Mercurial > libavcodec.hg
changeset 10003:24952f1a8979 libavcodec
cosmetics: K&R coding style
| author | diego |
|---|---|
| date | Sun, 02 Aug 2009 10:34:30 +0000 |
| parents | 878d449793e1 |
| children | 15c5048b9a49 |
| files | celp_filters.c celp_filters.h |
| diffstat | 2 files changed, 54 insertions(+), 68 deletions(-) [+] |
line wrap: on
line diff
--- a/celp_filters.c Sun Aug 02 08:47:02 2009 +0000 +++ b/celp_filters.c Sun Aug 02 10:34:30 2009 +0000 @@ -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 @@ /* 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 @@ 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]; } }
--- a/celp_filters.h Sun Aug 02 08:47:02 2009 +0000 +++ b/celp_filters.h Sun Aug 02 10:34:30 2009 +0000 @@ -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 @@ * * 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 @@ * * 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 @@ * * 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 */