diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 495bad0..1505f08 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -623,7 +623,7 @@
                 } else {
                     int64_t currentTime = mBufferTimestamps.top();
                     currentTime += mStreamInfo->aacSamplesPerFrame *
-                            1000000ll / mStreamInfo->sampleRate;
+                            1000000ll / mStreamInfo->aacSampleRate;
                     mBufferTimestamps.add(currentTime);
                 }
             } else {
@@ -874,7 +874,7 @@
                         // adjust/interpolate next time stamp
                         *currentBufLeft -= decodedSize;
                         *nextTimeStamp += mStreamInfo->aacSamplesPerFrame *
-                                1000000ll / mStreamInfo->sampleRate;
+                                1000000ll / mStreamInfo->aacSampleRate;
                         ALOGV("adjusted nextTimeStamp/size to %lld/%d",
                                 (long long) *nextTimeStamp, *currentBufLeft);
                     } else {
diff --git a/media/libstagefright/omx/Android.mk b/media/libstagefright/omx/Android.mk
index be8cf46..07ea605 100644
--- a/media/libstagefright/omx/Android.mk
+++ b/media/libstagefright/omx/Android.mk
@@ -1,10 +1,6 @@
 LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 
-ifeq ($(TARGET_DEVICE), manta)
-    LOCAL_CFLAGS += -DSURFACE_IS_BGR32
-endif
-
 LOCAL_SRC_FILES:=                     \
         FrameDropper.cpp              \
         GraphicBufferSource.cpp       \
diff --git a/services/audioflinger/AudioResamplerFirOps.h b/services/audioflinger/AudioResamplerFirOps.h
index 563c22f..658285d 100644
--- a/services/audioflinger/AudioResamplerFirOps.h
+++ b/services/audioflinger/AudioResamplerFirOps.h
@@ -25,7 +25,7 @@
 #define USE_INLINE_ASSEMBLY (false)
 #endif
 
-#if USE_INLINE_ASSEMBLY && defined(__ARM_NEON__)
+#if defined(__aarch64__) || defined(__ARM_NEON__)
 #define USE_NEON (true)
 #include <arm_neon.h>
 #else
diff --git a/services/audioflinger/AudioResamplerFirProcess.h b/services/audioflinger/AudioResamplerFirProcess.h
index 8a615a7..176202e 100644
--- a/services/audioflinger/AudioResamplerFirProcess.h
+++ b/services/audioflinger/AudioResamplerFirProcess.h
@@ -243,6 +243,9 @@
     }
 }
 
+/* Calculates a single output frame from a polyphase resampling filter.
+ * See Process() for parameter details.
+ */
 template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO>
 static inline
 void ProcessL(TO* const out,
@@ -256,6 +259,39 @@
     ProcessBase<CHANNELS, STRIDE, InterpNull>(out, count, coefsP, coefsN, sP, sN, 0, volumeLR);
 }
 
+/*
+ * Calculates a single output frame from a polyphase resampling filter,
+ * with filter phase interpolation.
+ *
+ * @param out should point to the output buffer with space for at least one output frame.
+ *
+ * @param count should be half the size of the total filter length (halfNumCoefs), as we
+ * use symmetry in filter coefficients to evaluate two dot products.
+ *
+ * @param coefsP is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
+ * to the positive sP.
+ *
+ * @param coefsN is one phase of the polyphase filter bank of size halfNumCoefs, corresponding
+ * to the negative sN.
+ *
+ * @param coefsP1 is the next phase of coefsP (used for interpolation).
+ *
+ * @param coefsN1 is the next phase of coefsN (used for interpolation).
+ *
+ * @param sP is the positive half of the coefficients (as viewed by a convolution),
+ * starting at the original samples pointer and decrementing (by CHANNELS).
+ *
+ * @param sN is the negative half of the samples (as viewed by a convolution),
+ * starting at the original samples pointer + CHANNELS and incrementing (by CHANNELS).
+ *
+ * @param lerpP The fractional siting between the polyphase indices is given by the bits
+ * below coefShift. See fir() for details.
+ *
+ * @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
+ * expressed as a S32 integer or float.  A negative value inverts the channel 180 degrees.
+ * The pointer volumeLR should be aligned to a minimum of 8 bytes.
+ * A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
+ */
 template <int CHANNELS, int STRIDE, typename TC, typename TI, typename TO, typename TINTERP>
 static inline
 void Process(TO* const out,
@@ -274,7 +310,7 @@
 }
 
