By convention const goes before the type specifier (54c3b664) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioFlinger.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -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);

services/audioflinger/AudioMixer.cpp

+11 −10

Original line number	Diff line number	Diff line
		@@ -604,7 +604,7 @@ 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;
		@@ -643,7 +643,7 @@ 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]);
		@@ -681,7 +681,7 @@ 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]);
		@@ -694,7 +694,7 @@ 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
		@@ -916,6 +916,7 @@ 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;

		@@ -996,7 +997,7 @@ 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.
		@@ -1012,7 +1013,7 @@ 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;
		@@ -1023,7 +1024,7 @@ 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;
		@@ -1053,12 +1054,12 @@ 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;
		@@ -1066,7 +1067,7 @@ 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) {

services/audioflinger/AudioMixer.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -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;

services/audioflinger/AudioResamplerSinc.cpp

+7 −7

Original line number	Diff line number	Diff line
		@@ -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 {

services/audioflinger/AudioResamplerSinc.h

+4 −4

Original line number	Diff line number	Diff line
		@@ -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[];