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cosmetics: K&R coding style

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Originally committed as revision 19561 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Diego Biurrun 2009-08-02 10:34:30 +00:00
parent 6b620372a8
commit 954489244e
2 changed files with 54 additions and 68 deletions
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76
libavcodec/celp_filters.c
View File

@ -25,11 +25,10 @@
#include "avcodec.h"
#include "celp_filters.h"
void ff_celp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int len)
void ff_celp_convolve_circ(int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int len)
{
int i, k;
@ -37,44 +36,39 @@ void ff_celp_convolve_circ(
/* Since there are few pulses over an entire subframe (i.e. almost
all fc_in[i] are zero) it is faster to loop over fc_in first. */
for(i=0; i<len; i++)
{
if(fc_in[i])
{
for(k=0; k<i; k++)
for (i = 0; i < len; i++) {
if (fc_in[i]) {
for (k = 0; k < i; k++)
fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15;
for(k=i; k<len; k++)
for (k = i; k < len; k++)
fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15;
}
}
}
int ff_celp_lp_synthesis_filter(
int16_t *out,
const int16_t* filter_coeffs,
const int16_t* in,
int buffer_length,
int filter_length,
int stop_on_overflow,
int rounder)
int ff_celp_lp_synthesis_filter(int16_t *out,
const int16_t* filter_coeffs,
const int16_t* in,
int buffer_length,
int filter_length,
int stop_on_overflow,
int rounder)
{
int i,n;
// Avoids a +1 in the inner loop.
filter_length++;
for(n=0; n<buffer_length; n++)
{
for (n = 0; n < buffer_length; n++) {
int sum = rounder;
for(i=1; i<filter_length; i++)
for (i = 1; i < filter_length; i++)
sum -= filter_coeffs[i-1] * out[n-i];
sum = (sum >> 12) + in[n];
if(sum + 0x8000 > 0xFFFFU)
{
if(stop_on_overflow)
if (sum + 0x8000 > 0xFFFFU) {
if (stop_on_overflow)
return 1;
sum = (sum >> 31) ^ 32767;
}
@ -84,42 +78,38 @@ int ff_celp_lp_synthesis_filter(
return 0;
}
void ff_celp_lp_synthesis_filterf(
float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length)
void ff_celp_lp_synthesis_filterf(float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length)
{
int i,n;
// Avoids a +1 in the inner loop.
filter_length++;
for(n=0; n<buffer_length; n++)
{
for (n = 0; n < buffer_length; n++) {
out[n] = in[n];
for(i=1; i<filter_length; i++)
for (i = 1; i < filter_length; i++)
out[n] -= filter_coeffs[i-1] * out[n-i];
}
}
void ff_celp_lp_zero_synthesis_filterf(
float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length)
void ff_celp_lp_zero_synthesis_filterf(float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length)
{
int i,n;
// Avoids a +1 in the inner loop.
filter_length++;
for(n=0; n<buffer_length; n++)
{
for (n = 0; n < buffer_length; n++) {
out[n] = in[n];
for(i=1; i<filter_length; i++)
for (i = 1; i < filter_length; i++)
out[n] -= filter_coeffs[i-1] * in[n-i];
}
}

46
libavcodec/celp_filters.h
View File

@ -36,11 +36,10 @@
*
* \note fc_in and fc_out should not overlap!
*/
void ff_celp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int len);
void ff_celp_convolve_circ(int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int len);
/**
* LP synthesis filter.
@ -60,14 +59,13 @@ void ff_celp_convolve_circ(
*
* Routine applies 1/A(z) filter to given speech data.
*/
int ff_celp_lp_synthesis_filter(
int16_t *out,
const int16_t* filter_coeffs,
const int16_t* in,
int buffer_length,
int filter_length,
int stop_on_overflow,
int rounder);
int ff_celp_lp_synthesis_filter(int16_t *out,
const int16_t* filter_coeffs,
const int16_t* in,
int buffer_length,
int filter_length,
int stop_on_overflow,
int rounder);
/**
* LP synthesis filter.
@ -84,12 +82,11 @@ int ff_celp_lp_synthesis_filter(
*
* Routine applies 1/A(z) filter to given speech data.
*/
void ff_celp_lp_synthesis_filterf(
float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length);
void ff_celp_lp_synthesis_filterf(float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length);
/**
* LP zero synthesis filter.
@ -106,11 +103,10 @@ void ff_celp_lp_synthesis_filterf(
*
* Routine applies A(z) filter to given speech data.
*/
void ff_celp_lp_zero_synthesis_filterf(
float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length);
void ff_celp_lp_zero_synthesis_filterf(float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length);
#endif /* AVCODEC_CELP_FILTERS_H */