 /*
- * Calculates a single output frame (two samples) from input sample pointer.
+ * Calculates a single output frame from input sample pointer.
  *
  * This sets up the params for the accelerated Process() and ProcessL()
  * functions to do the appropriate dot products.
@@ -309,7 +345,7 @@
  * the positive half of the filter is dot product from samples to samples-halfNumCoefs+1.
  *
  * @param volumeLR is a pointer to an array of two 32 bit volume values, one per stereo channel,
- * expressed as a S32 integer.  A negative value inverts the channel 180 degrees.
+ * expressed as a S32 integer or float.  A negative value inverts the channel 180 degrees.
  * The pointer volumeLR should be aligned to a minimum of 8 bytes.
  * A typical value for volume is 0x1000 to align to a unity gain output of 20.12.
  *
diff --git a/services/audioflinger/AudioResamplerFirProcessNeon.h b/services/audioflinger/AudioResamplerFirProcessNeon.h
index 107175b..3de9edd 100644
--- a/services/audioflinger/AudioResamplerFirProcessNeon.h
+++ b/services/audioflinger/AudioResamplerFirProcessNeon.h
@@ -22,10 +22,35 @@
 // depends on AudioResamplerFirOps.h, AudioResamplerFirProcess.h
 
 #if USE_NEON
+
+// use intrinsics if inline arm32 assembly is not possible
+#if !USE_INLINE_ASSEMBLY
+#define USE_INTRINSIC
+#endif
+
+// following intrinsics available only on ARM 64 bit ACLE
+#ifndef __aarch64__
+#undef vld1q_f32_x2
+#undef vld1q_s32_x2
+#endif
+
+#define TO_STRING2(x) #x
+#define TO_STRING(x) TO_STRING2(x)
+// uncomment to print GCC version, may be relevant for intrinsic optimizations
+/* #pragma message ("GCC version: " TO_STRING(__GNUC__) \
+        "." TO_STRING(__GNUC_MINOR__) \
+        "." TO_STRING(__GNUC_PATCHLEVEL__)) */
+
 //
-// NEON specializations are enabled for Process() and ProcessL()
+// NEON specializations are enabled for Process() and ProcessL() in AudioResamplerFirProcess.h
+//
+// Two variants are presented here:
+// ARM NEON inline assembly which appears up to 10-15% faster than intrinsics (gcc 4.9) for arm32.
+// ARM NEON intrinsics which can also be used by arm64 and x86/64 with NEON header.
+//
 
 // Macros to save a mono/stereo accumulator sample in q0 (and q4) as stereo out.
+// These are only used for inline assembly.
 #define ASSEMBLY_ACCUMULATE_MONO \
         "vld1.s32       {d2}, [%[vLR]:64]        \n"/* (1) load volumes */\
         "vld1.s32       {d3}, %[out]             \n"/* (2) unaligned load the output */\
@@ -45,6 +70,458 @@
         "vqadd.s32      d3, d3, d0               \n"/* (1+4d) accumulate result (saturating)*/\
         "vst1.s32       {d3}, %[out]             \n"/* (2+2d)store result*/
 
