Merge commit 'remotes/korg/cupcake'

Conflicts:
	core/java/com/android/internal/app/AlertController.java
	core/res/res/values/strings.xml
	media/java/android/media/AudioSystem.java
	services/java/com/android/server/LocationManagerService.java

1 files changed, 324 insertions, 26 deletions

diff --git a/libs/audioflinger/AudioResampler.cpp b/libs/audioflinger/AudioResampler.cpp
index c93ead37d9..5dabacbb70 100644
--- a/libs/audioflinger/AudioResampler.cpp
+++ b/libs/audioflinger/AudioResampler.cpp
@@ -14,17 +14,23 @@
  * limitations under the License.
  */
 
+#define LOG_TAG "AudioResampler"
+//#define LOG_NDEBUG 0
+
 #include <stdint.h>
 #include <stdlib.h>
 #include <sys/types.h>
 #include <cutils/log.h>
 #include <cutils/properties.h>
-
 #include "AudioResampler.h"
 #include "AudioResamplerSinc.h"
 #include "AudioResamplerCubic.h"
 
 namespace android {
+
+#ifdef __ARM_ARCH_5E__  // optimized asm option
+    #define ASM_ARM_RESAMP1 // enable asm optimisation for ResamplerOrder1
+#endif // __ARM_ARCH_5E__
 // ----------------------------------------------------------------------------
 
 class AudioResamplerOrder1 : public AudioResampler {
@@ -46,6 +52,15 @@ private:
             AudioBufferProvider* provider);
     void resampleStereo16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+    void AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement);
+    void AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement);
+#endif  // ASM_ARM_RESAMP1
+
     static inline int32_t Interp(int32_t x0, int32_t x1, uint32_t f) {
         return x0 + (((x1 - x0) * (int32_t)(f >> kPreInterpShift)) >> kNumInterpBits);
     }
@@ -73,20 +88,23 @@ AudioResampler* AudioResampler::create(int bitDepth, int inChannelCount,
 
     if (quality == DEFAULT)
         quality = LOW_QUALITY;
-    
+
     switch (quality) {
     default:
     case LOW_QUALITY:
+        LOGV("Create linear Resampler");
         resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
         break;
     case MED_QUALITY:
+        LOGV("Create cubic Resampler");
         resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
         break;
     case HIGH_QUALITY:
+        LOGV("Create sinc Resampler");
         resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
         break;
     }
-    
+
     // initialize resampler
     resampler->init();
     return resampler;
@@ -103,10 +121,10 @@ AudioResampler::AudioResampler(int bitDepth, int inChannelCount,
                 inChannelCount);
         // LOG_ASSERT(0);
     }
-    
+
     // initialize common members
     mVolume[0] = mVolume[1] = 0;
-    mBuffer.raw = NULL;
+    mBuffer.frameCount = 0;
 
     // save format for quick lookup
     if (inChannelCount == 1) {
@@ -160,19 +178,31 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
+    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
     // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
-    //		outFrameCount, inputIndex, phaseFraction, phaseIncrement);
+    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
 
     while (outputIndex < outputSampleCount) {
 
         // buffer is empty, fetch a new one
-        if (mBuffer.raw == NULL) {
+        while (mBuffer.frameCount == 0) {
+            mBuffer.frameCount = inFrameCount;
             provider->getNextBuffer(&mBuffer);
-            if (mBuffer.raw == NULL)
-                break;
+            if (mBuffer.raw == NULL) {
+                goto resampleStereo16_exit;
+            }
+
             // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+            if (mBuffer.frameCount > inputIndex) break;
+
+            inputIndex -= mBuffer.frameCount;
+            mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
+            mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
+            provider->releaseBuffer(&mBuffer);
+             // mBuffer.frameCount == 0 now so we reload a new buffer
         }
+
         int16_t *in = mBuffer.i16;
 
         // handle boundary case
@@ -187,34 +217,47 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
 
         // process input samples
         // LOGE("general case\n");
-        while (outputIndex < outputSampleCount) {
+
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+        if (inputIndex + 2 < mBuffer.frameCount) {
+            int32_t* maxOutPt;
+            int32_t maxInIdx;
+
+            maxOutPt = out + (outputSampleCount - 2);   // 2 because 2 frames per loop
+            maxInIdx = mBuffer.frameCount - 2;
+            AsmStereo16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
+                    phaseFraction, phaseIncrement);
+        }
+#endif  // ASM_ARM_RESAMP1
+
+        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
             out[outputIndex++] += vl * Interp(in[inputIndex*2-2],
                     in[inputIndex*2], phaseFraction);
             out[outputIndex++] += vr * Interp(in[inputIndex*2-1],
                     in[inputIndex*2+1], phaseFraction);
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
-            if (inputIndex >= mBuffer.frameCount)
-                break;
         }
+
         // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
         // if done with buffer, save samples
         if (inputIndex >= mBuffer.frameCount) {
             inputIndex -= mBuffer.frameCount;
 
