AudioResampler: Minor test code fixes
Test: resampler_tests
Bug: 123920996
Change-Id: I11aa8f883854ebc492e230bc48311009d5a8920f
diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index eeeecce..52936cc 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp
@@ -243,7 +243,7 @@
{
// compute the normalized transition bandwidth
const double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
- const double halfbw = tbw / 2.;
+ const double halfbw = tbw * 0.5;
double fcr; // compute fcr, the 3 dB amplitude cut-off.
if (inSampleRate < outSampleRate) { // upsample
@@ -290,7 +290,7 @@
#if 0
// Keep this debug code in case an app causes resampler design issues.
- const double halfbw = tbw / 2.;
+ const double halfbw = tbw * 0.5;
// print basic filter stats
ALOGD("L:%d hnc:%d stopBandAtten:%lf fcr:%lf atten:%lf tbw:%lf\n",
c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, attenuation, tbw);
@@ -305,7 +305,7 @@
const int32_t passSteps = 1000;
- testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.ML /*stopSteps*/,
+ testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.mL /*stopSteps*/,
passMin, passMax, passRipple, stopMax, stopRipple);
ALOGD("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
ALOGD("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
diff --git a/media/libaudioprocessing/AudioResamplerFirGen.h b/media/libaudioprocessing/AudioResamplerFirGen.h
index 39cafeb..100baa3 100644
--- a/media/libaudioprocessing/AudioResamplerFirGen.h
+++ b/media/libaudioprocessing/AudioResamplerFirGen.h
@@ -547,8 +547,8 @@
wstart += wstep;
}
// renormalize - this is needed for integer filter types, use 1 for float or double.
- constexpr int64_t integralShift = std::is_integral<T>::value ? (sizeof(T) * 8 - 1) : 0;
- const double norm = 1. / (L << integralShift);
+ constexpr int integralShift = std::is_integral<T>::value ? (sizeof(T) * CHAR_BIT - 1) : 0;
+ const double norm = 1. / (int64_t{L} << integralShift);
firMin = fmin * norm;
firMax = fmax * norm;
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index 0c8e5bb..d990111 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -5,8 +5,8 @@
header_libs: ["libbase_headers"],
shared_libs: [
- "libaudioutils",
"libaudioprocessing",
+ "libaudioutils",
"libcutils",
"liblog",
"libutils",