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audio_coding: Replaced macro WEBRTC_SPL_RSHIFT_W16 with >>

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Replaced trivial shift macro with >>. The actual implementation of the macro is simply >>.

Affected codecs:
* ilbc
* isac/fix

BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7396 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org
2014-10-08 15:36:30 +00:00
parent 575d126a3d
commit f71785cd3b
16 changed files with 47 additions and 58 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
View File

@ -100,7 +100,7 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
enh_buf+ENH_BUFL-inLen, /* Input samples */
(int16_t)(inLen+ENH_BUFL_FILTEROVERHEAD),
downsampled,
(int16_t)WEBRTC_SPL_RSHIFT_W16(inLen, 1),
(int16_t)(inLen / 2),
(int16_t*)WebRtcIlbcfix_kLpFiltCoefs, /* Coefficients in Q12 */
FILTERORDER_DS_PLUS1, /* Length of filter (order-1) */
FACTOR_DS,
@ -292,7 +292,7 @@ int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
/* Calculate increase parameter for window part (16 last samples) */
/* (1-2*SqrtEnChange)/16 in Q15 */
inc=(2048-WEBRTC_SPL_RSHIFT_W16(SqrtEnChange, 3));
inc = 2048 - (SqrtEnChange >> 3);
win=0;
tmpW16ptr=&plc_pred[plc_blockl-16];

4
webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
View File

@ -41,7 +41,7 @@ void WebRtcIlbcfix_GetCbVec(
base_size=lMem-cbveclen+1;
if (cbveclen==SUBL) {
base_size+=WEBRTC_SPL_RSHIFT_W16(cbveclen,1);
base_size += cbveclen / 2;
}
/* No filter -> First codebook section */
@ -60,7 +60,7 @@ void WebRtcIlbcfix_GetCbVec(
k=(int16_t)WEBRTC_SPL_MUL_16_16(2, (index-(lMem-cbveclen+1)))+cbveclen;
lag=WEBRTC_SPL_RSHIFT_W16(k, 1);
lag = k / 2;
WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec);

2
webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
View File

@ -43,7 +43,7 @@ void WebRtcIlbcfix_Lsf2Lsp(
Lower 8 bits give the difference, which needs
to be approximated linearly
*/
k = WEBRTC_SPL_RSHIFT_W16(freq, 8);
k = freq >> 8;
diff = (freq&0x00ff);
/* Guard against getting outside table */

2
webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
View File

@ -91,7 +91,7 @@ void WebRtcIlbcfix_Poly2Lsp(
/* Run 4 times to reduce the interval */
for (i = 0; i < 4; i++) {
/* xmid =(xlow + xhigh)/2 */
xmid = WEBRTC_SPL_RSHIFT_W16(xlow, 1) + WEBRTC_SPL_RSHIFT_W16(xhigh, 1);
xmid = (xlow >> 1) + (xhigh >> 1);
ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]);
if (WEBRTC_SPL_MUL_16_16(ylow, ymid) <= 0) {

4
webrtc/modules/audio_coding/codecs/ilbc/refiner.c
View File

@ -53,7 +53,7 @@ void WebRtcIlbcfix_Refiner(
/* defining array bounds */
estSegPosRounded=WEBRTC_SPL_RSHIFT_W16((estSegPos - 2),2);
estSegPosRounded = (estSegPos - 2) >> 2;
searchSegStartPos=estSegPosRounded-ENH_SLOP;
@ -103,7 +103,7 @@ void WebRtcIlbcfix_Refiner(
bounds */
*updStartPos = (int16_t)WEBRTC_SPL_MUL_16_16(searchSegStartPos,4) + tloc + 4;
tloc2 = WEBRTC_SPL_RSHIFT_W16((tloc+3), 2);
tloc2 = (tloc + 3) >> 2;
st=searchSegStartPos+tloc2-ENH_FL0;

2
webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
View File

@ -37,7 +37,7 @@ int32_t WebRtcIlbcfix_Smooth_odata(
errs=0;
for(i=0;i<80;i++) {
err=(int16_t)WEBRTC_SPL_RSHIFT_W16((psseq[i]-odata[i]), 3);
err = (psseq[i] - odata[i]) >> 3;
errs+=WEBRTC_SPL_MUL_16_16(err, err); /* errs in Q-6 */
}