diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 46e3d6c..e49b7b1 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -41,7 +41,7 @@
     AudioResamplerOrder1(int inChannelCount, int32_t sampleRate) :
         AudioResampler(inChannelCount, sampleRate, LOW_QUALITY), mX0L(0), mX0R(0) {
     }
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 private:
     // number of bits used in interpolation multiply - 15 bits avoids overflow
@@ -51,9 +51,9 @@
     static const int kPreInterpShift = kNumPhaseBits - kNumInterpBits;
 
     void init() {}
-    void resampleMono16(int32_t* out, size_t outFrameCount,
+    size_t resampleMono16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
-    void resampleStereo16(int32_t* out, size_t outFrameCount,
+    size_t 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,
@@ -329,7 +329,7 @@
 
 // ----------------------------------------------------------------------------
 
-void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     // should never happen, but we overflow if it does
@@ -338,15 +338,16 @@
     // select the appropriate resampler
     switch (mChannelCount) {
     case 1:
-        resampleMono16(out, outFrameCount, provider);
-        break;
+        return resampleMono16(out, outFrameCount, provider);
     case 2:
-        resampleStereo16(out, outFrameCount, provider);
-        break;
+        return resampleStereo16(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
     }
 }
 
-void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -442,9 +443,10 @@
     // save state
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex / 2 /* channels for stereo */;
 }
 
-void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -538,6 +540,7 @@
     // save state
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex;
 }
 
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 863614a..a8e3e6f 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -67,12 +67,18 @@
     // Resample int16_t samples from provider and accumulate into 'out'.
     // A mono provider delivers a sequence of samples.
     // A stereo provider delivers a sequence of interleaved pairs of samples.
-    // Multi-channel providers are not supported.
+    //
     // In either case, 'out' holds interleaved pairs of fixed-point Q4.27.
     // That is, for a mono provider, there is an implicit up-channeling.
     // Since this method accumulates, the caller is responsible for clearing 'out' initially.
-    // FIXME assumes provider is always successful; it should return the actual frame count.
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    //
+    // For a float resampler, 'out' holds interleaved pairs of float samples.
+    //
+    // Multichannel interleaved frames for n > 2 is supported for quality DYN_LOW_QUALITY,
+    // DYN_MED_QUALITY, and DYN_HIGH_QUALITY.
+    //
+    // Returns the number of frames resampled into the out buffer.
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider) = 0;
 
     virtual void reset();
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index d3cbd1c..172c2a5 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#define LOG_TAG "AudioSRC"
+#define LOG_TAG "AudioResamplerCubic"
 
 #include <stdint.h>
 #include <string.h>
@@ -32,7 +32,7 @@
     memset(&right, 0, sizeof(state));
 }
 
-void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     // should never happen, but we overflow if it does
@@ -41,15 +41,16 @@
     // select the appropriate resampler
     switch (mChannelCount) {
     case 1:
-        resampleMono16(out, outFrameCount, provider);
-        break;
+        return resampleMono16(out, outFrameCount, provider);
     case 2:
-        resampleStereo16(out, outFrameCount, provider);
-        break;
+        return resampleStereo16(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
     }
 }
 
-void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -67,7 +68,7 @@
         mBuffer.frameCount = inFrameCount;
         provider->getNextBuffer(&mBuffer, mPTS);
         if (mBuffer.raw == NULL) {
-            return;
+            return 0;
         }
         // ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
     }
@@ -115,9 +116,10 @@
     // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex / 2 /* channels for stereo */;
 }
 
-void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider) {
 
     int32_t vl = mVolume[0];
@@ -135,7 +137,7 @@
         mBuffer.frameCount = inFrameCount;
         provider->getNextBuffer(&mBuffer, mPTS);
         if (mBuffer.raw == NULL) {
-            return;
+            return 0;
         }
         // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
     }
@@ -182,6 +184,7 @@
     // ALOGW("Done: index=%d, fraction=%u", inputIndex, phaseFraction);
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex;
 }
 
 // ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index 1ddc5f9..4b45b0b 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -31,7 +31,7 @@
     AudioResamplerCubic(int inChannelCount, int32_t sampleRate) :
         AudioResampler(inChannelCount, sampleRate, MED_QUALITY) {
     }
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 private:
     // number of bits used in interpolation multiply - 14 bits avoids overflow
@@ -43,9 +43,9 @@
         int32_t a, b, c, y0, y1, y2, y3;
     } state;
     void init();
-    void resampleMono16(int32_t* out, size_t outFrameCount,
+    size_t resampleMono16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
-    void resampleStereo16(int32_t* out, size_t outFrameCount,
+    size_t resampleStereo16(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
     static inline int32_t interp(state* p, int32_t x) {
         return (((((p->a * x >> 14) + p->b) * x >> 14) + p->c) * x >> 14) + p->y1;
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp
index c21d4ca..6481b85 100644
--- a/services/audioflinger/AudioResamplerDyn.cpp
+++ b/services/audioflinger/AudioResamplerDyn.cpp
@@ -477,15 +477,15 @@
 }
 
 template<typename TC, typename TI, typename TO>
-void AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider)
 {
-    (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
+    return (this->*mResampleFunc)(reinterpret_cast<TO*>(out), outFrameCount, provider);
 }
 
 template<typename TC, typename TI, typename TO>
 template<int CHANNELS, bool LOCKED, int STRIDE>
-void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
+size_t AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
         AudioBufferProvider* provider)
 {
     // TODO Mono -> Mono is not supported. OUTPUT_CHANNELS reflects minimum of stereo out.
@@ -610,6 +610,7 @@
     ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer
     mInBuffer.setImpulse(impulse);
     mPhaseFraction = phaseFraction;
+    return outputIndex / OUTPUT_CHANNELS;
 }
 
 /* instantiate templates used by AudioResampler::create */
diff --git a/services/audioflinger/AudioResamplerDyn.h b/services/audioflinger/AudioResamplerDyn.h
index 238b163..3b1c381 100644
--- a/services/audioflinger/AudioResamplerDyn.h
+++ b/services/audioflinger/AudioResamplerDyn.h
@@ -52,7 +52,7 @@
 
     virtual void setVolume(float left, float right);
 
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
 private:
@@ -111,10 +111,10 @@
             int inSampleRate, int outSampleRate, double tbwCheat);
 
     template<int CHANNELS, bool LOCKED, int STRIDE>
-    void resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
+    size_t resample(TO* out, size_t outFrameCount, AudioBufferProvider* provider);
 
     // define a pointer to member function type for resample
-    typedef void (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
+    typedef size_t (AudioResamplerDyn<TC, TI, TO>::*resample_ABP_t)(TO* out,
             size_t outFrameCount, AudioBufferProvider* provider);
 
     // data - the contiguous storage and layout of these is important.
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index ba9a356..41730ee 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -256,7 +256,7 @@
     mVolumeSIMD[1] = u4_28_from_float(clampFloatVol(right));
 }
 
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider)
 {
     // FIXME store current state (up or down sample) and only load the coefs when the state
@@ -272,17 +272,18 @@
     // select the appropriate resampler
     switch (mChannelCount) {
     case 1:
-        resample<1>(out, outFrameCount, provider);
-        break;
+        return resample<1>(out, outFrameCount, provider);
     case 2:
-        resample<2>(out, outFrameCount, provider);
-        break;
+        return resample<2>(out, outFrameCount, provider);
+    default:
+        LOG_ALWAYS_FATAL("invalid channel count: %d", mChannelCount);
+        return 0;
     }
 }
 
 
 template<int CHANNELS>
-void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
+size_t AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount,
         AudioBufferProvider* provider)
 {
     const Constants& c(*mConstants);
@@ -357,6 +358,7 @@
     mImpulse = impulse;
     mInputIndex = inputIndex;
     mPhaseFraction = phaseFraction;
+    return outputIndex / CHANNELS;
 }
 
 template<int CHANNELS>
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 6d8e85d..0fbeac8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -39,7 +39,7 @@
 
     virtual ~AudioResamplerSinc();
 
-    virtual void resample(int32_t* out, size_t outFrameCount,
+    virtual size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 private:
     void init();
@@ -47,7 +47,7 @@
     virtual void setVolume(float left, float right);
 
     template<int CHANNELS>
-    void resample(int32_t* out, size_t outFrameCount,
+    size_t resample(int32_t* out, size_t outFrameCount,
             AudioBufferProvider* provider);
 
     template<int CHANNELS>
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index d6217ba..9e375db 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -48,7 +48,10 @@
         if (thisFrames == 0 || thisFrames > outputFrames - i) {
             thisFrames = outputFrames - i;
         }
-        resampler->resample((int32_t*) output + channels*i, thisFrames, provider);
+        size_t framesResampled = resampler->resample(
+                (int32_t*) output + channels*i, thisFrames, provider);
+        // we should have enough buffer space, so there is no short count.
+        ASSERT_EQ(thisFrames, framesResampled);
         i += thisFrames;
     }
 }