak,order);
-
+
E = 0.0;
for(i=0; i<=order; i++)
E += ak[i]*R[i];
-
+
for(i=0; i<order; i++)
wt[i] = 1.0;
@@ -386,7 +386,7 @@ float lpc_model_amplitudes(
for(i=0; i<order; i++)
lsp_hz[i] = (4000.0/PI)*lsp[i];
-
+
/* simple uniform scalar quantisers */
for(i=0; i<10; i++) {
@@ -396,7 +396,7 @@ float lpc_model_amplitudes(
index = quantise(cb, &lsp_hz[i], wt, k, m, &se);
lsp_hz[i] = cb[index*k];
}
-
+
/* experiment: simulating uniform quantisation error
for(i=0; i<order; i++)
lsp[i] += PI*(12.5/4000.0)*(1.0 - 2.0*(float)rand()/RAND_MAX);
@@ -428,7 +428,7 @@ float lpc_model_amplitudes(
lsp[i] = lsp[i-1] + PI*(75.0/4000.0);
}
- for(j=0; j<order; j++)
+ for(j=0; j<order; j++)
lsp_[j] = lsp[j];
lsp_to_lpc(lsp_, ak, order);
@@ -455,13 +455,13 @@ float lpc_model_amplitudes(
}
/*---------------------------------------------------------------------------*\
-
- aks_to_M2()
-
- Transforms the linear prediction coefficients to spectral amplitude
- samples. This function determines A(m) from the average energy per
- band using an FFT.
-
+
+ aks_to_M2()
+
+ Transforms the linear prediction coefficients to spectral amplitude
+ samples. This function determines A(m) from the average energy per
+ band using an FFT.
+
\*---------------------------------------------------------------------------*/
void aks_to_M2(
@@ -487,7 +487,7 @@ void aks_to_M2(
for(i=0; i<FFT_DEC; i++) {
Pw[i].real = 0.0;
- Pw[i].imag = 0.0;
+ Pw[i].imag = 0.0;
}
for(i=0; i<=order; i++)
@@ -499,7 +499,7 @@ void aks_to_M2(
for(i=0; i<FFT_DEC/2; i++)
Pw[i].real = E/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag);
#ifdef DUMP
- if (dump)
+ if (dump)
dump_Pw(Pw);
#endif
@@ -523,10 +523,10 @@ void aks_to_M2(
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: encode_Wo()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: encode_Wo()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Encodes Wo using a WO_LEVELS quantiser.
@@ -548,10 +548,10 @@ int encode_Wo(float Wo)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: decode_Wo()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: decode_Wo()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Decodes Wo using a WO_LEVELS quantiser.
@@ -571,10 +571,10 @@ float decode_Wo(int index)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: speech_to_uq_lsps()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: speech_to_uq_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Analyse a windowed frame of time domain speech to determine LPCs
which are the converted to LSPs for quantisation and transmission
@@ -584,7 +584,7 @@ float decode_Wo(int index)
float speech_to_uq_lsps(float lsp[],
float ak[],
- float Sn[],
+ float Sn[],
float w[],
int order
)
@@ -598,19 +598,19 @@ float speech_to_uq_lsps(float lsp[],
Wn[i] = Sn[i]*w[i];
autocorrelate(Wn, R, M, order);
levinson_durbin(R, ak, order);
-
+
E = 0.0;
for(i=0; i<=order; i++)
E += ak[i]*R[i];
-
+
roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
if (roots != order) {
/* for some reason LSP roots could not be found */
/* some alpha testers are reporting this condition */
fprintf(stderr, "LSP roots not found!\nroots = %d\n", roots);
for(i=0; i<=order; i++)
- fprintf(stderr, "a[%d] = %f\n", i, ak[i]);
-
+ fprintf(stderr, "a[%d] = %f\n", i, ak[i]);
+
/* some benign LSP values we can use instead */
for(i=0; i<order; i++)
lsp[i] = (PI/order)*(float)i;
@@ -620,10 +620,10 @@ float speech_to_uq_lsps(float lsp[],
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: encode_lsps()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: encode_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
From a vector of unquantised (floating point) LSPs finds the quantised
LSP indexes.
@@ -643,7 +643,7 @@ void encode_lsps(int indexes[], float lsp[], int order)
for(i=0; i<order; i++)
lsp_hz[i] = (4000.0/PI)*lsp[i];
-
+
/* simple uniform scalar quantisers */
wt[0] = 1.0;
@@ -656,10 +656,10 @@ void encode_lsps(int indexes[], float lsp[], int order)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: decode_lsps()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: decode_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
From a vector of quantised LSP indexes, returns the quantised
(floating point) LSPs.