+template <int CHANNELS, int STRIDE, bool FIXED>
+static inline void ProcessNeonIntrinsic(int32_t* out,
+        int count,
+        const int16_t* coefsP,
+        const int16_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* volumeLR,
+        uint32_t lerpP,
+        const int16_t* coefsP1,
+        const int16_t* coefsN1)
+{
+    ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
+    COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS == 1 || CHANNELS == 2);
+
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    coefsP = (const int16_t*)__builtin_assume_aligned(coefsP, 16);
+    coefsN = (const int16_t*)__builtin_assume_aligned(coefsN, 16);
+
+    int16x4_t interp;
+    if (!FIXED) {
+        interp = vdup_n_s16(lerpP);
+        //interp = (int16x4_t)vset_lane_s32 ((int32x2_t)lerpP, interp, 0);
+        coefsP1 = (const int16_t*)__builtin_assume_aligned(coefsP1, 16);
+        coefsN1 = (const int16_t*)__builtin_assume_aligned(coefsN1, 16);
+    }
+    int32x4_t accum, accum2;
+    // warning uninitialized if we use veorq_s32
+    // (alternative to below) accum = veorq_s32(accum, accum);
+    accum = vdupq_n_s32(0);
+    if (CHANNELS == 2) {
+        // (alternative to below) accum2 = veorq_s32(accum2, accum2);
+        accum2 = vdupq_n_s32(0);
+    }
+    do {
+        int16x8_t posCoef = vld1q_s16(coefsP);
+        coefsP += 8;
+        int16x8_t negCoef = vld1q_s16(coefsN);
+        coefsN += 8;
+        if (!FIXED) { // interpolate
+            int16x8_t posCoef1 = vld1q_s16(coefsP1);
+            coefsP1 += 8;
+            int16x8_t negCoef1 = vld1q_s16(coefsN1);
+            coefsN1 += 8;
+
+            posCoef1 = vsubq_s16(posCoef1, posCoef);
+            negCoef = vsubq_s16(negCoef, negCoef1);
+
+            posCoef1 = vqrdmulhq_lane_s16(posCoef1, interp, 0);
+            negCoef = vqrdmulhq_lane_s16(negCoef, interp, 0);
+
+            posCoef = vaddq_s16(posCoef, posCoef1);
+            negCoef = vaddq_s16(negCoef, negCoef1);
+        }
+        switch (CHANNELS) {
+        case 1: {
+            int16x8_t posSamp = vld1q_s16(sP);
+            int16x8_t negSamp = vld1q_s16(sN);
+            sN += 8;
+            posSamp = vrev64q_s16(posSamp);
+
+            // dot product
+            accum = vmlal_s16(accum, vget_low_s16(posSamp), vget_high_s16(posCoef)); // reversed
+            accum = vmlal_s16(accum, vget_high_s16(posSamp), vget_low_s16(posCoef)); // reversed
+            accum = vmlal_s16(accum, vget_low_s16(negSamp), vget_low_s16(negCoef));
+            accum = vmlal_s16(accum, vget_high_s16(negSamp), vget_high_s16(negCoef));
+            sP -= 8;
+        } break;
+        case 2: {
+            int16x8x2_t posSamp = vld2q_s16(sP);
+            int16x8x2_t negSamp = vld2q_s16(sN);
+            sN += 16;
+            posSamp.val[0] = vrev64q_s16(posSamp.val[0]);
+            posSamp.val[1] = vrev64q_s16(posSamp.val[1]);
+
+            // dot product
+            accum = vmlal_s16(accum, vget_low_s16(posSamp.val[0]), vget_high_s16(posCoef)); // r
+            accum = vmlal_s16(accum, vget_high_s16(posSamp.val[0]), vget_low_s16(posCoef)); // r
+            accum2 = vmlal_s16(accum2, vget_low_s16(posSamp.val[1]), vget_high_s16(posCoef)); // r
+            accum2 = vmlal_s16(accum2, vget_high_s16(posSamp.val[1]), vget_low_s16(posCoef)); // r
+            accum = vmlal_s16(accum, vget_low_s16(negSamp.val[0]), vget_low_s16(negCoef));
+            accum = vmlal_s16(accum, vget_high_s16(negSamp.val[0]), vget_high_s16(negCoef));
+            accum2 = vmlal_s16(accum2, vget_low_s16(negSamp.val[1]), vget_low_s16(negCoef));
+            accum2 = vmlal_s16(accum2, vget_high_s16(negSamp.val[1]), vget_high_s16(negCoef));
+            sP -= 16;
+        }
+        } break;
+    } while (count -= 8);
+
+    // multiply by volume and save
+    volumeLR = (const int32_t*)__builtin_assume_aligned(volumeLR, 8);
+    int32x2_t vLR = vld1_s32(volumeLR);
+    int32x2_t outSamp = vld1_s32(out);
+    // combine and funnel down accumulator
+    int32x2_t outAccum = vpadd_s32(vget_low_s32(accum), vget_high_s32(accum));
+    if (CHANNELS == 1) {
+        // duplicate accum to both L and R
+        outAccum = vpadd_s32(outAccum, outAccum);
+    } else if (CHANNELS == 2) {
+        // accum2 contains R, fold in
+        int32x2_t outAccum2 = vpadd_s32(vget_low_s32(accum2), vget_high_s32(accum2));
+        outAccum = vpadd_s32(outAccum, outAccum2);
+    }
+    outAccum = vqrdmulh_s32(outAccum, vLR);
+    outSamp = vqadd_s32(outSamp, outAccum);
+    vst1_s32(out, outSamp);
+}
+
+template <int CHANNELS, int STRIDE, bool FIXED>
+static inline void ProcessNeonIntrinsic(int32_t* out,
+        int count,
+        const int32_t* coefsP,
+        const int32_t* coefsN,
+        const int16_t* sP,
+        const int16_t* sN,
