diff --git a/media/libaaudio/src/client/AAudioFlowGraph.cpp b/media/libaaudio/src/client/AAudioFlowGraph.cpp
index 5444565..69be050 100644
--- a/media/libaaudio/src/client/AAudioFlowGraph.cpp
+++ b/media/libaaudio/src/client/AAudioFlowGraph.cpp
@@ -39,18 +39,21 @@
 
 aaudio_result_t AAudioFlowGraph::configure(audio_format_t sourceFormat,
                           int32_t sourceChannelCount,
+                          int32_t sourceSampleRate,
                           audio_format_t sinkFormat,
                           int32_t sinkChannelCount,
+                          int32_t sinkSampleRate,
                           bool useMonoBlend,
+                          bool useVolumeRamps,
                           float audioBalance,
-                          bool isExclusive) {
+                          aaudio::resampler::MultiChannelResampler::Quality resamplerQuality) {
     FlowGraphPortFloatOutput *lastOutput = nullptr;
 
-    // TODO change back to ALOGD
-    ALOGI("%s() source format = 0x%08x, channels = %d, sink format = 0x%08x, channels = %d, "
-          "useMonoBlend = %d, audioBalance = %f, isExclusive %d",
-          __func__, sourceFormat, sourceChannelCount, sinkFormat, sinkChannelCount,
-          useMonoBlend, audioBalance, isExclusive);
+    ALOGD("%s() source format = 0x%08x, channels = %d, sample rate = %d, "
+          "sink format = 0x%08x, channels = %d, sample rate = %d, "
+          "useMonoBlend = %d, audioBalance = %f, useVolumeRamps %d",
+          __func__, sourceFormat, sourceChannelCount, sourceSampleRate, sinkFormat,
+          sinkChannelCount, sinkSampleRate, useMonoBlend, audioBalance, useVolumeRamps);
 
     switch (sourceFormat) {
         case AUDIO_FORMAT_PCM_FLOAT:
@@ -85,6 +88,15 @@
         lastOutput = &mLimiter->output;
     }
 
+    if (sourceSampleRate != sinkSampleRate) {
+        mResampler.reset(aaudio::resampler::MultiChannelResampler::make(sinkChannelCount,
+                sourceSampleRate, sinkSampleRate, resamplerQuality));
+        mRateConverter = std::make_unique<SampleRateConverter>(sinkChannelCount,
+                                                               *mResampler);
+        lastOutput->connect(&mRateConverter->input);
+        lastOutput = &mRateConverter->output;
+    }
+
     // Expand the number of channels if required.
     if (sourceChannelCount == 1 && sinkChannelCount > 1) {
         mChannelConverter = std::make_unique<MonoToMultiConverter>(sinkChannelCount);
@@ -95,8 +107,7 @@
         return AAUDIO_ERROR_UNIMPLEMENTED;
     }
 
-    // Apply volume ramps for only exclusive streams.
-    if (isExclusive) {
+    if (useVolumeRamps) {
         // Apply volume ramps to set the left/right audio balance and target volumes.
         // The signals will be decoupled, volume ramps will be applied, before the signals are
         // combined again.
@@ -137,9 +148,14 @@
     return AAUDIO_OK;
 }
 
-void AAudioFlowGraph::process(const void *source, void *destination, int32_t numFrames) {
-    mSource->setData(source, numFrames);
-    mSink->read(destination, numFrames);
+int32_t AAudioFlowGraph::pull(void *destination, int32_t targetFramesToRead) {
+    return mSink->read(destination, targetFramesToRead);
+}
+
+int32_t AAudioFlowGraph::process(const void *source, int32_t numFramesToWrite, void *destination,
+                    int32_t targetFramesToRead) {
+    mSource->setData(source, numFramesToWrite);
+    return mSink->read(destination, targetFramesToRead);
 }
 
 /**
diff --git a/media/libaaudio/src/client/AAudioFlowGraph.h b/media/libaaudio/src/client/AAudioFlowGraph.h
index 35fef37..e1d517e 100644
--- a/media/libaaudio/src/client/AAudioFlowGraph.h
+++ b/media/libaaudio/src/client/AAudioFlowGraph.h
@@ -30,6 +30,7 @@
 #include <flowgraph/MonoToMultiConverter.h>
 #include <flowgraph/MultiToManyConverter.h>
 #include <flowgraph/RampLinear.h>
+#include <flowgraph/SampleRateConverter.h>
 
 class AAudioFlowGraph {
 public:
@@ -38,23 +39,57 @@
      *
      * @param sourceFormat
      * @param sourceChannelCount
+     * @param sourceSampleRate
      * @param sinkFormat
      * @param sinkChannelCount
+     * @param sinkSampleRate
      * @param useMonoBlend
+     * @param useVolumeRamps
      * @param audioBalance
-     * @param channelMask
-     * @param isExclusive
+     * @param resamplerQuality
      * @return
      */
     aaudio_result_t configure(audio_format_t sourceFormat,
                               int32_t sourceChannelCount,
+                              int32_t sourceSampleRate,
                               audio_format_t sinkFormat,
                               int32_t sinkChannelCount,
+                              int32_t sinkSampleRate,
                               bool useMonoBlend,
+                              bool useVolumeRamps,
                               float audioBalance,
-                              bool isExclusive);
+                              aaudio::resampler::MultiChannelResampler::Quality resamplerQuality);
 
-    void process(const void *source, void *destination, int32_t numFrames);
+    /**
+     * Attempt to read targetFramesToRead from the flowgraph.
+     * This function returns the number of frames actually read.
+     *
+     * This function does nothing if process() was not called before.
+     *
+     * @param destination
+     * @param targetFramesToRead
+     * @return numFramesRead
+     */
+    int32_t pull(void *destination, int32_t targetFramesToRead);
+
+    /**
+     * Set numFramesToWrite frames from the source into the flowgraph.
+     * Then, attempt to read targetFramesToRead from the flowgraph.
+     * This function returns the number of frames actually read.
+     *
+     * There may be data still in the flowgraph if targetFramesToRead is not large enough.
+     * Before calling process() again, pull() must be called until until all the data is consumed.
+     *
+     * TODO: b/289510598 - Calculate the exact number of input frames needed for Y output frames.
+     *
+     * @param source
+     * @param numFramesToWrite
+     * @param destination
+     * @param targetFramesToRead
+     * @return numFramesRead
+     */
+    int32_t process(const void *source, int32_t numFramesToWrite, void *destination,
+                    int32_t targetFramesToRead);
 
     /**
      * @param volume between 0.0 and 1.0
@@ -73,6 +108,8 @@
 
 private:
     std::unique_ptr<FLOWGRAPH_OUTER_NAMESPACE::flowgraph::FlowGraphSourceBuffered> mSource;
+    std::unique_ptr<RESAMPLER_OUTER_NAMESPACE::resampler::MultiChannelResampler> mResampler;
+    std::unique_ptr<FLOWGRAPH_OUTER_NAMESPACE::flowgraph::SampleRateConverter> mRateConverter;
     std::unique_ptr<FLOWGRAPH_OUTER_NAMESPACE::flowgraph::MonoBlend> mMonoBlend;
     std::unique_ptr<FLOWGRAPH_OUTER_NAMESPACE::flowgraph::Limiter> mLimiter;
     std::unique_ptr<FLOWGRAPH_OUTER_NAMESPACE::flowgraph::MonoToMultiConverter> mChannelConverter;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 93ac966..d75832f 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -63,6 +63,8 @@
 
 #define LOG_TIMESTAMPS            0
 
+#define ENABLE_SAMPLE_RATE_CONVERTER 1
+
 AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface  &serviceInterface, bool inService)
         : AudioStream()
         , mClockModel()
@@ -179,7 +181,6 @@
 
     mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
 
-    setSampleRate(configurationOutput.getSampleRate());
     setDeviceId(configurationOutput.getDeviceId());
     setSessionId(configurationOutput.getSessionId());
     setSharingMode(configurationOutput.getSharingMode());
@@ -190,6 +191,18 @@
     setIsContentSpatialized(configurationOutput.isContentSpatialized());
     setInputPreset(configurationOutput.getInputPreset());
 