@@ -685,10 +685,10 @@ void decode_lsps(float lsp[], int indexes[], int order)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: bw_expand_lsps()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: bw_expand_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Applies Bandwidth Expansion (BW) to a vector of LSPs. Prevents any
two LSPs getting too close together after quantisation. We know
@@ -724,10 +724,10 @@ void bw_expand_lsps(float lsp[],
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: apply_lpc_correction()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: apply_lpc_correction()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Apply first harmonic LPC correction at decoder. This helps improve
low pitch males after LPC modelling, like hts1a and morig.
@@ -742,10 +742,10 @@ void apply_lpc_correction(MODEL *model)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: encode_energy()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: encode_energy()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Encodes LPC energy using an E_LEVELS quantiser.
@@ -768,10 +768,10 @@ int encode_energy(float e)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: decode_energy()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: decode_energy()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Decodes energy using a WO_BITS quantiser.
@@ -792,10 +792,10 @@ float decode_energy(int index)
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: encode_amplitudes()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: encode_amplitudes()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Time domain LPC is used model the amplitudes which are then
converted to LSPs and quantised. So we don't actually encode the
@@ -804,10 +804,10 @@ float decode_energy(int index)
\*---------------------------------------------------------------------------*/
-void encode_amplitudes(int lsp_indexes[],
+void encode_amplitudes(int lsp_indexes[],
int *energy_index,
- MODEL *model,
- float Sn[],
+ MODEL *model,
+ float Sn[],
float w[])
{
float lsps[LPC_ORD];
@@ -820,19 +820,19 @@ void encode_amplitudes(int lsp_indexes[],
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: decode_amplitudes()
- AUTHOR......: David Rowe
- DATE CREATED: 22/8/2010
+
+ FUNCTION....: decode_amplitudes()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
Given the amplitude quantiser indexes recovers the harmonic
amplitudes.
\*---------------------------------------------------------------------------*/
-float decode_amplitudes(MODEL *model,
+float decode_amplitudes(MODEL *model,
float ak[],
- int lsp_indexes[],
+ int lsp_indexes[],
int energy_index,
float lsps[],
float *e
@@ -844,7 +844,7 @@ float decode_amplitudes(MODEL *model,
bw_expand_lsps(lsps, LPC_ORD);
lsp_to_lpc(lsps, ak, LPC_ORD);
*e = decode_energy(energy_index);
- aks_to_M2(ak, LPC_ORD, model, *e, &snr, 1);
+ aks_to_M2(ak, LPC_ORD, model, *e, &snr, 1);
apply_lpc_correction(model);
return snr;
diff --git a/gr-vocoder/lib/codec2/quantise.h b/gr-vocoder/lib/codec2/quantise.h
index 90a3661ffd..42b05f43bd 100644
--- a/gr-vocoder/lib/codec2/quantise.h
+++ b/gr-vocoder/lib/codec2/quantise.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: quantise.h
- AUTHOR......: David Rowe
- DATE CREATED: 31/5/92
-
- Quantisation functions for the sinusoidal coder.
-
+ AUTHOR......: David Rowe
+ DATE CREATED: 31/5/92
+
+ Quantisation functions for the sinusoidal coder.
+
\*---------------------------------------------------------------------------*/
/*
@@ -36,7 +36,7 @@
void quantise_init();
float lpc_model_amplitudes(float Sn[], float w[], MODEL *model, int order,
int lsp,float ak[]);
-void aks_to_M2(float ak[], int order, MODEL *model, float E, float *snr,
+void aks_to_M2(float ak[], int order, MODEL *model, float E, float *snr,
int dump);
int encode_Wo(float Wo);
@@ -45,15 +45,15 @@ float decode_Wo(int index);
void encode_lsps(int indexes[], float lsp[], int order);
void decode_lsps(float lsp[], int indexes[], int order);
void lspd_quantise(float lsp[], float lsp_[], int order);
-void lspdvq_quantise(float lsp[], float lsp_[], int order);
+void lspdvq_quantise(float lsp[], float lsp_[], int order);
int encode_energy(float e);
float decode_energy(int index);
-void encode_amplitudes(int lsp_indexes[],
+void encode_amplitudes(int lsp_indexes[],
int *energy_index,
- MODEL *model,
- float Sn[],
+ MODEL *model,
+ float Sn[],
float w[]);
float decode_amplitudes(MODEL *model,
@@ -71,7 +71,7 @@ int lsp_bits(int i);
void apply_lpc_correction(MODEL *model);
float speech_to_uq_lsps(float lsp[],
float ak[],
- float Sn[],
+ float Sn[],
float w[],
int order
);
diff --git a/gr-vocoder/lib/codec2/sine.c b/gr-vocoder/lib/codec2/sine.c
index 45cc9de718..b30f9abad6 100644
--- a/gr-vocoder/lib/codec2/sine.c
+++ b/gr-vocoder/lib/codec2/sine.c
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: sine.c
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 19/8/2010
-
+
Sinusoidal analysis and synthesis functions.