+        const int32_t* volumeLR,
+        uint32_t lerpP,
+        const int32_t* coefsP1,
+        const int32_t* coefsN1)
+{
+    ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
+    COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS == 1 || CHANNELS == 2);
+
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    coefsP = (const int32_t*)__builtin_assume_aligned(coefsP, 16);
+    coefsN = (const int32_t*)__builtin_assume_aligned(coefsN, 16);
+
+    int32x2_t interp;
+    if (!FIXED) {
+        interp = vdup_n_s32(lerpP);
+        coefsP1 = (const int32_t*)__builtin_assume_aligned(coefsP1, 16);
+        coefsN1 = (const int32_t*)__builtin_assume_aligned(coefsN1, 16);
+    }
+    int32x4_t accum, accum2;
+    // warning uninitialized if we use veorq_s32
+    // (alternative to below) accum = veorq_s32(accum, accum);
+    accum = vdupq_n_s32(0);
+    if (CHANNELS == 2) {
+        // (alternative to below) accum2 = veorq_s32(accum2, accum2);
+        accum2 = vdupq_n_s32(0);
+    }
+    do {
+#ifdef vld1q_s32_x2
+        int32x4x2_t posCoef = vld1q_s32_x2(coefsP);
+        coefsP += 8;
+        int32x4x2_t negCoef = vld1q_s32_x2(coefsN);
+        coefsN += 8;
+#else
+        int32x4x2_t posCoef;
+        posCoef.val[0] = vld1q_s32(coefsP);
+        coefsP += 4;
+        posCoef.val[1] = vld1q_s32(coefsP);
+        coefsP += 4;
+        int32x4x2_t negCoef;
+        negCoef.val[0] = vld1q_s32(coefsN);
+        coefsN += 4;
+        negCoef.val[1] = vld1q_s32(coefsN);
+        coefsN += 4;
+#endif
+        if (!FIXED) { // interpolate
+#ifdef vld1q_s32_x2
+            int32x4x2_t posCoef1 = vld1q_s32_x2(coefsP1);
+            coefsP1 += 8;
+            int32x4x2_t negCoef1 = vld1q_s32_x2(coefsN1);
+            coefsN1 += 8;
+#else
+            int32x4x2_t posCoef1;
+            posCoef1.val[0] = vld1q_s32(coefsP1);
+            coefsP1 += 4;
+            posCoef1.val[1] = vld1q_s32(coefsP1);
+            coefsP1 += 4;
+            int32x4x2_t negCoef1;
+            negCoef1.val[0] = vld1q_s32(coefsN1);
+            coefsN1 += 4;
+            negCoef1.val[1] = vld1q_s32(coefsN1);
+            coefsN1 += 4;
+#endif
+
+            posCoef1.val[0] = vsubq_s32(posCoef1.val[0], posCoef.val[0]);
+            posCoef1.val[1] = vsubq_s32(posCoef1.val[1], posCoef.val[1]);
+            negCoef.val[0] = vsubq_s32(negCoef.val[0], negCoef1.val[0]);
+            negCoef.val[1] = vsubq_s32(negCoef.val[1], negCoef1.val[1]);
+
+            posCoef1.val[0] = vqrdmulhq_lane_s32(posCoef1.val[0], interp, 0);
+            posCoef1.val[1] = vqrdmulhq_lane_s32(posCoef1.val[1], interp, 0);
+            negCoef.val[0] = vqrdmulhq_lane_s32(negCoef.val[0], interp, 0);
+            negCoef.val[1] = vqrdmulhq_lane_s32(negCoef.val[1], interp, 0);
+
+            posCoef.val[0] = vaddq_s32(posCoef.val[0], posCoef1.val[0]);
+            posCoef.val[1] = vaddq_s32(posCoef.val[1], posCoef1.val[1]);
+            negCoef.val[0] = vaddq_s32(negCoef.val[0], negCoef1.val[0]);
+            negCoef.val[1] = vaddq_s32(negCoef.val[1], negCoef1.val[1]);
+        }
+        switch (CHANNELS) {
+        case 1: {
+            int16x8_t posSamp = vld1q_s16(sP);
+            int16x8_t negSamp = vld1q_s16(sN);
+            sN += 8;
+            posSamp = vrev64q_s16(posSamp);
+
+            int32x4_t posSamp0 = vshll_n_s16(vget_low_s16(posSamp), 15);
+            int32x4_t posSamp1 = vshll_n_s16(vget_high_s16(posSamp), 15);
+            int32x4_t negSamp0 = vshll_n_s16(vget_low_s16(negSamp), 15);
+            int32x4_t negSamp1 = vshll_n_s16(vget_high_s16(negSamp), 15);
+
+            // dot product
+            posSamp0 = vqrdmulhq_s32(posSamp0, posCoef.val[1]); // reversed
+            posSamp1 = vqrdmulhq_s32(posSamp1, posCoef.val[0]); // reversed
+            negSamp0 = vqrdmulhq_s32(negSamp0, negCoef.val[0]);
+            negSamp1 = vqrdmulhq_s32(negSamp1, negCoef.val[1]);
+
+            accum = vaddq_s32(accum, posSamp0);
+            negSamp0 = vaddq_s32(negSamp0, negSamp1);
+            accum = vaddq_s32(accum, posSamp1);
+            accum = vaddq_s32(accum, negSamp0);
+
+            sP -= 8;
+        } break;
+        case 2: {
+            int16x8x2_t posSamp = vld2q_s16(sP);
+            int16x8x2_t negSamp = vld2q_s16(sN);
+            sN += 16;
+            posSamp.val[0] = vrev64q_s16(posSamp.val[0]);
+            posSamp.val[1] = vrev64q_s16(posSamp.val[1]);
+
+            // left
+            int32x4_t posSamp0 = vshll_n_s16(vget_low_s16(posSamp.val[0]), 15);
+            int32x4_t posSamp1 = vshll_n_s16(vget_high_s16(posSamp.val[0]), 15);
+            int32x4_t negSamp0 = vshll_n_s16(vget_low_s16(negSamp.val[0]), 15);
+            int32x4_t negSamp1 = vshll_n_s16(vget_high_s16(negSamp.val[0]), 15);
+
+            // dot product