+    setDeviceSampleRate(configurationOutput.getSampleRate());
+
+    if (getSampleRate() == AAUDIO_UNSPECIFIED) {
+        setSampleRate(configurationOutput.getSampleRate());
+    }
+
+#if !ENABLE_SAMPLE_RATE_CONVERTER
+    if (getSampleRate() != getDeviceSampleRate()) {
+        goto error;
+    }
+#endif
+
     // Save device format so we can do format conversion and volume scaling together.
     setDeviceFormat(configurationOutput.getFormat());
 
@@ -229,39 +242,46 @@
 }
 
 aaudio_result_t AudioStreamInternal::configureDataInformation(int32_t callbackFrames) {
-    int32_t framesPerHardwareBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
+    int32_t deviceFramesPerBurst = mEndpointDescriptor.dataQueueDescriptor.framesPerBurst;
 
     // Scale up the burst size to meet the minimum equivalent in microseconds.
     // This is to avoid waking the CPU too often when the HW burst is very small
-    // or at high sample rates.
-    int32_t framesPerBurst = framesPerHardwareBurst;
+    // or at high sample rates. The actual number of frames that we call back to
+    // the app with will be 0 < N <= framesPerBurst so round up the division.
+    int32_t framesPerBurst = (static_cast<int64_t>(deviceFramesPerBurst) * getSampleRate() +
+             getDeviceSampleRate() - 1) / getDeviceSampleRate();
     int32_t burstMicros = 0;
     const int32_t burstMinMicros = android::AudioSystem::getAAudioHardwareBurstMinUsec();
     do {
         if (burstMicros > 0) {  // skip first loop
+            deviceFramesPerBurst *= 2;
             framesPerBurst *= 2;
         }
         burstMicros = framesPerBurst * static_cast<int64_t>(1000000) / getSampleRate();
     } while (burstMicros < burstMinMicros);
     ALOGD("%s() original HW burst = %d, minMicros = %d => SW burst = %d\n",
-          __func__, framesPerHardwareBurst, burstMinMicros, framesPerBurst);
+          __func__, deviceFramesPerBurst, burstMinMicros, framesPerBurst);
 
     // Validate final burst size.
     if (framesPerBurst < MIN_FRAMES_PER_BURST || framesPerBurst > MAX_FRAMES_PER_BURST) {
         ALOGE("%s - framesPerBurst out of range = %d", __func__, framesPerBurst);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
+    setDeviceFramesPerBurst(deviceFramesPerBurst);
     setFramesPerBurst(framesPerBurst); // only save good value
 
-    mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
+    mDeviceBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
+
+    mBufferCapacityInFrames = static_cast<int64_t>(mDeviceBufferCapacityInFrames)
+            * getSampleRate() / getDeviceSampleRate();
     if (mBufferCapacityInFrames < getFramesPerBurst()
             || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
         ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
         return AAUDIO_ERROR_OUT_OF_RANGE;
     }
 
-    mClockModel.setSampleRate(getSampleRate());
-    mClockModel.setFramesPerBurst(framesPerHardwareBurst);
+    mClockModel.setSampleRate(getDeviceSampleRate());
+    mClockModel.setFramesPerBurst(deviceFramesPerBurst);
 
     if (isDataCallbackSet()) {
         mCallbackFrames = callbackFrames;
@@ -311,7 +331,8 @@
         mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
     }
 
-    setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
+    // Default buffer size to match Q
+    setBufferSize(mBufferCapacityInFrames / 2);
     return AAUDIO_OK;
 }
 
@@ -370,9 +391,9 @@
     // Cache the buffer size which may be from client.
     const int32_t previousBufferSize = mBufferSizeInFrames;
     // Copy all available data from current data queue.
-    uint8_t buffer[getBufferCapacity() * getBytesPerFrame()];
-    android::fifo_frames_t fullFramesAvailable =
-            mAudioEndpoint->read(buffer, getBufferCapacity());
+    uint8_t buffer[getDeviceBufferCapacity() * getBytesPerFrame()];
+    android::fifo_frames_t fullFramesAvailable = mAudioEndpoint->read(buffer,
+            getDeviceBufferCapacity());
     mEndPointParcelable.closeDataFileDescriptor();
     aaudio_result_t result = mServiceInterface.exitStandby(
             mServiceStreamHandleInfo, endpointParcelable);
@@ -404,7 +425,7 @@
         goto exit;
     }
     // Write data from previous data buffer to new endpoint.
-    if (android::fifo_frames_t framesWritten =
+    if (const android::fifo_frames_t framesWritten =
                 mAudioEndpoint->write(buffer, fullFramesAvailable);
             framesWritten != fullFramesAvailable) {
         ALOGW("Some data lost after exiting standby, frames written: %d, "
@@ -444,7 +465,7 @@
         ALOGD("requestStart() but DISCONNECTED");
         return AAUDIO_ERROR_DISCONNECTED;
     }
-    aaudio_stream_state_t originalState = getState();
+    const aaudio_stream_state_t originalState = getState();
     setState(AAUDIO_STREAM_STATE_STARTING);
 
     // Clear any stale timestamps from the previous run.
@@ -601,7 +622,11 @@
     // Generated in server and passed to client. Return latest.
     if (mAtomicInternalTimestamp.isValid()) {
         Timestamp timestamp = mAtomicInternalTimestamp.read();
-        int64_t position = timestamp.getPosition() + mFramesOffsetFromService;
+        // This should not overflow as timestamp.getPosition() should be a position in a buffer and
+        // not the actual timestamp. timestamp.getNanoseconds() below uses the actual timestamp.
+        // At 48000 Hz we can run for over 100 years before overflowing the int64_t.
+        int64_t position = (timestamp.getPosition() + mFramesOffsetFromService) * getSampleRate() /
+                getDeviceSampleRate();
         if (position >= 0) {
             *framePosition = position;
             *timeNanoseconds = timestamp.getNanoseconds();
@@ -885,7 +910,8 @@
         adjustedFrames = maximumSize;
     } else {
         // Round to the next highest burst size.
-        int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
+        int32_t numBursts = (static_cast<int64_t>(adjustedFrames) + getFramesPerBurst() - 1) /
+                getFramesPerBurst();
         adjustedFrames = numBursts * getFramesPerBurst();
         // Clip just in case maximumSize is not a multiple of getFramesPerBurst().
         adjustedFrames = std::min(maximumSize, adjustedFrames);
@@ -893,23 +919,32 @@
 
     if (mAudioEndpoint) {
         // Clip against the actual size from the endpoint.
-        int32_t actualFrames = 0;
+        int32_t actualFramesDevice = 0;
+        int32_t maximumFramesDevice = (static_cast<int64_t>(maximumSize) * getDeviceSampleRate()
+                + getSampleRate() - 1) / getSampleRate();
         // Set to maximum size so we can write extra data when ready in order to reduce glitches.
         // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
-        mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);
+        mAudioEndpoint->setBufferSizeInFrames(maximumFramesDevice, &actualFramesDevice);
+        int32_t actualFrames = (static_cast<int64_t>(actualFramesDevice) * getSampleRate() +
+                 getDeviceSampleRate() - 1) / getDeviceSampleRate();
         // actualFrames should be <= actual maximum size of endpoint
         adjustedFrames = std::min(actualFrames, adjustedFrames);
     }
 
