diff --git a/media/common_time/utils.cpp b/media/common_time/utils.cpp
index 6539171..91cf2fd 100644
--- a/media/common_time/utils.cpp
+++ b/media/common_time/utils.cpp
@@ -59,7 +59,7 @@
 }
 
 void deserializeSockaddr(const Parcel* p, struct sockaddr_storage* addr) {
-    memset(addr, 0, sizeof(addr));
+    memset(addr, 0, sizeof(*addr));
 
     addr->ss_family = p->readInt32();
     switch(addr->ss_family) {
diff --git a/services/audioflinger/AudioResamplerFirGen.h b/services/audioflinger/AudioResamplerFirGen.h
index 8833758..d024b2f 100644
--- a/services/audioflinger/AudioResamplerFirGen.h
+++ b/services/audioflinger/AudioResamplerFirGen.h
@@ -365,6 +365,28 @@
     }
 }
 
+/* A speed optimized version of the Modified Bessel I0() which incorporates
+ * the sqrt and numerator multiply and denominator divide into the computation.
+ * This speeds up filter computation by about 10-15%.
+ */
+static inline double I0SqrRat(double x2, double num, double den) {
+    if (x2 < (3.75 * 3.75)) {
+        return Poly7(I0Term<0>::value, I0Term<1>::value,
+                I0Term<2>::value, I0Term<3>::value,
+                I0Term<4>::value, I0Term<5>::value,
+                I0Term<6>::value, x2) * num / den; // e < 1.6e-7
+    }
+    num *= Poly9(-0.13544938430e9, -0.33153754512e8,
+            -0.19406631946e7, -0.48058318783e5,
+            -0.63269783360e3, -0.49520779070e1,
+            -0.24970910370e-1, -0.74741159550e-4,
+            -0.18257612460e-6, x2); // e < 10^(-7.13).
+    double y = x2 - 225.; // reflection around 15 (squared)
+    den *= Poly4(-0.34598737196e8, 0.23852643181e6,
+            -0.70699387620e3, 0.10000000000e1, y);
+    return num / den;
+}
+
 /*
  * calculates the transition bandwidth for a Kaiser filter
  *
@@ -645,6 +667,7 @@
     const double xstep = (2. * M_PI) * fcr / L;
     const double xfrac = 1. / N;
     const double yscale = atten * L / (I0(beta) * M_PI);
+    const double sqrbeta = sqr(beta);
 
     // We use sine generators, which computes sines on regular step intervals.
     // This speeds up overall computation about 40% from computing the sine directly.
@@ -663,7 +686,8 @@
                 double x = static_cast<double>(ix);
 
                 // sine generator: sg.valueAdvance() returns sin(ix*xstep);
-                y = I0(beta * sqrt(1.0 - sqr(x * xfrac))) * yscale * sg.valueAdvance() / x;
+                // y = I0(beta * sqrt(1.0 - sqr(x * xfrac))) * yscale * sg.valueAdvance() / x;
+                y = I0SqrRat(sqrbeta * (1.0 - sqr(x * xfrac)), yscale * sg.valueAdvance(), x);
             } else {
                 y = 2. * atten * fcr; // center of filter, sinc(0) = 1.
                 sg.advance();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index feedd89..ae3dd8b 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -3951,6 +3951,16 @@
         AudioParameter param = AudioParameter(keyValuePair);
         int value;
 
+        if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
+            // forward device change to effects that have requested to be
+            // aware of attached audio device.
+            if (value != AUDIO_DEVICE_NONE) {
+                mOutDevice = value;
+                for (size_t i = 0; i < mEffectChains.size(); i++) {
+                    mEffectChains[i]->setDevice_l(mOutDevice);
+                }
+            }
+        }
         if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
             // do not accept frame count changes if tracks are open as the track buffer
             // size depends on frame count and correct behavior would not be garantied