+            posSamp0 = vqrdmulhq_s32(posSamp0, posCoef.val[1]); // reversed
+            posSamp1 = vqrdmulhq_s32(posSamp1, posCoef.val[0]); // reversed
+            negSamp0 = vqrdmulhq_s32(negSamp0, negCoef.val[0]);
+            negSamp1 = vqrdmulhq_s32(negSamp1, negCoef.val[1]);
+
+            accum = vaddq_s32(accum, posSamp0);
+            negSamp0 = vaddq_s32(negSamp0, negSamp1);
+            accum = vaddq_s32(accum, posSamp1);
+            accum = vaddq_s32(accum, negSamp0);
+
+            // right
+            posSamp0 = vshll_n_s16(vget_low_s16(posSamp.val[1]), 15);
+            posSamp1 = vshll_n_s16(vget_high_s16(posSamp.val[1]), 15);
+            negSamp0 = vshll_n_s16(vget_low_s16(negSamp.val[1]), 15);
+            negSamp1 = vshll_n_s16(vget_high_s16(negSamp.val[1]), 15);
+
+            // dot product
+            posSamp0 = vqrdmulhq_s32(posSamp0, posCoef.val[1]); // reversed
+            posSamp1 = vqrdmulhq_s32(posSamp1, posCoef.val[0]); // reversed
+            negSamp0 = vqrdmulhq_s32(negSamp0, negCoef.val[0]);
+            negSamp1 = vqrdmulhq_s32(negSamp1, negCoef.val[1]);
+
+            accum2 = vaddq_s32(accum2, posSamp0);
+            negSamp0 = vaddq_s32(negSamp0, negSamp1);
+            accum2 = vaddq_s32(accum2, posSamp1);
+            accum2 = vaddq_s32(accum2, negSamp0);
+
+            sP -= 16;
+        } break;
+        }
+    } while (count -= 8);
+
+    // multiply by volume and save
+    volumeLR = (const int32_t*)__builtin_assume_aligned(volumeLR, 8);
+    int32x2_t vLR = vld1_s32(volumeLR);
+    int32x2_t outSamp = vld1_s32(out);
+    // combine and funnel down accumulator
+    int32x2_t outAccum = vpadd_s32(vget_low_s32(accum), vget_high_s32(accum));
+    if (CHANNELS == 1) {
+        // duplicate accum to both L and R
+        outAccum = vpadd_s32(outAccum, outAccum);
+    } else if (CHANNELS == 2) {
+        // accum2 contains R, fold in
+        int32x2_t outAccum2 = vpadd_s32(vget_low_s32(accum2), vget_high_s32(accum2));
+        outAccum = vpadd_s32(outAccum, outAccum2);
+    }
+    outAccum = vqrdmulh_s32(outAccum, vLR);
+    outSamp = vqadd_s32(outSamp, outAccum);
+    vst1_s32(out, outSamp);
+}
+
+template <int CHANNELS, int STRIDE, bool FIXED>
+static inline void ProcessNeonIntrinsic(float* out,
+        int count,
+        const float* coefsP,
+        const float* coefsN,
+        const float* sP,
+        const float* sN,
+        const float* volumeLR,
+        float lerpP,
+        const float* coefsP1,
+        const float* coefsN1)
+{
+    ALOG_ASSERT(count > 0 && (count & 7) == 0); // multiple of 8
+    COMPILE_TIME_ASSERT_FUNCTION_SCOPE(CHANNELS == 1 || CHANNELS == 2);
+
+    sP -= CHANNELS*((STRIDE>>1)-1);
+    coefsP = (const float*)__builtin_assume_aligned(coefsP, 16);
+    coefsN = (const float*)__builtin_assume_aligned(coefsN, 16);
+
+    float32x2_t interp;
+    if (!FIXED) {
+        interp = vdup_n_f32(lerpP);
+        coefsP1 = (const float*)__builtin_assume_aligned(coefsP1, 16);
+        coefsN1 = (const float*)__builtin_assume_aligned(coefsN1, 16);
+    }
+    float32x4_t accum, accum2;
+    // warning uninitialized if we use veorq_s32
+    // (alternative to below) accum = veorq_s32(accum, accum);
+    accum = vdupq_n_f32(0);
+    if (CHANNELS == 2) {
+        // (alternative to below) accum2 = veorq_s32(accum2, accum2);
+        accum2 = vdupq_n_f32(0);
+    }
+    do {
+#ifdef vld1q_f32_x2
+        float32x4x2_t posCoef = vld1q_f32_x2(coefsP);
+        coefsP += 8;
+        float32x4x2_t negCoef = vld1q_f32_x2(coefsN);
+        coefsN += 8;
+#else
+        float32x4x2_t posCoef;
+        posCoef.val[0] = vld1q_f32(coefsP);
+        coefsP += 4;
+        posCoef.val[1] = vld1q_f32(coefsP);
+        coefsP += 4;
+        float32x4x2_t negCoef;
+        negCoef.val[0] = vld1q_f32(coefsN);
+        coefsN += 4;
+        negCoef.val[1] = vld1q_f32(coefsN);
+        coefsN += 4;
+#endif
+        if (!FIXED) { // interpolate
+#ifdef vld1q_f32_x2
+            float32x4x2_t posCoef1 = vld1q_f32_x2(coefsP1);
+            coefsP1 += 8;
+            float32x4x2_t negCoef1 = vld1q_f32_x2(coefsN1);
+            coefsN1 += 8;
+#else
+            float32x4x2_t posCoef1;
+            posCoef1.val[0] = vld1q_f32(coefsP1);
+            coefsP1 += 4;
+            posCoef1.val[1] = vld1q_f32(coefsP1);
+            coefsP1 += 4;
+            float32x4x2_t negCoef1;
+            negCoef1.val[0] = vld1q_f32(coefsN1);
+            coefsN1 += 4;
+            negCoef1.val[1] = vld1q_f32(coefsN1);
+            coefsN1 += 4;
+#endif
+            posCoef1.val[0] = vsubq_f32(posCoef1.val[0], posCoef.val[0]);
+            posCoef1.val[1] = vsubq_f32(posCoef1.val[1], posCoef.val[1]);
+            negCoef.val[0] = vsubq_f32(negCoef.val[0], negCoef1.val[0]);