-    if (adjustedFrames != mBufferSizeInFrames) {
+    const int32_t bufferSizeInFrames = adjustedFrames;
+    const int32_t deviceBufferSizeInFrames = static_cast<int64_t>(bufferSizeInFrames) *
+            getDeviceSampleRate() / getSampleRate();
+
+    if (deviceBufferSizeInFrames != mDeviceBufferSizeInFrames) {
         android::mediametrics::LogItem(mMetricsId)
                 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_SETBUFFERSIZE)
-                .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, adjustedFrames)
+                .set(AMEDIAMETRICS_PROP_BUFFERSIZEFRAMES, deviceBufferSizeInFrames)
                 .set(AMEDIAMETRICS_PROP_UNDERRUN, (int32_t) getXRunCount())
                 .record();
     }
 
-    mBufferSizeInFrames = adjustedFrames;
+    mBufferSizeInFrames = bufferSizeInFrames;
+    mDeviceBufferSizeInFrames = deviceBufferSizeInFrames;
     ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
     return (aaudio_result_t) adjustedFrames;
 }
@@ -918,10 +953,18 @@
     return mBufferSizeInFrames;
 }
 
+int32_t AudioStreamInternal::getDeviceBufferSize() const {
+    return mDeviceBufferSizeInFrames;
+}
+
 int32_t AudioStreamInternal::getBufferCapacity() const {
     return mBufferCapacityInFrames;
 }
 
+int32_t AudioStreamInternal::getDeviceBufferCapacity() const {
+    return mDeviceBufferCapacityInFrames;
+}
+
 bool AudioStreamInternal::isClockModelInControl() const {
     return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
 }
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 9c06121..0dc9995 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -22,8 +22,9 @@
 
 #include "binding/AudioEndpointParcelable.h"
 #include "binding/AAudioServiceInterface.h"
-#include "client/IsochronousClockModel.h"
+#include "client/AAudioFlowGraph.h"
 #include "client/AudioEndpoint.h"
+#include "client/IsochronousClockModel.h"
 #include "core/AudioStream.h"
 #include "utility/AudioClock.h"
 
@@ -56,8 +57,12 @@
 
     int32_t getBufferSize() const override;
 
+    int32_t getDeviceBufferSize() const;
+
     int32_t getBufferCapacity() const override;
 
+    int32_t getDeviceBufferCapacity() const override;
+
     int32_t getXRunCount() const override {
         return mXRunCount;
     }
@@ -177,6 +182,8 @@
     int64_t                  mLastFramesWritten = 0;
     int64_t                  mLastFramesRead = 0;
 
+    AAudioFlowGraph          mFlowGraph;
+
 private:
     /*
      * Asynchronous write with data conversion.
@@ -211,8 +218,9 @@
     int32_t                  mDeviceChannelCount = 0;
 
     int32_t                  mBufferSizeInFrames = 0; // local threshold to control latency
+    int32_t                  mDeviceBufferSizeInFrames = 0;
     int32_t                  mBufferCapacityInFrames = 0;
-
+    int32_t                  mDeviceBufferCapacityInFrames = 0;
 
 };
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index f5cc2be..47518d7 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -47,6 +47,27 @@
 
 }
 
+aaudio_result_t AudioStreamInternalCapture::open(const AudioStreamBuilder &builder) {
+    aaudio_result_t result = AudioStreamInternal::open(builder);
+    if (result == AAUDIO_OK) {
+        result = mFlowGraph.configure(getDeviceFormat(),
+                             getDeviceChannelCount(),
+                             getDeviceSampleRate(),
+                             getFormat(),
+                             getSamplesPerFrame(),
+                             getSampleRate(),
+                             getRequireMonoBlend(),
+                             false /* useVolumeRamps */,
+                             getAudioBalance(),
+                             aaudio::resampler::MultiChannelResampler::Quality::Medium);
+
+        if (result != AAUDIO_OK) {
+            safeReleaseClose();
+        }
+    }
+    return result;
+}
+
 void AudioStreamInternalCapture::advanceClientToMatchServerPosition(int32_t serverMargin) {
     int64_t readCounter = mAudioEndpoint->getDataReadCounter();
     int64_t writeCounter = mAudioEndpoint->getDataWriteCounter() + serverMargin;
@@ -149,7 +170,8 @@
                 // Calculate frame position based off of the readCounter because
                 // the writeCounter might have just advanced in the background,
                 // causing us to sleep until a later burst.
-                int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + getFramesPerBurst();
+                const int64_t nextPosition = mAudioEndpoint->getDataReadCounter() +
+                        getDeviceFramesPerBurst();
                 wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
             }
                 break;
@@ -166,42 +188,73 @@
 
 aaudio_result_t AudioStreamInternalCapture::readNowWithConversion(void *buffer,
                                                                 int32_t numFrames) {
-    // ALOGD("readNowWithConversion(%p, %d)",
-    //              buffer, numFrames);
     WrappingBuffer wrappingBuffer;
-    uint8_t *destination = (uint8_t *) buffer;
-    int32_t framesLeft = numFrames;
+    uint8_t *byteBuffer = (uint8_t *) buffer;
+    int32_t framesLeftInByteBuffer = numFrames;
+
+    if (framesLeftInByteBuffer > 0) {
+        // Pull data from the flowgraph in case there is residual data.
+        const int32_t framesActuallyWrittenToByteBuffer = mFlowGraph.pull(
+                (void *)byteBuffer,
+                framesLeftInByteBuffer);
+
+        const int32_t numBytesActuallyWrittenToByteBuffer =
+                framesActuallyWrittenToByteBuffer * getBytesPerFrame();
+        byteBuffer += numBytesActuallyWrittenToByteBuffer;
+        framesLeftInByteBuffer -= framesActuallyWrittenToByteBuffer;
+    }
 
     mAudioEndpoint->getFullFramesAvailable(&wrappingBuffer);
 
-    // Read data in one or two parts.
-    for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
-        int32_t framesToProcess = framesLeft;
-        const int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
-        if (framesAvailable <= 0) break;
+    // Write data in one or two parts.
+    int partIndex = 0;
+    int framesReadFromAudioEndpoint = 0;
+    while (framesLeftInByteBuffer > 0 && partIndex < WrappingBuffer::SIZE) {
+        const int32_t totalFramesInWrappingBuffer = wrappingBuffer.numFrames[partIndex];
+        int32_t framesAvailableInWrappingBuffer = totalFramesInWrappingBuffer;
+        uint8_t *currentWrappingBuffer = (uint8_t *) wrappingBuffer.data[partIndex];
 
-        if (framesToProcess > framesAvailable) {
-            framesToProcess = framesAvailable;
+        // Put data from the wrapping buffer into the flowgraph 8 frames at a time.
+        // Continuously pull as much data as possible from the flowgraph into the byte buffer.
+        // The return value of mFlowGraph.process is the number of frames actually pulled.
+        while (framesAvailableInWrappingBuffer > 0 && framesLeftInByteBuffer > 0) {
+            const int32_t framesToReadFromWrappingBuffer = std::min(flowgraph::kDefaultBufferSize,
+                    framesAvailableInWrappingBuffer);
+
+            const int32_t numBytesToReadFromWrappingBuffer = getBytesPerDeviceFrame() *
+                    framesToReadFromWrappingBuffer;
+
+            // If framesActuallyWrittenToByteBuffer < framesLeftInByteBuffer, it is guaranteed
+            // that all the data is pulled. If there is no more space in the byteBuffer, the
+            // remaining data will be pulled in the following readNowWithConversion().
+            const int32_t framesActuallyWrittenToByteBuffer = mFlowGraph.process(
+                    (void *)currentWrappingBuffer,
+                    framesToReadFromWrappingBuffer,
+                    (void *)byteBuffer,
+                    framesLeftInByteBuffer);
+
+            const int32_t numBytesActuallyWrittenToByteBuffer =
+                    framesActuallyWrittenToByteBuffer * getBytesPerFrame();
+            byteBuffer += numBytesActuallyWrittenToByteBuffer;
+            framesLeftInByteBuffer -= framesActuallyWrittenToByteBuffer;
+            currentWrappingBuffer += numBytesToReadFromWrappingBuffer;
+            framesAvailableInWrappingBuffer -= framesToReadFromWrappingBuffer;
+
+            //ALOGD("%s() numBytesActuallyWrittenToByteBuffer %d, framesLeftInByteBuffer %d"
+            //      "framesAvailableInWrappingBuffer %d, framesReadFromAudioEndpoint %d"
+            //      , __func__, numBytesActuallyWrittenToByteBuffer, framesLeftInByteBuffer,
+            //      framesAvailableInWrappingBuffer, framesReadFromAudioEndpoint);
         }
-
-        const int32_t numBytes = getBytesPerFrame() * framesToProcess;
-        const int32_t numSamples = framesToProcess * getSamplesPerFrame();
-
-        const audio_format_t sourceFormat = getDeviceFormat();
-        const audio_format_t destinationFormat = getFormat();
-
-        memcpy_by_audio_format(destination, destinationFormat,
-                wrappingBuffer.data[partIndex], sourceFormat, numSamples);
-
-        destination += numBytes;
-        framesLeft -= framesToProcess;
+        framesReadFromAudioEndpoint += totalFramesInWrappingBuffer -
+                framesAvailableInWrappingBuffer;
+        partIndex++;
     }
 
