diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index b206116..ca98f16 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -341,7 +341,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
@@ -439,7 +439,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index dc33f29..0592855 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -110,6 +110,38 @@
     uint64_t mLocalTimeFreq;
     int64_t mPTS;
 
+    // returns the inFrameCount required to generate outFrameCount frames.
+    //
+    // Placed here to be a consistent for all resamplers.
+    //
+    // Right now, we use the upper bound without regards to the current state of the
+    // input buffer using integer arithmetic, as follows:
+    //
+    // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;
+    //
+    // The double precision equivalent (float may not be precise enough):
+    // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);
+    //
+    // this relies on the fact that the mPhaseIncrement is rounded down from
+    // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).
+    // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums
+    //
+    // (so long as double precision is computed accurately enough to be considered
+    // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this
+    // will not necessarily hold for floats).
+    //
+    // TODO:
+    // Greater accuracy and a tight bound is obtained by:
+    // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.
+    // 2) using the exact integer formula where (ignoring 64b casting)
+    //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;
+    //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.
+    //
+    inline size_t getInFrameCountRequired(size_t outFrameCount) {
+        return (static_cast<uint64_t>(outFrameCount)*mInSampleRate
+                + (mSampleRate - 1))/mSampleRate;
+    }
+
 private:
     const src_quality mQuality;
 
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 1f9714b..8f14ff9 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -60,7 +60,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // fetch first buffer
     if (mBuffer.frameCount == 0) {
@@ -128,7 +128,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     // fetch first buffer
     if (mBuffer.frameCount == 0) {
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index 54c2309..939b128 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -466,7 +466,7 @@
     const uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;   // stereo output
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
     const uint32_t phaseWrapLimit = c.mL << c.mShift;
 
     // NOTE: be very careful when modifying the code here. register
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 207f26b..d0a7a58 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -540,7 +540,7 @@
     uint32_t phaseIncrement = mPhaseIncrement;
     size_t outputIndex = 0;
     size_t outputSampleCount = outFrameCount * 2;
-    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
+    size_t inFrameCount = getInFrameCountRequired(outFrameCount);
 
     while (outputIndex < outputSampleCount) {
         // buffer is empty, fetch a new one