-            // LOGE("buffer done, new input index", inputIndex);
+            // LOGE("buffer done, new input index %d", inputIndex);
 
             mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
             mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
             provider->releaseBuffer(&mBuffer);
 
-            // verify that the releaseBuffer NULLS the buffer pointer 
-            // LOG_ASSERT(mBuffer.raw == NULL);
+            // verify that the releaseBuffer resets the buffer frameCount
+            // LOG_ASSERT(mBuffer.frameCount == 0);
         }
     }
 
     // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
+resampleStereo16_exit:
     // save state
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
@@ -231,18 +274,27 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
+    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
     // LOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
-
     while (outputIndex < outputSampleCount) {
-
         // buffer is empty, fetch a new one
-        if (mBuffer.raw == NULL) {
+        while (mBuffer.frameCount == 0) {
+            mBuffer.frameCount = inFrameCount;
             provider->getNextBuffer(&mBuffer);
-            if (mBuffer.raw == NULL)
-                break;
+            if (mBuffer.raw == NULL) {
+                mInputIndex = inputIndex;
+                mPhaseFraction = phaseFraction;
+                goto resampleMono16_exit;
+            }
             // LOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+            if (mBuffer.frameCount >  inputIndex) break;
+
+            inputIndex -= mBuffer.frameCount;
+            mX0L = mBuffer.i16[mBuffer.frameCount-1];
+            provider->releaseBuffer(&mBuffer);
+            // mBuffer.frameCount == 0 now so we reload a new buffer
         }
         int16_t *in = mBuffer.i16;
 
@@ -259,38 +311,284 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
 
         // process input samples
         // LOGE("general case\n");
-        while (outputIndex < outputSampleCount) {
+
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+        if (inputIndex + 2 < mBuffer.frameCount) {
+            int32_t* maxOutPt;
+            int32_t maxInIdx;
+
+            maxOutPt = out + (outputSampleCount - 2);
+            maxInIdx = (int32_t)mBuffer.frameCount - 2;
+                AsmMono16Loop(in, maxOutPt, maxInIdx, outputIndex, out, inputIndex, vl, vr,
+                        phaseFraction, phaseIncrement);
+        }
+#endif  // ASM_ARM_RESAMP1
+
+        while (outputIndex < outputSampleCount && inputIndex < mBuffer.frameCount) {
             int32_t sample = Interp(in[inputIndex-1], in[inputIndex],
                     phaseFraction);
             out[outputIndex++] += vl * sample;
             out[outputIndex++] += vr * sample;
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
-            if (inputIndex >= mBuffer.frameCount)
-                break;
         }
+
+
         // LOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
         // if done with buffer, save samples
         if (inputIndex >= mBuffer.frameCount) {
             inputIndex -= mBuffer.frameCount;
 
-            // LOGE("buffer done, new input index", inputIndex);
+            // LOGE("buffer done, new input index %d", inputIndex);
 
             mX0L = mBuffer.i16[mBuffer.frameCount-1];
             provider->releaseBuffer(&mBuffer);
 
-            // verify that the releaseBuffer NULLS the buffer pointer 
-            // LOG_ASSERT(mBuffer.raw == NULL);
+            // verify that the releaseBuffer resets the buffer frameCount
+            // LOG_ASSERT(mBuffer.frameCount == 0);
         }
     }
 
     // LOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
 
+resampleMono16_exit:
     // save state
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
 }
 