+            negCoef.val[1] = vsubq_f32(negCoef.val[1], negCoef1.val[1]);
+
+            posCoef.val[0] = vmlaq_lane_f32(posCoef.val[0], posCoef1.val[0], interp, 0);
+            posCoef.val[1] = vmlaq_lane_f32(posCoef.val[1], posCoef1.val[1], interp, 0);
+            negCoef.val[0] = vmlaq_lane_f32(negCoef1.val[0], negCoef.val[0], interp, 0); // rev
+            negCoef.val[1] = vmlaq_lane_f32(negCoef1.val[1], negCoef.val[1], interp, 0); // rev
+        }
+        switch (CHANNELS) {
+        case 1: {
+#ifdef vld1q_f32_x2
+            float32x4x2_t posSamp = vld1q_f32_x2(sP);
+            float32x4x2_t negSamp = vld1q_f32_x2(sN);
+            sN += 8;
+            sP -= 8;
+#else
+            float32x4x2_t posSamp;
+            posSamp.val[0] = vld1q_f32(sP);
+            sP += 4;
+            posSamp.val[1] = vld1q_f32(sP);
+            sP -= 12;
+            float32x4x2_t negSamp;
+            negSamp.val[0] = vld1q_f32(sN);
+            sN += 4;
+            negSamp.val[1] = vld1q_f32(sN);
+            sN += 4;
+#endif
+            // effectively we want a vrev128q_f32()
+            posSamp.val[0] = vrev64q_f32(posSamp.val[0]);
+            posSamp.val[1] = vrev64q_f32(posSamp.val[1]);
+            posSamp.val[0] = vcombine_f32(
+                    vget_high_f32(posSamp.val[0]), vget_low_f32(posSamp.val[0]));
+            posSamp.val[1] = vcombine_f32(
+                    vget_high_f32(posSamp.val[1]), vget_low_f32(posSamp.val[1]));
+
+            accum = vmlaq_f32(accum, posSamp.val[0], posCoef.val[1]);
+            accum = vmlaq_f32(accum, posSamp.val[1], posCoef.val[0]);
+            accum = vmlaq_f32(accum, negSamp.val[0], negCoef.val[0]);
+            accum = vmlaq_f32(accum, negSamp.val[1], negCoef.val[1]);
+        } break;
+        case 2: {
+            float32x4x2_t posSamp0 = vld2q_f32(sP);
+            sP += 8;
+            float32x4x2_t negSamp0 = vld2q_f32(sN);
+            sN += 8;
+            posSamp0.val[0] = vrev64q_f32(posSamp0.val[0]);
+            posSamp0.val[1] = vrev64q_f32(posSamp0.val[1]);
+            posSamp0.val[0] = vcombine_f32(
+                    vget_high_f32(posSamp0.val[0]), vget_low_f32(posSamp0.val[0]));
+            posSamp0.val[1] = vcombine_f32(
+                    vget_high_f32(posSamp0.val[1]), vget_low_f32(posSamp0.val[1]));
+
+            float32x4x2_t posSamp1 = vld2q_f32(sP);
+            sP -= 24;
+            float32x4x2_t negSamp1 = vld2q_f32(sN);
+            sN += 8;
+            posSamp1.val[0] = vrev64q_f32(posSamp1.val[0]);
+            posSamp1.val[1] = vrev64q_f32(posSamp1.val[1]);
+            posSamp1.val[0] = vcombine_f32(
+                    vget_high_f32(posSamp1.val[0]), vget_low_f32(posSamp1.val[0]));
+            posSamp1.val[1] = vcombine_f32(
+                    vget_high_f32(posSamp1.val[1]), vget_low_f32(posSamp1.val[1]));
+
+            // Note: speed is affected by accumulation order.
+            // Also, speed appears slower using vmul/vadd instead of vmla for
+            // stereo case, comparable for mono.
+
+            accum = vmlaq_f32(accum, negSamp0.val[0], negCoef.val[0]);
+            accum = vmlaq_f32(accum, negSamp1.val[0], negCoef.val[1]);
+            accum2 = vmlaq_f32(accum2, negSamp0.val[1], negCoef.val[0]);
+            accum2 = vmlaq_f32(accum2, negSamp1.val[1], negCoef.val[1]);
+
+            accum = vmlaq_f32(accum, posSamp0.val[0], posCoef.val[1]); // reversed
+            accum = vmlaq_f32(accum, posSamp1.val[0], posCoef.val[0]); // reversed
+            accum2 = vmlaq_f32(accum2, posSamp0.val[1], posCoef.val[1]); // reversed
+            accum2 = vmlaq_f32(accum2, posSamp1.val[1], posCoef.val[0]); // reversed
+        } break;
+        }
+    } while (count -= 8);
+
+    // multiply by volume and save
+    volumeLR = (const float*)__builtin_assume_aligned(volumeLR, 8);
+    float32x2_t vLR = vld1_f32(volumeLR);
+    float32x2_t outSamp = vld1_f32(out);
+    // combine and funnel down accumulator
+    float32x2_t outAccum = vpadd_f32(vget_low_f32(accum), vget_high_f32(accum));
+    if (CHANNELS == 1) {
+        // duplicate accum to both L and R
+        outAccum = vpadd_f32(outAccum, outAccum);
+    } else if (CHANNELS == 2) {
+        // accum2 contains R, fold in
+        float32x2_t outAccum2 = vpadd_f32(vget_low_f32(accum2), vget_high_f32(accum2));
+        outAccum = vpadd_f32(outAccum, outAccum2);
+    }
+    outSamp = vmla_f32(outSamp, outAccum, vLR);
+    vst1_f32(out, outSamp);
+}
+
 template <>
 inline void ProcessL<1, 16>(int32_t* const out,
         int count,
@@ -54,6 +531,10 @@
         const int16_t* sN,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
     const int CHANNELS = 1; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -95,6 +576,7 @@
           "q0", "q1", "q2", "q3",
           "q8", "q10"
     );
+#endif
 }
 