-    int32_t framesProcessed = numFrames - framesLeft;
-    mAudioEndpoint->advanceReadIndex(framesProcessed);
+    // The audio endpoint should reference the number of frames written to the wrapping buffer.
+    mAudioEndpoint->advanceReadIndex(framesReadFromAudioEndpoint);
 
-    //ALOGD("readNowWithConversion() returns %d", framesProcessed);
-    return framesProcessed;
+    // The internal code should use the number of frames read from the app.
+    return numFrames - framesLeftInByteBuffer;
 }
 
 int64_t AudioStreamInternalCapture::getFramesWritten() {
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.h b/media/libaaudio/src/client/AudioStreamInternalCapture.h
index 87017de..10e247d 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.h
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.h
@@ -32,6 +32,8 @@
                                         bool inService = false);
     virtual ~AudioStreamInternalCapture() = default;
 
+    aaudio_result_t open(const AudioStreamBuilder &builder) override;
+
     aaudio_result_t read(void *buffer,
                          int32_t numFrames,
                          int64_t timeoutNanoseconds) override;
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 89dd8ff..99aa910 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -48,14 +48,18 @@
 
 aaudio_result_t AudioStreamInternalPlay::open(const AudioStreamBuilder &builder) {
     aaudio_result_t result = AudioStreamInternal::open(builder);
+    const bool useVolumeRamps = (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE);
     if (result == AAUDIO_OK) {
         result = mFlowGraph.configure(getFormat(),
                              getSamplesPerFrame(),
+                             getSampleRate(),
                              getDeviceFormat(),
                              getDeviceChannelCount(),
+                             getDeviceSampleRate(),
                              getRequireMonoBlend(),
+                             useVolumeRamps,
                              getAudioBalance(),
-                             (getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE));
+                             aaudio::resampler::MultiChannelResampler::Quality::Medium);
 
         if (result != AAUDIO_OK) {
             safeReleaseClose();
@@ -186,7 +190,7 @@
     // Sleep if there is too much data in the buffer.
     // Calculate an ideal time to wake up.
     if (wakeTimePtr != nullptr
-            && (mAudioEndpoint->getFullFramesAvailable() >= getBufferSize())) {
+            && (mAudioEndpoint->getFullFramesAvailable() >= getDeviceBufferSize())) {
         // By default wake up a few milliseconds from now.  // TODO review
         int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
         aaudio_stream_state_t state = getState();
@@ -206,12 +210,12 @@
                 // If the appBufferSize is smaller than the endpointBufferSize then
                 // we will have room to write data beyond the appBufferSize.
                 // That is a technique used to reduce glitches without adding latency.
-                const int32_t appBufferSize = getBufferSize();
+                const int64_t appBufferSize = getDeviceBufferSize();
                 // The endpoint buffer size is set to the maximum that can be written.
                 // If we use it then we must carve out some room to write data when we wake up.
-                const int32_t endBufferSize = mAudioEndpoint->getBufferSizeInFrames()
-                        - getFramesPerBurst();
-                const int32_t bestBufferSize = std::min(appBufferSize, endBufferSize);
+                const int64_t endBufferSize = mAudioEndpoint->getBufferSizeInFrames()
+                        - getDeviceFramesPerBurst();
+                const int64_t bestBufferSize = std::min(appBufferSize, endBufferSize);
                 int64_t targetReadPosition = mAudioEndpoint->getDataWriteCounter() - bestBufferSize;
                 wakeTime = mClockModel.convertPositionToTime(targetReadPosition);
             }
@@ -232,37 +236,78 @@
                                                             int32_t numFrames) {
     WrappingBuffer wrappingBuffer;
     uint8_t *byteBuffer = (uint8_t *) buffer;
-    int32_t framesLeft = numFrames;
+    int32_t framesLeftInByteBuffer = numFrames;
 
     mAudioEndpoint->getEmptyFramesAvailable(&wrappingBuffer);
 
     // Write data in one or two parts.
     int partIndex = 0;
-    while (framesLeft > 0 && partIndex < WrappingBuffer::SIZE) {
-        int32_t framesToWrite = framesLeft;
-        int32_t framesAvailable = wrappingBuffer.numFrames[partIndex];
-        if (framesAvailable > 0) {
-            if (framesToWrite > framesAvailable) {
-                framesToWrite = framesAvailable;
-            }
+    int framesWrittenToAudioEndpoint = 0;
+    while (framesLeftInByteBuffer > 0 && partIndex < WrappingBuffer::SIZE) {
+        int32_t framesAvailableInWrappingBuffer = wrappingBuffer.numFrames[partIndex];
+        uint8_t *currentWrappingBuffer = (uint8_t *) wrappingBuffer.data[partIndex];
 
-            int32_t numBytes = getBytesPerFrame() * framesToWrite;
+        if (framesAvailableInWrappingBuffer > 0) {
+            // Pull data from the flowgraph in case there is residual data.
+            const int32_t framesActuallyWrittenToWrappingBuffer = mFlowGraph.pull(
+                (void*) currentWrappingBuffer,
+                framesAvailableInWrappingBuffer);
 
-            mFlowGraph.process((void *)byteBuffer,
-                               wrappingBuffer.data[partIndex],
-                               framesToWrite);
+            const int32_t numBytesActuallyWrittenToWrappingBuffer =
+                framesActuallyWrittenToWrappingBuffer * getBytesPerDeviceFrame();
+            currentWrappingBuffer += numBytesActuallyWrittenToWrappingBuffer;
+            framesAvailableInWrappingBuffer -= framesActuallyWrittenToWrappingBuffer;
+            framesWrittenToAudioEndpoint += framesActuallyWrittenToWrappingBuffer;
+        }
 