-
+
\*---------------------------------------------------------------------------*/
/*
@@ -26,9 +26,9 @@
*/
/*---------------------------------------------------------------------------*\
-
- INCLUDES
-
+
+ INCLUDES
+
\*---------------------------------------------------------------------------*/
#include <stdlib.h>
@@ -42,25 +42,25 @@
#define HPF_BETA 0.125
/*---------------------------------------------------------------------------*\
-
- HEADERS
-
+
+ HEADERS
+
\*---------------------------------------------------------------------------*/
-void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
+void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
float pstep);
/*---------------------------------------------------------------------------*\
-
- FUNCTIONS
-
+
+ FUNCTIONS
+
\*---------------------------------------------------------------------------*/
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: make_analysis_window
- AUTHOR......: David Rowe
- DATE CREATED: 11/5/94
+
+ FUNCTION....: make_analysis_window
+ AUTHOR......: David Rowe
+ DATE CREATED: 11/5/94
Init function that generates the time domain analysis window and it's DFT.
@@ -72,15 +72,15 @@ void make_analysis_window(float w[],COMP W[])
COMP temp;
int i,j;
- /*
+ /*
Generate Hamming window centered on M-sample pitch analysis window
-
+
0 M/2 M-1
|-------------|-------------|
|-------|-------|
NW samples
- All our analysis/synthsis is centred on the M/2 sample.
+ All our analysis/synthsis is centred on the M/2 sample.
*/
m = 0.0;
@@ -92,7 +92,7 @@ void make_analysis_window(float w[],COMP W[])
}
for(i=M/2+NW/2; i<M; i++)
w[i] = 0.0;
-
+
/* Normalise - makes freq domain amplitude estimation straight
forward */
@@ -101,7 +101,7 @@ void make_analysis_window(float w[],COMP W[])
w[i] *= m;
}
- /*
+ /*
Generate DFT of analysis window, used for later processing. Note
we modulo FFT_ENC shift the time domain window w[], this makes the
imaginary part of the DFT W[] equal to zero as the shifted w[] is
@@ -112,13 +112,13 @@ void make_analysis_window(float w[],COMP W[])
|-------------------------|
----\ /----
- \ /
+ \ /
\ / <- shifted version of window w[n]
\ /
\ /
-------
- |---------| |---------|
+ |---------| |---------|
NW/2 NW/2
*/
@@ -133,8 +133,8 @@ void make_analysis_window(float w[],COMP W[])
fft(&W[0].real,FFT_ENC,-1); /* "Numerical Recipes in C" FFT */
- /*
- Re-arrange W[] to be symmetrical about FFT_ENC/2. Makes later
+ /*
+ Re-arrange W[] to be symmetrical about FFT_ENC/2. Makes later
analysis convenient.
Before:
@@ -142,21 +142,21 @@ void make_analysis_window(float w[],COMP W[])
0 FFT_ENC-1
|----------|---------|
- __ _
- \ /
- \_______________/
+ __ _
+ \ /
+ \_______________/
After:
0 FFT_ENC-1
|----------|---------|
- ___
- / \
- ________/ \_______
+ ___
+ / \
+ ________/ \_______
*/
-
-
+
+
for(i=0; i<FFT_ENC/2; i++) {
temp.real = W[i].real;
temp.imag = W[i].imag;
@@ -169,15 +169,15 @@ void make_analysis_window(float w[],COMP W[])
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: hpf
- AUTHOR......: David Rowe
+
+ FUNCTION....: hpf
+ AUTHOR......: David Rowe
DATE CREATED: 16 Nov 2010
High pass filter with a -3dB point of about 160Hz.
y(n) = -HPF_BETA*y(n-1) + x(n) - x(n-1)
-
+
\*---------------------------------------------------------------------------*/
float hpf(float x, float states[])
@@ -189,10 +189,10 @@ float hpf(float x, float states[])
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: dft_speech
- AUTHOR......: David Rowe
- DATE CREATED: 27/5/94
+
+ FUNCTION....: dft_speech
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
Finds the DFT of the current speech input speech frame.