 template <>
@@ -106,6 +588,10 @@
         const int16_t* sN,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
     const int CHANNELS = 2; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -153,6 +639,7 @@
           "q4", "q5", "q6",
           "q8", "q10"
      );
+#endif
 }
 
 template <>
@@ -167,6 +654,11 @@
         uint32_t lerpP,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            lerpP, coefsP1, coefsN1);
+#else
+
     const int CHANNELS = 1; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -223,6 +715,7 @@
           "q0", "q1", "q2", "q3",
           "q8", "q9", "q10", "q11"
     );
+#endif
 }
 
 template <>
@@ -237,6 +730,10 @@
         uint32_t lerpP,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            lerpP, coefsP1, coefsN1);
+#else
     const int CHANNELS = 2; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -299,6 +796,7 @@
           "q4", "q5", "q6",
           "q8", "q9", "q10", "q11"
     );
+#endif
 }
 
 template <>
@@ -310,6 +808,10 @@
         const int16_t* sN,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
     const int CHANNELS = 1; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -360,6 +862,7 @@
           "q8", "q9", "q10", "q11",
           "q12", "q13", "q14", "q15"
     );
+#endif
 }
 
 template <>
@@ -371,6 +874,10 @@
         const int16_t* sN,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+#else
     const int CHANNELS = 2; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -440,6 +947,7 @@
           "q8", "q9", "q10", "q11",
           "q12", "q13", "q14", "q15"
     );
+#endif
 }
 
 template <>
@@ -454,6 +962,10 @@
         uint32_t lerpP,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            lerpP, coefsP1, coefsN1);
+#else
     const int CHANNELS = 1; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -525,6 +1037,7 @@
           "q8", "q9", "q10", "q11",
           "q12", "q13", "q14", "q15"
     );
+#endif
 }
 