-            byteBuffer += numBytes;
-            framesLeft -= framesToWrite;
-        } else {
-            break;
+        // Put data from byteBuffer into the flowgraph one buffer (8 frames) at a time.
+        // Continuously pull as much data as possible from the flowgraph into the wrapping buffer.
+        // The return value of mFlowGraph.process is the number of frames actually pulled.
+        while (framesAvailableInWrappingBuffer > 0 && framesLeftInByteBuffer > 0) {
+            const int32_t framesToWriteFromByteBuffer = std::min(flowgraph::kDefaultBufferSize,
+                    framesLeftInByteBuffer);
+
+            const int32_t numBytesToWriteFromByteBuffer = getBytesPerFrame() *
+                    framesToWriteFromByteBuffer;
+
+            //ALOGD("%s() framesLeftInByteBuffer %d, framesAvailableInWrappingBuffer %d"
+            //      "framesToWriteFromByteBuffer %d, numBytesToWriteFromByteBuffer %d"
+            //      , __func__, framesLeftInByteBuffer, framesAvailableInWrappingBuffer,
+            //      framesToWriteFromByteBuffer, numBytesToWriteFromByteBuffer);
+
+            const int32_t framesActuallyWrittenToWrappingBuffer = mFlowGraph.process(
+                    (void *)byteBuffer,
+                    framesToWriteFromByteBuffer,
+                    (void *)currentWrappingBuffer,
+                    framesAvailableInWrappingBuffer);
+
+            byteBuffer += numBytesToWriteFromByteBuffer;
+            framesLeftInByteBuffer -= framesToWriteFromByteBuffer;
+            const int32_t numBytesActuallyWrittenToWrappingBuffer =
+                    framesActuallyWrittenToWrappingBuffer * getBytesPerDeviceFrame();
+            currentWrappingBuffer += numBytesActuallyWrittenToWrappingBuffer;
+            framesAvailableInWrappingBuffer -= framesActuallyWrittenToWrappingBuffer;
+            framesWrittenToAudioEndpoint += framesActuallyWrittenToWrappingBuffer;
+
+            //ALOGD("%s() numBytesActuallyWrittenToWrappingBuffer %d, framesLeftInByteBuffer %d"
+            //      "framesActuallyWrittenToWrappingBuffer %d, numBytesToWriteFromByteBuffer %d"
+            //      "framesWrittenToAudioEndpoint %d"
+            //      , __func__, numBytesActuallyWrittenToWrappingBuffer, framesLeftInByteBuffer,
+            //      framesActuallyWrittenToWrappingBuffer, numBytesToWriteFromByteBuffer,
+            //      framesWrittenToAudioEndpoint);
         }
         partIndex++;
     }
-    int32_t framesWritten = numFrames - framesLeft;
-    mAudioEndpoint->advanceWriteIndex(framesWritten);
+    //ALOGD("%s() framesWrittenToAudioEndpoint %d, numFrames %d"
+    //              "framesLeftInByteBuffer %d"
+    //              , __func__, framesWrittenToAudioEndpoint, numFrames,
+    //              framesLeftInByteBuffer);
 
-    return framesWritten;
+    // The audio endpoint should reference the number of frames written to the wrapping buffer.
+    mAudioEndpoint->advanceWriteIndex(framesWrittenToAudioEndpoint);
+
+    // The internal code should use the number of frames read from the app.
+    return numFrames - framesLeftInByteBuffer;
 }
 
 int64_t AudioStreamInternalPlay::getFramesRead() {
@@ -284,7 +329,6 @@
     return mLastFramesWritten;
 }
 
-
 // Render audio in the application callback and then write the data to the stream.
 void *AudioStreamInternalPlay::callbackLoop() {
     ALOGD("%s() entering >>>>>>>>>>>>>>>", __func__);
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index e761807..b51b5d0 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -21,7 +21,6 @@
 #include <aaudio/AAudio.h>
 
 #include "binding/AAudioServiceInterface.h"
-#include "client/AAudioFlowGraph.h"
 #include "client/AudioStreamInternal.h"
 
 using android::sp;
@@ -89,13 +88,11 @@
      * Asynchronous write with data conversion.
      * @param buffer
      * @param numFrames
-     * @return fdrames written or negative error
+     * @return frames written or negative error
      */
     aaudio_result_t writeNowWithConversion(const void *buffer,
                                            int32_t numFrames);
 
-    AAudioFlowGraph          mFlowGraph;
-
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 8a13a6f..1e27a81 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -571,13 +571,15 @@
 AAUDIO_API int64_t AAudioStream_getFramesWritten(AAudioStream* stream)
 {
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
-    return audioStream->getFramesWritten();
+    return audioStream->getFramesWritten() * audioStream->getSampleRate() /
+            audioStream->getDeviceSampleRate();
 }
 
 AAUDIO_API int64_t AAudioStream_getFramesRead(AAudioStream* stream)
 {
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
-    return audioStream->getFramesRead();
+    return audioStream->getFramesRead() * audioStream->getSampleRate() /
+            audioStream->getDeviceSampleRate();
 }
 
 AAUDIO_API aaudio_result_t AAudioStream_getTimestamp(AAudioStream* stream,
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 9b4b734..1649eaf 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -204,10 +204,18 @@
         return mBufferCapacity;
     }
 
+    virtual int32_t getDeviceBufferCapacity() const {
+        return mDeviceBufferCapacity;
+    }
+
     virtual int32_t getFramesPerBurst() const {
         return mFramesPerBurst;
     }
 
+    virtual int32_t getDeviceFramesPerBurst() const {
+        return mDeviceFramesPerBurst;
+    }
+
     virtual int32_t getXRunCount() const {
         return AAUDIO_ERROR_UNIMPLEMENTED;
     }
@@ -224,6 +232,10 @@
         return mSampleRate;
     }
 
+    aaudio_result_t getDeviceSampleRate() const {
+        return mDeviceSampleRate;
+    }
+
     aaudio_result_t getHardwareSampleRate() const {
         return mHardwareSampleRate;
     }
@@ -542,6 +554,11 @@
     }
 
     // This should not be called after the open() call.
+    void setDeviceSampleRate(int32_t deviceSampleRate) {
+        mDeviceSampleRate = deviceSampleRate;
+    }
+
+    // This should not be called after the open() call.
     void setHardwareSampleRate(int32_t hardwareSampleRate) {
         mHardwareSampleRate = hardwareSampleRate;
     }
@@ -552,11 +569,21 @@
     }
 
     // This should not be called after the open() call.
+    void setDeviceFramesPerBurst(int32_t deviceFramesPerBurst) {
+        mDeviceFramesPerBurst = deviceFramesPerBurst;
+    }
+
+    // This should not be called after the open() call.
     void setBufferCapacity(int32_t bufferCapacity) {
         mBufferCapacity = bufferCapacity;
     }
 
     // This should not be called after the open() call.
+    void setDeviceBufferCapacity(int32_t deviceBufferCapacity) {
+        mDeviceBufferCapacity = deviceBufferCapacity;
+    }
+
+    // This should not be called after the open() call.
     void setSharingMode(aaudio_sharing_mode_t sharingMode) {
         mSharingMode = sharingMode;
     }
@@ -724,6 +751,7 @@
     int32_t                     mHardwareSamplesPerFrame = AAUDIO_UNSPECIFIED;
     aaudio_channel_mask_t       mChannelMask = AAUDIO_UNSPECIFIED;
     int32_t                     mSampleRate = AAUDIO_UNSPECIFIED;
+    int32_t                     mDeviceSampleRate = AAUDIO_UNSPECIFIED;
     int32_t                     mHardwareSampleRate = AAUDIO_UNSPECIFIED;
     int32_t                     mDeviceId = AAUDIO_UNSPECIFIED;
     aaudio_sharing_mode_t       mSharingMode = AAUDIO_SHARING_MODE_SHARED;
@@ -732,7 +760,9 @@
     audio_format_t              mHardwareFormat = AUDIO_FORMAT_DEFAULT;
     aaudio_performance_mode_t   mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
     int32_t                     mFramesPerBurst = 0;
+    int32_t                     mDeviceFramesPerBurst = 0;
     int32_t                     mBufferCapacity = 0;
+    int32_t                     mDeviceBufferCapacity = 0;
 
     aaudio_usage_t              mUsage           = AAUDIO_UNSPECIFIED;
     aaudio_content_type_t       mContentType     = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index e760dab..7b4821f 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -208,6 +208,10 @@
     setBufferCapacity(getBufferCapacityFromDevice());
     setFramesPerBurst(getFramesPerBurstFromDevice());
 