+#ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
+
+/*******************************************************************
+*
+*   AsmMono16Loop
+*   asm optimized monotonic loop version; one loop is 2 frames
+*   Input:
+*       in : pointer on input samples
+*       maxOutPt : pointer on first not filled
+*       maxInIdx : index on first not used
+*       outputIndex : pointer on current output index
+*       out : pointer on output buffer
+*       inputIndex : pointer on current input index
+*       vl, vr : left and right gain
+*       phaseFraction : pointer on current phase fraction
+*       phaseIncrement
+*   Ouput:
+*       outputIndex :
+*       out : updated buffer
+*       inputIndex : index of next to use
+*       phaseFraction : phase fraction for next interpolation
+*
+*******************************************************************/
+void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement)
+{
+#define MO_PARAM5   "36"        // offset of parameter 5 (outputIndex)
+
+    asm(
+        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, lr}\n"
+        // get parameters
+        "   ldr r6, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
+        "   ldr r6, [r6]\n"                         // phaseFraction
+        "   ldr r7, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
+        "   ldr r7, [r7]\n"                         // inputIndex
+        "   ldr r8, [sp, #" MO_PARAM5 " + 4]\n"     // out
+        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
+        "   ldr r0, [r0]\n"                         // outputIndex
+        "   add r8, r0, asl #2\n"                   // curOut
+        "   ldr r9, [sp, #" MO_PARAM5 " + 24]\n"    // phaseIncrement
+        "   ldr r10, [sp, #" MO_PARAM5 " + 12]\n"   // vl
+        "   ldr r11, [sp, #" MO_PARAM5 " + 16]\n"   // vr
+
+        // r0 pin, x0, Samp
+
+        // r1 in
+        // r2 maxOutPt
+        // r3 maxInIdx
+
+        // r4 x1, i1, i3, Out1
+        // r5 out0
+
+        // r6 frac
+        // r7 inputIndex
+        // r8 curOut
+
+        // r9 inc
+        // r10 vl
+        // r11 vr
+
+        // r12
+        // r13 sp
+        // r14
+
+        // the following loop works on 2 frames
+
+        ".Y4L01:\n"
+        "   cmp r8, r2\n"                   // curOut - maxCurOut
+        "   bcs .Y4L02\n"
+
+#define MO_ONE_FRAME \
+    "   add r0, r1, r7, asl #1\n"       /* in + inputIndex */\
+    "   ldrsh r4, [r0]\n"               /* in[inputIndex] */\
+    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
+    "   ldrsh r0, [r0, #-2]\n"          /* in[inputIndex-1] */\
+    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
+    "   sub r4, r4, r0\n"               /* in[inputIndex] - in[inputIndex-1] */\
+    "   mov r4, r4, lsl #2\n"           /* <<2 */\
+    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
+    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
+    "   add r0, r0, r4\n"               /* x0 - (..) */\
+    "   mla r5, r0, r10, r5\n"          /* vl*interp + out[] */\
+    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
+    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
+    "   mla r4, r0, r11, r4\n"          /* vr*interp + out[] */\
+    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */\
+    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */
+
+        MO_ONE_FRAME    // frame 1
+        MO_ONE_FRAME    // frame 2
+
+        "   cmp r7, r3\n"                   // inputIndex - maxInIdx
+        "   bcc .Y4L01\n"
+        ".Y4L02:\n"
+
+        "   bic r6, r6, #0xC0000000\n"             // phaseFraction & ...
+        // save modified values
+        "   ldr r0, [sp, #" MO_PARAM5 " + 20]\n"    // &phaseFraction
+        "   str r6, [r0]\n"                         // phaseFraction
+        "   ldr r0, [sp, #" MO_PARAM5 " + 8]\n"     // &inputIndex
+        "   str r7, [r0]\n"                         // inputIndex
+        "   ldr r0, [sp, #" MO_PARAM5 " + 4]\n"     // out
+        "   sub r8, r0\n"                           // curOut - out
+        "   asr r8, #2\n"                           // new outputIndex
+        "   ldr r0, [sp, #" MO_PARAM5 " + 0]\n"     // &outputIndex
+        "   str r8, [r0]\n"                         // save outputIndex
+
+        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, pc}\n"
+    );
+}
+
+/*******************************************************************
+*
+*   AsmStereo16Loop
+*   asm optimized stereo loop version; one loop is 2 frames
+*   Input:
+*       in : pointer on input samples
+*       maxOutPt : pointer on first not filled
+*       maxInIdx : index on first not used
+*       outputIndex : pointer on current output index
