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-rw-r--r--gr-vocoder/lib/codec2/interp.c86
1 files changed, 43 insertions, 43 deletions
diff --git a/gr-vocoder/lib/codec2/interp.c b/gr-vocoder/lib/codec2/interp.c
index 257c4a81c..64372b050 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);
}
generated by cgit (git 2.34.1) at 2024-12-22 18:49:36 +0000