@@ -201,7 +201,7 @@ float hpf(float x, float states[])
void dft_speech(COMP Sw[], float Sn[], float w[])
{
int i;
-
+
for(i=0; i<FFT_ENC; i++) {
Sw[i].real = 0.0;
Sw[i].imag = 0.0;
@@ -209,7 +209,7 @@ void dft_speech(COMP Sw[], float Sn[], float w[])
/* Centre analysis window on time axis, we need to arrange input
to FFT this way to make FFT phases correct */
-
+
/* move 2nd half to start of FFT input vector */
for(i=0; i<NW/2; i++)
@@ -224,10 +224,10 @@ void dft_speech(COMP Sw[], float Sn[], float w[])
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: two_stage_pitch_refinement
+
+ FUNCTION....: two_stage_pitch_refinement
AUTHOR......: David Rowe
- DATE CREATED: 27/5/94
+ DATE CREATED: 27/5/94
Refines the current pitch estimate using the harmonic sum pitch
estimation technique.
@@ -236,7 +236,7 @@ void dft_speech(COMP Sw[], float Sn[], float w[])
void two_stage_pitch_refinement(MODEL *model, COMP Sw[])
{
- float pmin,pmax,pstep; /* pitch refinment minimum, maximum and step */
+ float pmin,pmax,pstep; /* pitch refinment minimum, maximum and step */
/* Coarse refinement */
@@ -244,16 +244,16 @@ void two_stage_pitch_refinement(MODEL *model, COMP Sw[])
pmin = TWO_PI/model->Wo - 5;
pstep = 1.0;
hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
-
+
/* Fine refinement */
-
+
pmax = TWO_PI/model->Wo + 1;
pmin = TWO_PI/model->Wo - 1;
pstep = 0.25;
hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
-
+
/* Limit range */
-
+
if (model->Wo < TWO_PI/P_MAX)
model->Wo = TWO_PI/P_MAX;
if (model->Wo > TWO_PI/P_MIN)
@@ -263,20 +263,20 @@ void two_stage_pitch_refinement(MODEL *model, COMP Sw[])
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: hs_pitch_refinement
- AUTHOR......: David Rowe
- DATE CREATED: 27/5/94
-
- Harmonic sum pitch refinement function.
-
- pmin pitch search range minimum
- pmax pitch search range maximum
- step pitch search step size
- model current pitch estimate in model.Wo
-
- model refined pitch estimate in model.Wo
-
+
+ FUNCTION....: hs_pitch_refinement
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Harmonic sum pitch refinement function.
+
+ pmin pitch search range minimum
+ pmax pitch search range maximum
+ step pitch search step size
+ model current pitch estimate in model.Wo
+
+ model refined pitch estimate in model.Wo
+
\*---------------------------------------------------------------------------*/
void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float pstep)
@@ -289,14 +289,14 @@ void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float
float Em; /* mamimum energy */
float r; /* number of rads/bin */
float p; /* current pitch */
-
+
/* Initialisation */
-
+
model->L = PI/model->Wo; /* use initial pitch est. for L */
Wom = model->Wo;
Em = 0.0;
r = TWO_PI/FFT_ENC;
-
+
/* Determine harmonic sum for a range of Wo values */
for(p=pmin; p<=pmax; p+=pstep) {
@@ -308,10 +308,10 @@ void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float
for(m=1; m<=model->L; m++) {
b = floor(m*Wo/r + 0.5);
E += Sw[b].real*Sw[b].real + Sw[b].imag*Sw[b].imag;
- }
+ }
/* Compare to see if this is a maximum */
-
+
if (E > Em) {
Em = E;
Wom = Wo;
@@ -322,13 +322,13 @@ void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: estimate_amplitudes
- AUTHOR......: David Rowe
- DATE CREATED: 27/5/94
-
- Estimates the complex amplitudes of the harmonics.
-
+
+ FUNCTION....: estimate_amplitudes
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Estimates the complex amplitudes of the harmonics.
+
\*---------------------------------------------------------------------------*/
void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[])
@@ -369,14 +369,14 @@ void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[])
}
/*---------------------------------------------------------------------------*\
-
- est_voicing_mbe()
-
+
+ est_voicing_mbe()
+
Returns the error of the MBE cost function for a fiven F0.
Note: I think a lot of the operations below can be simplified as
W[].imag = 0 and has been normalised such that den always equals 1.