+    setDeviceSampleRate(mAudioRecord->getSampleRate());
+    setDeviceBufferCapacity(getBufferCapacityFromDevice());
+    setDeviceFramesPerBurst(getFramesPerBurstFromDevice());
+
     setHardwareSamplesPerFrame(mAudioRecord->getHalChannelCount());
     setHardwareSampleRate(mAudioRecord->getHalSampleRate());
     setHardwareFormat(mAudioRecord->getHalFormat());
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 67ee42e..723b419 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -202,6 +202,9 @@
     setSampleRate(mAudioTrack->getSampleRate());
     setBufferCapacity(getBufferCapacityFromDevice());
     setFramesPerBurst(getFramesPerBurstFromDevice());
+    setDeviceSampleRate(mAudioTrack->getSampleRate());
+    setDeviceBufferCapacity(getBufferCapacityFromDevice());
+    setDeviceFramesPerBurst(getFramesPerBurstFromDevice());
 
     setHardwareSamplesPerFrame(mAudioTrack->getHalChannelCount());
     setHardwareSampleRate(mAudioTrack->getHalSampleRate());
diff --git a/media/libaaudio/src/utility/AudioClock.h b/media/libaaudio/src/utility/AudioClock.h
index d5d4ef4..37f5b39 100644
--- a/media/libaaudio/src/utility/AudioClock.h
+++ b/media/libaaudio/src/utility/AudioClock.h
@@ -33,7 +33,7 @@
 public:
     static int64_t getNanoseconds(clockid_t clockId = CLOCK_MONOTONIC) {
         struct timespec time;
-        int result = clock_gettime(clockId, &time);
+        const int result = clock_gettime(clockId, &time);
         if (result < 0) {
             return -errno;
         }
@@ -56,7 +56,7 @@
             time.tv_sec = nanoTime / AAUDIO_NANOS_PER_SECOND;
             // Calculate the fractional nanoseconds. Avoids expensive % operation.
             time.tv_nsec = nanoTime - (time.tv_sec * AAUDIO_NANOS_PER_SECOND);
-            int err = clock_nanosleep(clockId, TIMER_ABSTIME, &time, nullptr);
+            const int err = clock_nanosleep(clockId, TIMER_ABSTIME, &time, nullptr);
             switch (err) {
             case EINTR:
                 return 1;
@@ -86,7 +86,7 @@
             // Calculate the fractional nanoseconds. Avoids expensive % operation.
             time.tv_nsec = nanoseconds - (time.tv_sec * AAUDIO_NANOS_PER_SECOND);
             const int flags = 0; // documented as relative sleep
-            int err = clock_nanosleep(clockId, flags, &time, nullptr);
+            const int err = clock_nanosleep(clockId, flags, &time, nullptr);
             switch (err) {
             case EINTR:
                 return 1;
diff --git a/media/libaaudio/src/utility/MonotonicCounter.h b/media/libaaudio/src/utility/MonotonicCounter.h
index 51eb69b..b58634f 100644
--- a/media/libaaudio/src/utility/MonotonicCounter.h
+++ b/media/libaaudio/src/utility/MonotonicCounter.h
@@ -104,7 +104,7 @@
      */
     void roundUp64(int32_t period) {
         if (period > 0) {
-            int64_t numPeriods = (mCounter64 + period - 1) / period;
+            const int64_t numPeriods = (mCounter64 + period - 1) / period;
             mCounter64 = numPeriods * period;
         }
     }
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index 24041bc..0cfdfb2 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -145,6 +145,7 @@
     srcs: ["test_flowgraph.cpp"],
     shared_libs: [
         "libaaudio_internal",
+        "libaudioutils",
         "libbinder",
         "libcutils",
         "libutils",
diff --git a/media/libaaudio/tests/test_flowgraph.cpp b/media/libaaudio/tests/test_flowgraph.cpp
index 6f75f5a..7eb8b0d 100644
--- a/media/libaaudio/tests/test_flowgraph.cpp
+++ b/media/libaaudio/tests/test_flowgraph.cpp
@@ -25,6 +25,8 @@
 
 #include <gtest/gtest.h>
 
+#include <aaudio/AAudio.h>
+#include "client/AAudioFlowGraph.h"
 #include "flowgraph/ClipToRange.h"
 #include "flowgraph/Limiter.h"
 #include "flowgraph/MonoBlend.h"
@@ -37,8 +39,18 @@
 #include "flowgraph/SinkI32.h"
 #include "flowgraph/SourceI16.h"
 #include "flowgraph/SourceI24.h"
+#include "flowgraph/resampler/IntegerRatio.h"
 
 using namespace FLOWGRAPH_OUTER_NAMESPACE::flowgraph;
+using namespace RESAMPLER_OUTER_NAMESPACE::resampler;
+
+using TestFlowgraphResamplerParams = std::tuple<int32_t, int32_t, MultiChannelResampler::Quality>;
+
+enum {
+    PARAM_SOURCE_SAMPLE_RATE = 0,
+    PARAM_SINK_SAMPLE_RATE,
+    PARAM_RESAMPLER_QUALITY
+};
 
 constexpr int kBytesPerI24Packed = 3;
 
