Merge "By convention const goes before the type specifier"

5 files changed, 24 insertions, 23 deletions

diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index cf8d495..ff8dedb 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -63,7 +63,7 @@ class AudioFlinger :
 {
     friend class BinderService<AudioFlinger>;
 public:
-    static char const* getServiceName() { return "media.audio_flinger"; }
+    static const char* getServiceName() { return "media.audio_flinger"; }
 
     virtual     status_t    dump(int fd, const Vector<String16>& args);
 
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index bc5cb9b..8df4605 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -601,7 +601,7 @@ void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32
 
 void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
 {
-    int16_t const *in = static_cast<int16_t const *>(t->in);
+    const int16_t *in = static_cast<const int16_t *>(t->in);
 
     if (CC_UNLIKELY(aux != NULL)) {
         int32_t l;
@@ -640,7 +640,7 @@ void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount
             const uint32_t vrl = t->volumeRL;
             const int16_t va = (int16_t)t->auxLevel;
             do {
-                uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
                 in += 2;
                 out[0] = mulAddRL(1, rl, vrl, out[0]);
@@ -678,7 +678,7 @@ void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount
         else {
             const uint32_t vrl = t->volumeRL;
             do {
-                uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
                 out[0] = mulAddRL(1, rl, vrl, out[0]);
                 out[1] = mulAddRL(0, rl, vrl, out[1]);
@@ -691,7 +691,7 @@ void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount
 
 void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
 {
-    int16_t const *in = static_cast<int16_t const *>(t->in);
+    const int16_t *in = static_cast<int16_t const *>(t->in);
 
     if (CC_UNLIKELY(aux != NULL)) {
         // ramp gain
@@ -913,6 +913,7 @@ void AudioMixer::process__genericNoResampling(state_t* state)
 // generic code with resampling
 void AudioMixer::process__genericResampling(state_t* state)
 {
+    // this const just means that local variable outTemp doesn't change
     int32_t* const outTemp = state->outputTemp;
     const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
 
@@ -993,7 +994,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
     while (numFrames) {
         b.frameCount = numFrames;
         t.bufferProvider->getNextBuffer(&b);
-        int16_t const *in = b.i16;
+        const int16_t *in = b.i16;
 
         // in == NULL can happen if the track was flushed just after having
         // been enabled for mixing.
@@ -1009,7 +1010,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
             // volume is boosted, so we might need to clamp even though
             // we process only one track.
             do {
-                uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
                 int32_t l = mulRL(1, rl, vrl) >> 12;
                 int32_t r = mulRL(0, rl, vrl) >> 12;
@@ -1020,7 +1021,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
             } while (--outFrames);
         } else {
             do {
-                uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
                 in += 2;
                 int32_t l = mulRL(1, rl, vrl) >> 12;
                 int32_t r = mulRL(0, rl, vrl) >> 12;
@@ -1050,12 +1051,12 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
     const track_t& t1 = state->tracks[i];
     AudioBufferProvider::Buffer& b1(t1.buffer);
 
-    int16_t const *in0;
+    const int16_t *in0;
     const int16_t vl0 = t0.volume[0];
     const int16_t vr0 = t0.volume[1];
     size_t frameCount0 = 0;
 
-    int16_t const *in1;
+    const int16_t *in1;
     const int16_t vl1 = t1.volume[0];
     const int16_t vr1 = t1.volume[1];
     size_t frameCount1 = 0;
@@ -1063,7 +1064,7 @@ void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
     //FIXME: only works if two tracks use same buffer
     int32_t* out = t0.mainBuffer;
     size_t numFrames = state->frameCount;
-    int16_t const *buff = NULL;
+    const int16_t *buff = NULL;
 
 
     while (numFrames) {
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 4ba6845..84f6330 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -145,7 +145,7 @@ private:
         mutable AudioBufferProvider::Buffer buffer;
 
         hook_t      hook;
-        void const* in;             // current location in buffer
+        const void* in;             // current location in buffer
 
         AudioResampler*     resampler;
         uint32_t            sampleRate;
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 {
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index e6cb90b..0e1bc44 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -44,22 +44,22 @@ private:
 
     template<int CHANNELS>
     inline void 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);
 
     template<int CHANNELS>
     inline void 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);
 
     template<int CHANNELS>
     inline void read(int16_t*& impulse, uint32_t& phaseFraction,
-            int16_t const* in, size_t inputIndex);
+            const int16_t* in, size_t inputIndex);
 
     int16_t *mState;
     int16_t *mImpulse;
     int16_t *mRingFull;
 
-    int32_t const * mFirCoefs;
+    const int32_t * mFirCoefs;
     static const int32_t mFirCoefsDown[];
     static const int32_t mFirCoefsUp[];