-
+
\*---------------------------------------------------------------------------*/
float est_voicing_mbe(
@@ -393,7 +393,7 @@ float est_voicing_mbe(
int offset; /* centers Hw[] about current harmonic */
float den; /* denominator of Am expression */
float error; /* accumulated error between original and synthesised */
- float Wo;
+ float Wo;
float sig, snr;
float elow, ehigh, eratio;
float dF0, sixty;
@@ -445,16 +445,16 @@ float est_voicing_mbe(
error += Ew[m].imag*Ew[m].imag;
}
}
-
+
snr = 10.0*log10(sig/error);
if (snr > V_THRESH)
model->voiced = 1;
else
model->voiced = 0;
-
+
/* post processing, helps clean up some voicing errors ------------------*/
- /*
+ /*
Determine the ratio of low freancy to high frequency energy,
voiced speech tends to be dominated by low frequency energy,
unvoiced by high frequency. This measure can be used to
@@ -488,7 +488,7 @@ float est_voicing_mbe(
/* If pitch is jumping about it's likely this is UV */
dF0 = (model->Wo - prev_Wo)*FS/TWO_PI;
- if (fabs(dF0) > 15.0)
+ if (fabs(dF0) > 15.0)
model->voiced = 0;
/* A common source of Type 2 errors is the pitch estimator
@@ -507,10 +507,10 @@ float est_voicing_mbe(
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: make_synthesis_window
- AUTHOR......: David Rowe
- DATE CREATED: 11/5/94
+
+ FUNCTION....: make_synthesis_window
+ AUTHOR......: David Rowe
+ DATE CREATED: 11/5/94
Init function that generates the trapezoidal (Parzen) sythesis window.
@@ -539,15 +539,15 @@ void make_synthesis_window(float Pn[])
}
/*---------------------------------------------------------------------------*\
-
- FUNCTION....: synthesise
- AUTHOR......: David Rowe
- DATE CREATED: 20/2/95
-
+
+ FUNCTION....: synthesise
+ AUTHOR......: David Rowe
+ DATE CREATED: 20/2/95
+
Synthesise a speech signal in the frequency domain from the
sinusodal model parameters. Uses overlap-add with a trapezoidal
window to smoothly interpolate betwen frames.
-
+
\*---------------------------------------------------------------------------*/
void synthesise(
@@ -579,7 +579,7 @@ void synthesise(
gives better results for synthesis frames greater than 10ms. Inverse
FFT synthesis using a 512 pt FFT works well for 10ms window. I think
(but am not sure) that the problem is realted to the quantisation of
- the harmonic frequencies to the FFT bin size, e.g. there is a
+ the harmonic frequencies to the FFT bin size, e.g. there is a
8000/512 Hz step between FFT bins. For some reason this makes
the speech from longer frame > 10ms sound poor. The effect can also
be seen when synthesising test signals like single sine waves, some
@@ -619,7 +619,7 @@ void synthesise(
}
for(i=N-1,j=0; i<2*N; i++,j++)
Sw_[j].real += 2.0*model->A[l]*cos(j*model->Wo*l + model->phi[l]);
- }
+ }
#endif
/* Overlap add to previous samples */
diff --git a/gr-vocoder/lib/codec2/sine.h b/gr-vocoder/lib/codec2/sine.h
index ae578bf70b..f223e2afca 100644
--- a/gr-vocoder/lib/codec2/sine.h
+++ b/gr-vocoder/lib/codec2/sine.h
@@ -1,11 +1,11 @@
/*---------------------------------------------------------------------------*\
-
+
FILE........: sine.h
- AUTHOR......: David Rowe
+ AUTHOR......: David Rowe
DATE CREATED: 1/11/94
-
+
Header file for sinusoidal analysis and synthesis functions.
-
+
\*---------------------------------------------------------------------------*/
/*
@@ -36,7 +36,7 @@ float hpf(float x, float states[]);
void dft_speech(COMP Sw[], float Sn[], float w[]);
void two_stage_pitch_refinement(MODEL *model, COMP Sw[]);
void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[]);
-float est_voicing_mbe(MODEL *model, COMP Sw[], COMP W[], COMP Sw_[],COMP Ew[],
+float est_voicing_mbe(MODEL *model, COMP Sw[], COMP W[], COMP Sw_[],COMP Ew[],
float prev_Wo);
void make_synthesis_window(float Pn[]);
void synthesise(float Sn_[], MODEL *model, float Pn[], int shift);
--
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