@@ -394,3 +406,240 @@
         EXPECT_NEAR(expected[i], output[i], tolerance);
     }
 }
+
+TEST(test_flowgraph, module_sinki16_multiple_reads) {
+    static constexpr int kNumSamples = 8;
+    std::array<int16_t, kNumSamples + 10> output; // larger than input
+
+    SourceFloat sourceFloat{1};
+    SinkI16 sinkI16{1};
+
+    sourceFloat.setData(kInputFloat.data(), kNumSamples);
+    sourceFloat.output.connect(&sinkI16.input);
+
+    output.fill(777);
+
+    // Read the first half of the data
+    int32_t numRead = sinkI16.read(output.data(), kNumSamples / 2);
+    ASSERT_EQ(kNumSamples / 2, numRead);
+    for (int i = 0; i < numRead; i++) {
+        EXPECT_EQ(kExpectedI16.at(i), output.at(i)) << ", i = " << i;
+    }
+
+    // Read the rest of the data
+    numRead = sinkI16.read(output.data(), output.size());
+    ASSERT_EQ(kNumSamples / 2, numRead);
+    for (int i = 0; i < numRead; i++) {
+        EXPECT_EQ(kExpectedI16.at(i + kNumSamples / 2), output.at(i)) << ", i = " << i;
+    }
+}
+
+void checkSampleRateConversionVariedSizes(int32_t sourceSampleRate,
+                    int32_t sinkSampleRate,
+                    MultiChannelResampler::Quality resamplerQuality) {
+    AAudioFlowGraph flowgraph;
+    aaudio_result_t result = flowgraph.configure(AUDIO_FORMAT_PCM_FLOAT /* sourceFormat */,
+            1 /* sourceChannelCount */,
+            sourceSampleRate,
+            AUDIO_FORMAT_PCM_FLOAT /* sinkFormat */,
+            1 /* sinkChannelCount */,
+            sinkSampleRate,
+            false /* useMonoBlend */,
+            false /* useVolumeRamps */,
+            0.0f /* audioBalance */,
+            resamplerQuality);
+
+    IntegerRatio ratio(sourceSampleRate, sinkSampleRate);
+    ratio.reduce();
+
+    ASSERT_EQ(AAUDIO_OK, result);
+
+    const int inputSize = ratio.getNumerator();
+    const int outputSize = ratio.getDenominator();
+    float input[inputSize];
+    float output[outputSize];
+
+    for (int i = 0; i < inputSize; i++) {
+        input[i] = i * 1.0f / inputSize;
+    }
+
+    int inputUsed = 0;
+    int outputRead = 0;
+    int curInputSize = 1;
+
+    // Process the data with larger and larger input buffer sizes.
+    while (inputUsed < inputSize) {
+        outputRead += flowgraph.process((void *) (input + inputUsed),
+                curInputSize,
+                (void *) (output + outputRead),
+                outputSize - outputRead);
+        inputUsed += curInputSize;
+        curInputSize = std::min(curInputSize + 5, inputSize - inputUsed);
+    }
+
+    ASSERT_EQ(outputSize, outputRead);
+
+    for (int i = 1; i < outputSize; i++) {
+        // The first values of the flowgraph will be close to zero.
+        // Besides those, the values should be strictly increasing.
+        if (output[i - 1] > 0.01f) {
+            EXPECT_GT(output[i], output[i - 1]);
+        }
+    }
+}
+
+TEST(test_flowgraph, flowgraph_varied_sizes_all) {
+    const int rates[] = {8000, 11025, 22050, 32000, 44100, 48000, 64000, 88200, 96000};
+    const MultiChannelResampler::Quality qualities[] =
+    {
+        MultiChannelResampler::Quality::Fastest,
+        MultiChannelResampler::Quality::Low,
+        MultiChannelResampler::Quality::Medium,
+        MultiChannelResampler::Quality::High,
+        MultiChannelResampler::Quality::Best
+    };
+    for (int srcRate : rates) {
+        for (int destRate : rates) {
+            for (auto quality : qualities) {
+                if (srcRate != destRate) {
+                    checkSampleRateConversionVariedSizes(srcRate, destRate, quality);
+                }
+            }
+        }
+    }
+}
+
+void checkSampleRateConversionPullLater(int32_t sourceSampleRate,
+                    int32_t sinkSampleRate,
+                    MultiChannelResampler::Quality resamplerQuality) {
+    AAudioFlowGraph flowgraph;
+    aaudio_result_t result = flowgraph.configure(AUDIO_FORMAT_PCM_FLOAT /* sourceFormat */,
+            1 /* sourceChannelCount */,
+            sourceSampleRate,
+            AUDIO_FORMAT_PCM_FLOAT /* sinkFormat */,
+            1 /* sinkChannelCount */,
+            sinkSampleRate,
+            false /* useMonoBlend */,
+            false /* useVolumeRamps */,
+            0.0f /* audioBalance */,
+            resamplerQuality);
+
+    IntegerRatio ratio(sourceSampleRate, sinkSampleRate);
+    ratio.reduce();
+
+    ASSERT_EQ(AAUDIO_OK, result);
+
+    const int inputSize = ratio.getNumerator();
+    const int outputSize = ratio.getDenominator();
+    float input[inputSize];
+    float output[outputSize];
+
+    for (int i = 0; i < inputSize; i++) {
+        input[i] = i * 1.0f / inputSize;
+    }
+
+    // Read half the data with process.
+    int outputRead = flowgraph.process((void *) input,
+            inputSize,
+            (void *) output,
+            outputSize / 2);
+
+    ASSERT_EQ(outputSize / 2, outputRead);
+
+    // Now read the other half of the data with pull.
+    outputRead += flowgraph.pull(
+            (void *) (output + outputRead),
+            outputSize - outputRead);
+
+    ASSERT_EQ(outputSize, outputRead);
+    for (int i = 1; i < outputSize; i++) {
+        // The first values of the flowgraph will be close to zero.
+        // Besides those, the values should be strictly increasing.
+        if (output[i - 1] > 0.01f) {
+            EXPECT_GT(output[i], output[i - 1]);
+        }
+    }
+}
+
+// TODO: b/289508408 - Remove non-parameterized tests if they get noisy.
+TEST(test_flowgraph, flowgraph_pull_later_all) {
+    const int rates[] = {8000, 11025, 22050, 32000, 44100, 48000, 64000, 88200, 96000};
+    const MultiChannelResampler::Quality qualities[] =
+    {
+        MultiChannelResampler::Quality::Fastest,
+        MultiChannelResampler::Quality::Low,
+        MultiChannelResampler::Quality::Medium,
+        MultiChannelResampler::Quality::High,
+        MultiChannelResampler::Quality::Best
+    };
+    for (int srcRate : rates) {
+        for (int destRate : rates) {
+            for (auto quality : qualities) {
+                if (srcRate != destRate) {
+                    checkSampleRateConversionPullLater(srcRate, destRate, quality);
+                }
+            }
+        }
+    }
+}
+
+class TestFlowgraphSampleRateConversion : public ::testing::Test,
+                        public ::testing::WithParamInterface<TestFlowgraphResamplerParams> {
+};
+
+const char* resamplerQualityToString(MultiChannelResampler::Quality quality) {
+    switch (quality) {
+        case MultiChannelResampler::Quality::Fastest: return "FASTEST";
+        case MultiChannelResampler::Quality::Low: return "LOW";
+        case MultiChannelResampler::Quality::Medium: return "MEDIUM";
+        case MultiChannelResampler::Quality::High: return "HIGH";
+        case MultiChannelResampler::Quality::Best: return "BEST";
+    }
+    return "UNKNOWN";
+}
+
+static std::string getTestName(
+        const ::testing::TestParamInfo<TestFlowgraphResamplerParams>& info) {
+    return std::string()
+            + std::to_string(std::get<PARAM_SOURCE_SAMPLE_RATE>(info.param))
+            + "__" + std::to_string(std::get<PARAM_SINK_SAMPLE_RATE>(info.param))
+            + "__" + resamplerQualityToString(std::get<PARAM_RESAMPLER_QUALITY>(info.param));
+}
+
+TEST_P(TestFlowgraphSampleRateConversion, test_flowgraph_pull_later) {
+    checkSampleRateConversionPullLater(std::get<PARAM_SOURCE_SAMPLE_RATE>(GetParam()),
+            std::get<PARAM_SINK_SAMPLE_RATE>(GetParam()),
+            std::get<PARAM_RESAMPLER_QUALITY>(GetParam()));
+}
+
+TEST_P(TestFlowgraphSampleRateConversion, test_flowgraph_varied_sizes) {
+    checkSampleRateConversionVariedSizes(std::get<PARAM_SOURCE_SAMPLE_RATE>(GetParam()),
+            std::get<PARAM_SINK_SAMPLE_RATE>(GetParam()),
+            std::get<PARAM_RESAMPLER_QUALITY>(GetParam()));
+}
+
+INSTANTIATE_TEST_SUITE_P(
+        test_flowgraph,
+        TestFlowgraphSampleRateConversion,
+        ::testing::Values(
+                TestFlowgraphResamplerParams({8000, 11025, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({8000, 48000, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({8000, 44100, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({11025, 24000, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({11025, 48000,
+                        MultiChannelResampler::Quality::Fastest}),
+                TestFlowgraphResamplerParams({11025, 48000, MultiChannelResampler::Quality::Low}),
+                TestFlowgraphResamplerParams({11025, 48000,
+                        MultiChannelResampler::Quality::Medium}),
+                TestFlowgraphResamplerParams({11025, 48000, MultiChannelResampler::Quality::High}),
+                TestFlowgraphResamplerParams({11025, 48000, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({11025, 44100, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({11025, 88200, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({16000, 48000, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({44100, 48000, MultiChannelResampler::Quality::Low}),
+                TestFlowgraphResamplerParams({44100, 48000, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({48000, 11025, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({48000, 44100, MultiChannelResampler::Quality::Best}),
+                TestFlowgraphResamplerParams({44100, 11025, MultiChannelResampler::Quality::Best})),
+        &getTestName
+);
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 7f228c7..2ef6fe5 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -45,6 +45,8 @@
 #define OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS  (3 * AAUDIO_NANOS_PER_MILLISECOND)
 #define INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS   (-1 * AAUDIO_NANOS_PER_MILLISECOND)
 
+#define AAUDIO_MAX_OPEN_ATTEMPTS    10
+
 using namespace android;  // TODO just import names needed
 using namespace aaudio;   // TODO just import names needed
 
@@ -77,14 +79,23 @@
         {AUDIO_FORMAT_PCM_24_BIT_PACKED, AUDIO_FORMAT_PCM_16_BIT}
 };
 