+*       out : pointer on output buffer
+*       inputIndex : pointer on current input index
+*       vl, vr : left and right gain
+*       phaseFraction : pointer on current phase fraction
+*       phaseIncrement
+*   Ouput:
+*       outputIndex :
+*       out : updated buffer
+*       inputIndex : index of next to use
+*       phaseFraction : phase fraction for next interpolation
+*
+*******************************************************************/
+void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
+            size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
+            uint32_t &phaseFraction, uint32_t phaseIncrement)
+{
+#define ST_PARAM5    "40"     // offset of parameter 5 (outputIndex)
+    asm(
+        "stmfd  sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, lr}\n"
+        // get parameters
+        "   ldr r6, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
+        "   ldr r6, [r6]\n"                         // phaseFraction
+        "   ldr r7, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
+        "   ldr r7, [r7]\n"                         // inputIndex
+        "   ldr r8, [sp, #" ST_PARAM5 " + 4]\n"     // out
+        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
+        "   ldr r0, [r0]\n"                         // outputIndex
+        "   add r8, r0, asl #2\n"                   // curOut
+        "   ldr r9, [sp, #" ST_PARAM5 " + 24]\n"    // phaseIncrement
+        "   ldr r10, [sp, #" ST_PARAM5 " + 12]\n"   // vl
+        "   ldr r11, [sp, #" ST_PARAM5 " + 16]\n"   // vr
+
+        // r0 pin, x0, Samp
+
+        // r1 in
+        // r2 maxOutPt
+        // r3 maxInIdx
+
+        // r4 x1, i1, i3, out1
+        // r5 out0
+
+        // r6 frac
+        // r7 inputIndex
+        // r8 curOut
+
+        // r9 inc
+        // r10 vl
+        // r11 vr
+
+        // r12 temporary
+        // r13 sp
+        // r14
+
+        ".Y5L01:\n"
+        "   cmp r8, r2\n"                   // curOut - maxCurOut
+        "   bcs .Y5L02\n"
+
+#define ST_ONE_FRAME \
+    "   bic r6, r6, #0xC0000000\n"      /* phaseFraction & ... */\
+\
+    "   add r0, r1, r7, asl #2\n"       /* in + 2*inputIndex */\
+\
+    "   ldrsh r4, [r0]\n"               /* in[2*inputIndex] */\
+    "   ldr r5, [r8]\n"                 /* out[outputIndex] */\
+    "   ldrsh r12, [r0, #-4]\n"         /* in[2*inputIndex-2] */\
+    "   sub r4, r4, r12\n"              /* in[2*InputIndex] - in[2*InputIndex-2] */\
+    "   mov r4, r4, lsl #2\n"           /* <<2 */\
+    "   smulwt r4, r4, r6\n"            /* (x1-x0)*.. */\
+    "   add r12, r12, r4\n"             /* x0 - (..) */\
+    "   mla r5, r12, r10, r5\n"         /* vl*interp + out[] */\
+    "   ldr r4, [r8, #4]\n"             /* out[outputIndex+1] */\
+    "   str r5, [r8], #4\n"             /* out[outputIndex++] = ... */\
+\
+    "   ldrsh r12, [r0, #+2]\n"         /* in[2*inputIndex+1] */\
+    "   ldrsh r0, [r0, #-2]\n"          /* in[2*inputIndex-1] */\
+    "   sub r12, r12, r0\n"             /* in[2*InputIndex] - in[2*InputIndex-2] */\
+    "   mov r12, r12, lsl #2\n"         /* <<2 */\
+    "   smulwt r12, r12, r6\n"          /* (x1-x0)*.. */\
+    "   add r12, r0, r12\n"             /* x0 - (..) */\
+    "   mla r4, r12, r11, r4\n"         /* vr*interp + out[] */\
+    "   str r4, [r8], #4\n"             /* out[outputIndex++] = ... */\
+\
+    "   add r6, r6, r9\n"               /* phaseFraction + phaseIncrement */\
+    "   add r7, r7, r6, lsr #30\n"      /* inputIndex + phaseFraction>>30 */
+
+    ST_ONE_FRAME    // frame 1
+    ST_ONE_FRAME    // frame 1
+
+        "   cmp r7, r3\n"                       // inputIndex - maxInIdx
+        "   bcc .Y5L01\n"
+        ".Y5L02:\n"
+
+        "   bic r6, r6, #0xC0000000\n"              // phaseFraction & ...
+        // save modified values
+        "   ldr r0, [sp, #" ST_PARAM5 " + 20]\n"    // &phaseFraction
+        "   str r6, [r0]\n"                         // phaseFraction
+        "   ldr r0, [sp, #" ST_PARAM5 " + 8]\n"     // &inputIndex
+        "   str r7, [r0]\n"                         // inputIndex
+        "   ldr r0, [sp, #" ST_PARAM5 " + 4]\n"     // out
+        "   sub r8, r0\n"                           // curOut - out
+        "   asr r8, #2\n"                           // new outputIndex
+        "   ldr r0, [sp, #" ST_PARAM5 " + 0]\n"     // &outputIndex
+        "   str r8, [r0]\n"                         // save outputIndex
+
+        "   ldmfd   sp!, {r4, r5, r6, r7, r8, r9, r10, r11, r12, pc}\n"
+    );
+}
+
+#endif  // ASM_ARM_RESAMP1
+
+
 // ----------------------------------------------------------------------------
 }
 ; // namespace android