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Searched
defs:sP
(Results
1 - 3
of
3
) sorted by null
/frameworks/av/media/libaudioprocessing/
AudioResamplerFirProcess.h
184
const TI*
sP
,
200
// for (int j = 0; j < CHANNELS; ++j) mac(accum[j], c,
sP
+ j);
201
const TI *tmp_data =
sP
; // tmp_ptr seems to work better
205
sP
-= CHANNELS;
222
mac(l, r, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP),
sP
);
224
sP
-= CHANNELS;
234
mac(l, TFUNC::interpolatep(coefsP[0], coefsP[count], lerpP),
sP
);
236
sP
-= CHANNELS;
255
const TI*
sP
,
259
ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN,
sP
, sN, 0, volumeLR)
[
all
...]
AudioResamplerSinc.cpp
423
int16_t const*
sP
= samples;
432
interpolate<CHANNELS>(l, r, coefsP++, offset, lerpP,
sP
);
433
sP
-= CHANNELS;
444
sP
-= CHANNELS*3;
463
sampleP = vld1_s16(
sP
);
469
sP
-= 4;
512
sP
-= CHANNELS*3;
533
sampleP = vld2_s16(
sP
);
539
sP
-= 8;
/external/eclipse-basebuilder/basebuilder-3.6.2/org.eclipse.releng.basebuilder/plugins/org.apache.ant_1.7.1.v20090120-1145/lib/
ant-jdepend.jar
Completed in 6556 milliseconds