-audio_format_t getNextFormatToTry(audio_format_t curFormat, audio_format_t returnedFromAPM) {
-    if (returnedFromAPM != AUDIO_FORMAT_DEFAULT) {
-        return returnedFromAPM;
-    }
+audio_format_t getNextFormatToTry(audio_format_t curFormat) {
     const auto it = NEXT_FORMAT_TO_TRY.find(curFormat);
-    return it != NEXT_FORMAT_TO_TRY.end() ? it->second : AUDIO_FORMAT_DEFAULT;
+    return it != NEXT_FORMAT_TO_TRY.end() ? it->second : curFormat;
 }
 
+struct configComp {
+    bool operator() (const audio_config_base_t& lhs, const audio_config_base_t& rhs) const {
+        if (lhs.sample_rate != rhs.sample_rate) {
+            return lhs.sample_rate < rhs.sample_rate;
+        } else if (lhs.channel_mask != rhs.channel_mask) {
+            return lhs.channel_mask < rhs.channel_mask;
+        } else {
+            return lhs.format < rhs.format;
+        }
+    }
+};
+
 } // namespace
 
 aaudio_result_t AAudioServiceEndpointMMAP::open(const aaudio::AAudioStreamRequest &request) {
@@ -101,60 +112,66 @@
         legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
 
     audio_format_t audioFormat = getFormat();
-    std::set<audio_format_t> formatsTried;
-    while (true) {
-        if (formatsTried.find(audioFormat) != formatsTried.end()) {
+    int32_t sampleRate = getSampleRate();
+    if (sampleRate == AAUDIO_UNSPECIFIED) {
+        sampleRate = AAUDIO_SAMPLE_RATE_DEFAULT;
+    }
+
+    const aaudio_direction_t direction = getDirection();
+    audio_config_base_t config;
+    config.format = audioFormat;
+    config.sample_rate = sampleRate;
+    config.channel_mask = AAudio_getChannelMaskForOpen(
+            getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
+
+    std::set<audio_config_base_t, configComp> configsTried;
+    int32_t numberOfAttempts = 0;
+    while (numberOfAttempts < AAUDIO_MAX_OPEN_ATTEMPTS) {
+        if (configsTried.find(config) != configsTried.end()) {
             // APM returning something that has already tried.
-            ALOGW("Have already tried to open with format=%#x, but failed before", audioFormat);
+            ALOGW("Have already tried to open with format=%#x and sr=%d, but failed before",
+                  config.format, config.sample_rate);
             break;
         }
-        formatsTried.insert(audioFormat);
+        configsTried.insert(config);
 
-        audio_format_t nextFormatToTry = AUDIO_FORMAT_DEFAULT;
-        result = openWithFormat(audioFormat, &nextFormatToTry);
+        audio_config_base_t previousConfig = config;
+        result = openWithConfig(&config);
         if (result != AAUDIO_ERROR_UNAVAILABLE) {
             // Return if it is successful or there is an error that is not
             // AAUDIO_ERROR_UNAVAILABLE happens.
-            ALOGI("Opened format=%#x with result=%d", audioFormat, result);
+            ALOGI("Opened format=%#x sr=%d, with result=%d", previousConfig.format,
+                    previousConfig.sample_rate, result);
             break;
         }
 
-        nextFormatToTry = getNextFormatToTry(audioFormat, nextFormatToTry);
-        ALOGD("%s() %#x failed, perhaps due to format. Try again with %#x",
-              __func__, audioFormat, nextFormatToTry);
-        audioFormat = nextFormatToTry;
-        if (audioFormat == AUDIO_FORMAT_DEFAULT) {
-            // Nothing else to try
-            break;
+        // Try other formats if the config from APM is the same as our current config.
+        // Some HALs may report its format support incorrectly.
+        if ((previousConfig.format == config.format) &&
+                (previousConfig.sample_rate == config.sample_rate)) {
+            config.format = getNextFormatToTry(config.format);
         }
+
+        ALOGD("%s() %#x %d failed, perhaps due to format or sample rate. Try again with %#x %d",
+                __func__, previousConfig.format, previousConfig.sample_rate, config.format,
+                config.sample_rate);
+        numberOfAttempts++;
     }
     return result;
 }
 
-aaudio_result_t AAudioServiceEndpointMMAP::openWithFormat(
-        audio_format_t audioFormat, audio_format_t* nextFormatToTry) {
+aaudio_result_t AAudioServiceEndpointMMAP::openWithConfig(
+        audio_config_base_t* config) {
     aaudio_result_t result = AAUDIO_OK;
-    audio_config_base_t config;
+    audio_config_base_t currentConfig = *config;
     audio_port_handle_t deviceId;
 
     const audio_attributes_t attributes = getAudioAttributesFrom(this);
 
     deviceId = mRequestedDeviceId;
 
-    // Fill in config
-    config.format = audioFormat;
-
-    int32_t aaudioSampleRate = getSampleRate();
-    if (aaudioSampleRate == AAUDIO_UNSPECIFIED) {
-        aaudioSampleRate = AAUDIO_SAMPLE_RATE_DEFAULT;
-    }
-    config.sample_rate = aaudioSampleRate;
-
     const aaudio_direction_t direction = getDirection();
 
-    config.channel_mask = AAudio_getChannelMaskForOpen(
-            getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
-
     if (direction == AAUDIO_DIRECTION_OUTPUT) {
         mHardwareTimeOffsetNanos = OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at DAC later
 
@@ -177,11 +194,11 @@
     // Open HAL stream. Set mMmapStream
     ALOGD("%s trying to open MMAP stream with format=%#x, "
           "sample_rate=%u, channel_mask=%#x, device=%d",
-          __func__, config.format, config.sample_rate,
-          config.channel_mask, deviceId);
+          __func__, config->format, config->sample_rate,
+          config->channel_mask, deviceId);
     const status_t status = MmapStreamInterface::openMmapStream(streamDirection,
                                                                 &attributes,
-                                                                &config,
+                                                                config,
                                                                 mMmapClient,
                                                                 &deviceId,
                                                                 &sessionId,
@@ -195,9 +212,9 @@
         // not match the hardware.
         ALOGD("%s() - openMmapStream() returned status=%d, suggested format=%#x, sample_rate=%u, "
               "channel_mask=%#x",
-              __func__, status, config.format, config.sample_rate, config.channel_mask);
-        *nextFormatToTry = config.format != audioFormat ? config.format
-                                                        : *nextFormatToTry;
+              __func__, status, config->format, config->sample_rate, config->channel_mask);
+        // Keep the channel mask of the current config
+        config->channel_mask = currentConfig.channel_mask;
         return AAUDIO_ERROR_UNAVAILABLE;
     }
 
@@ -217,7 +234,7 @@
     setSessionId(actualSessionId);
 
     ALOGD("%s(format = 0x%X) deviceId = %d, sessionId = %d",
-          __func__, audioFormat, getDeviceId(), getSessionId());
+          __func__, config->format, getDeviceId(), getSessionId());
 
     // Create MMAP/NOIRQ buffer.
     result = createMmapBuffer();
@@ -227,11 +244,11 @@
 
     // Get information about the stream and pass it back to the caller.
     setChannelMask(AAudioConvert_androidToAAudioChannelMask(
-            config.channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
-            AAudio_isChannelIndexMask(config.channel_mask)));
+            config->channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
+            AAudio_isChannelIndexMask(config->channel_mask)));
 
-    setFormat(config.format);
-    setSampleRate(config.sample_rate);
+    setFormat(config->format);
+    setSampleRate(config->sample_rate);
     setHardwareSampleRate(getSampleRate());
     setHardwareFormat(getFormat());
     setHardwareSamplesPerFrame(AAudioConvert_channelMaskToCount(getChannelMask()));
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.h b/services/oboeservice/AAudioServiceEndpointMMAP.h
index 38cf0ba..f19005c 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.h
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.h
@@ -97,7 +97,7 @@
 
 private:
 
-    aaudio_result_t openWithFormat(audio_format_t audioFormat, audio_format_t* nextFormatToTry);
+    aaudio_result_t openWithConfig(audio_config_base_t* config);
 
     aaudio_result_t createMmapBuffer();