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| author | Glenn Kasten <gkasten@google.com> | 2012-01-06 07:46:30 -0800 |
|---|---|---|
| committer | Glenn Kasten <gkasten@google.com> | 2012-01-06 08:00:59 -0800 |
| commit | 54c3b66444ebfb9f2265ee70ac3b76ccefa0506a (patch) | |
| tree | 2ae6ae86501101399d639f2a1227742c120e7ac8 /services/audioflinger/AudioResamplerSinc.cpp | |
| parent | a2a0a5d7d56baa831870f4bf2a0d942a477d92ef (diff) | |
| download | frameworks_av-54c3b66444ebfb9f2265ee70ac3b76ccefa0506a.zip frameworks_av-54c3b66444ebfb9f2265ee70ac3b76ccefa0506a.tar.gz frameworks_av-54c3b66444ebfb9f2265ee70ac3b76ccefa0506a.tar.bz2 | |
By convention const goes before the type specifier
Change-Id: I70203abd6a6f54e5bd9f1412800cc01212157e58
Diffstat (limited to 'services/audioflinger/AudioResamplerSinc.cpp')
| -rw-r--r-- | services/audioflinger/AudioResamplerSinc.cpp | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp index 9e5e254..d012433 100644 --- a/services/audioflinger/AudioResamplerSinc.cpp +++ b/services/audioflinger/AudioResamplerSinc.cpp @@ -284,7 +284,7 @@ template<int CHANNELS> **/ void AudioResamplerSinc::read( int16_t*& impulse, uint32_t& phaseFraction, - int16_t const* in, size_t inputIndex) + const int16_t* in, size_t inputIndex) { const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits; impulse += CHANNELS; @@ -302,7 +302,7 @@ void AudioResamplerSinc::read( template<int CHANNELS> void AudioResamplerSinc::filterCoefficient( - int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples) + int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples) { // compute the index of the coefficient on the positive side and // negative side @@ -317,9 +317,9 @@ void AudioResamplerSinc::filterCoefficient( l = 0; r = 0; - int32_t const* coefs = mFirCoefs; - int16_t const *sP = samples; - int16_t const *sN = samples+CHANNELS; + const int32_t* coefs = mFirCoefs; + const int16_t *sP = samples; + const int16_t *sN = samples+CHANNELS; for (unsigned int i=0 ; i<halfNumCoefs/4 ; i++) { interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP); interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN); @@ -339,13 +339,13 @@ void AudioResamplerSinc::filterCoefficient( template<int CHANNELS> void AudioResamplerSinc::interpolate( int32_t& l, int32_t& r, - int32_t const* coefs, int16_t lerp, int16_t const* samples) + const int32_t* coefs, int16_t lerp, const int16_t* samples) { int32_t c0 = coefs[0]; int32_t c1 = coefs[1]; int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0); if (CHANNELS == 2) { - uint32_t rl = *reinterpret_cast<uint32_t const*>(samples); + uint32_t rl = *reinterpret_cast<const uint32_t*>(samples); l = mulAddRL(1, rl, sinc, l); r = mulAddRL(0, rl, sinc, r); } else { |