 template <>
@@ -539,6 +1052,10 @@
         uint32_t lerpP,
         const int32_t* const volumeLR)
 {
+#ifdef USE_INTRINSIC
+    ProcessNeonIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            lerpP, coefsP1, coefsN1);
+#else
     const int CHANNELS = 2; // template specialization does not preserve params
     const int STRIDE = 16;
     sP -= CHANNELS*((STRIDE>>1)-1);
@@ -629,6 +1146,65 @@
           "q8", "q9", "q10", "q11",
           "q12", "q13", "q14", "q15"
     );
+#endif
+}
+
+template<>
+inline void ProcessL<1, 16>(float* const out,
+        int count,
+        const float* coefsP,
+        const float* coefsN,
+        const float* sP,
+        const float* sN,
+        const float* const volumeLR)
+{
+    ProcessNeonIntrinsic<1, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+}
+
+template<>
+inline void ProcessL<2, 16>(float* const out,
+        int count,
+        const float* coefsP,
+        const float* coefsN,
+        const float* sP,
+        const float* sN,
+        const float* const volumeLR)
+{
+    ProcessNeonIntrinsic<2, 16, true>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            0 /*lerpP*/, NULL /*coefsP1*/, NULL /*coefsN1*/);
+}
+
+template<>
+inline void Process<1, 16>(float* const out,
+        int count,
+        const float* coefsP,
+        const float* coefsN,
+        const float* coefsP1,
+        const float* coefsN1,
+        const float* sP,
+        const float* sN,
+        float lerpP,
+        const float* const volumeLR)
+{
+    ProcessNeonIntrinsic<1, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            lerpP, coefsP1, coefsN1);
+}
+
+template<>
+inline void Process<2, 16>(float* const out,
+        int count,
+        const float* coefsP,
+        const float* coefsN,
+        const float* coefsP1,
+        const float* coefsN1,
+        const float* sP,
+        const float* sN,
+        float lerpP,
+        const float* const volumeLR)
+{
+    ProcessNeonIntrinsic<2, 16, false>(out, count, coefsP, coefsN, sP, sN, volumeLR,
+            lerpP, coefsP1, coefsN1);
 }
 
 #endif //USE_NEON
diff --git a/services/audioflinger/test-resample.cpp b/services/audioflinger/test-resample.cpp
index 84a655a..7893778 100644
--- a/services/audioflinger/test-resample.cpp
+++ b/services/audioflinger/test-resample.cpp
@@ -427,6 +427,14 @@
         printf("quality: %d  channels: %d  msec: %" PRId64 "  Mfrms/s: %.2lf\n",
                 quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6);
         resampler->reset();
+
+        // TODO fix legacy bug: reset does not clear buffers.
+        // delete and recreate resampler here.
+        delete resampler;
+        resampler = AudioResampler::create(format, channels,
+                    output_freq, quality);
+        resampler->setSampleRate(input_freq);
+        resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT);
     }
 
     memset(output_vaddr, 0, output_size);
diff --git a/services/audioflinger/tests/build_and_run_all_unit_tests.sh b/services/audioflinger/tests/build_and_run_all_unit_tests.sh
index 2c453b0..7f4d456 100755
--- a/services/audioflinger/tests/build_and_run_all_unit_tests.sh
+++ b/services/audioflinger/tests/build_and_run_all_unit_tests.sh
@@ -15,7 +15,7 @@
 echo "waiting for device"
 adb root && adb wait-for-device remount
 adb push $OUT/system/lib/libaudioresampler.so /system/lib
-adb push $OUT/system/bin/resampler_tests /system/bin
+adb push $OUT/data/nativetest/resampler_tests /system/bin
 
 sh $ANDROID_BUILD_TOP/frameworks/av/services/audioflinger/tests/run_all_unit_tests.sh
 
diff --git a/services/audioflinger/tests/mixer_to_wav_tests.sh b/services/audioflinger/tests/mixer_to_wav_tests.sh
index e60e6d5..d0482a1 100755
--- a/services/audioflinger/tests/mixer_to_wav_tests.sh
+++ b/services/audioflinger/tests/mixer_to_wav_tests.sh
@@ -60,7 +60,7 @@
     fi
 
 # Test:
-# process__genericResampling
+# process__genericResampling with mixed integer and float track input
 # track__Resample / track__genericResample
     adb shell test-mixer $1 -s 48000 \
         -o /sdcard/tm48000grif.wav \
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index b48dc80..53ec0f6 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -4467,7 +4467,7 @@
             if (device) break;
             device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE;
             if (device) break;
-            if (mPhoneState != AUDIO_MODE_IN_CALL) {
+            if (!isInCall()) {
                 device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY;
                 if (device) break;
                 device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;