diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 2688597..94f10e5 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -210,10 +210,10 @@
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
     }
-    mFramesPerBurst = framesPerBurst; // only save good value
+    setFramesPerBurst(framesPerBurst); // only save good value
 
     mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
-    if (mBufferCapacityInFrames < mFramesPerBurst
+    if (mBufferCapacityInFrames < getFramesPerBurst()
             || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
         ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
         result = AAUDIO_ERROR_OUT_OF_RANGE;
@@ -238,7 +238,7 @@
 
         }
         if (mCallbackFrames == AAUDIO_UNSPECIFIED) {
-            mCallbackFrames = mFramesPerBurst;
+            mCallbackFrames = getFramesPerBurst();
         }
 
         const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
@@ -756,9 +756,9 @@
 
 aaudio_result_t AudioStreamInternal::setBufferSize(int32_t requestedFrames) {
     int32_t adjustedFrames = requestedFrames;
-    const int32_t maximumSize = getBufferCapacity() - mFramesPerBurst;
+    const int32_t maximumSize = getBufferCapacity() - getFramesPerBurst();
     // Minimum size should be a multiple number of bursts.
-    const int32_t minimumSize = 1 * mFramesPerBurst;
+    const int32_t minimumSize = 1 * getFramesPerBurst();
 
     // Clip to minimum size so that rounding up will work better.
     adjustedFrames = std::max(minimumSize, adjustedFrames);
@@ -768,9 +768,9 @@
         adjustedFrames = maximumSize;
     } else {
         // Round to the next highest burst size.
-        int32_t numBursts = (adjustedFrames + mFramesPerBurst - 1) / mFramesPerBurst;
-        adjustedFrames = numBursts * mFramesPerBurst;
-        // Clip just in case maximumSize is not a multiple of mFramesPerBurst.
+        int32_t numBursts = (adjustedFrames + getFramesPerBurst() - 1) / getFramesPerBurst();
+        adjustedFrames = numBursts * getFramesPerBurst();
+        // Clip just in case maximumSize is not a multiple of getFramesPerBurst().
         adjustedFrames = std::min(maximumSize, adjustedFrames);
     }
 
@@ -805,10 +805,6 @@
     return mBufferCapacityInFrames;
 }
 
-int32_t AudioStreamInternal::getFramesPerBurst() const {
-    return mFramesPerBurst;
-}
-
 // This must be called under mStreamLock.
 aaudio_result_t AudioStreamInternal::joinThread(void** returnArg) {
     return AudioStream::joinThread(returnArg, calculateReasonableTimeout(getFramesPerBurst()));
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 162f098..d7024cf 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -64,8 +64,6 @@
 
     int32_t getBufferCapacity() const override;
 
-    int32_t getFramesPerBurst() const override;
-
     int32_t getXRunCount() const override {
         return mXRunCount;
     }
@@ -159,7 +157,6 @@
 
     aaudio_handle_t          mServiceStreamHandle; // opaque handle returned from service
 
-    int32_t                  mFramesPerBurst = MIN_FRAMES_PER_BURST; // frames per HAL transfer
     int32_t                  mXRunCount = 0;      // how many underrun events?
 
     // Offset from underlying frame position.
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 55fc986..5d311fc 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -149,7 +149,7 @@
                 // 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() + mFramesPerBurst;
+                int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + getFramesPerBurst();
                 wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
             }
                 break;
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index e0bd9d8..e438477 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -202,11 +202,11 @@
     }
 
     virtual int32_t getBufferCapacity() const {
-        return AAUDIO_ERROR_UNIMPLEMENTED;
+        return mBufferCapacity;
     }
 
     virtual int32_t getFramesPerBurst() const {
-        return AAUDIO_ERROR_UNIMPLEMENTED;
+        return mFramesPerBurst;
     }
 
     virtual int32_t getXRunCount() const {
@@ -498,30 +498,32 @@
         mSampleRate = sampleRate;
     }
 
-    /**
-     * This should not be called after the open() call.
-     */
+    // This should not be called after the open() call.
     void setSamplesPerFrame(int32_t samplesPerFrame) {
         mSamplesPerFrame = samplesPerFrame;
     }
 
-    /**
-     * This should not be called after the open() call.
-     */
+    // This should not be called after the open() call.
+    void setFramesPerBurst(int32_t framesPerBurst) {
+        mFramesPerBurst = framesPerBurst;
+    }
+
+    // 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 setSharingMode(aaudio_sharing_mode_t sharingMode) {
         mSharingMode = sharingMode;
     }
 
-    /**
-     * This should not be called after the open() call.
-     */
+    // This should not be called after the open() call.
     void setFormat(audio_format_t format) {
         mFormat = format;
     }
 
-    /**
-     * This should not be called after the open() call.
-     */
+    // This should not be called after the open() call.
     void setDeviceFormat(audio_format_t format) {
         mDeviceFormat = format;
     }
@@ -536,6 +538,7 @@
         mDeviceId = deviceId;
     }
 
+    // This should not be called after the open() call.
     void setSessionId(int32_t sessionId) {
         mSessionId = sessionId;
     }
@@ -623,6 +626,8 @@
     audio_format_t              mFormat = AUDIO_FORMAT_DEFAULT;
     aaudio_stream_state_t       mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
     aaudio_performance_mode_t   mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
+    int32_t                     mFramesPerBurst = 0;
+    int32_t                     mBufferCapacity = 0;
 
     aaudio_usage_t              mUsage           = AAUDIO_UNSPECIFIED;
     aaudio_content_type_t       mContentType     = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index fefe6e0..88ef270 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -112,6 +112,18 @@
         return mFramesRead.increment(frames);
     }
 
+    /**
+     * Get the framesPerBurst from the underlying API.
+     * @return framesPerBurst
+     */
+    virtual int32_t getFramesPerBurstFromDevice() const = 0;
+
+    /**
+     * Get the bufferCapacity from the underlying API.
+     * @return bufferCapacity in frames
+     */
+    virtual int32_t getBufferCapacityFromDevice() const = 0;
+
     // This is used for exact matching by MediaMetrics. So do not change it.
     // MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_AAUDIO
     static constexpr char     kCallerName[] = "aaudio";
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index a8ae0fb..d46ef56 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -210,9 +210,9 @@
 
     // Get the actual values from the AudioRecord.
     setSamplesPerFrame(mAudioRecord->channelCount());
-
-    int32_t actualSampleRate = mAudioRecord->getSampleRate();
-    setSampleRate(actualSampleRate);
+    setSampleRate(mAudioRecord->getSampleRate());
+    setBufferCapacity(getBufferCapacityFromDevice());
+    setFramesPerBurst(getFramesPerBurstFromDevice());
 
     // We may need to pass the data through a block size adapter to guarantee constant size.
     if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
@@ -488,7 +488,7 @@
     return getBufferCapacity(); // TODO implement in AudioRecord?
 }
 
-int32_t AudioStreamRecord::getBufferCapacity() const
+int32_t AudioStreamRecord::getBufferCapacityFromDevice() const
 {
     return static_cast<int32_t>(mAudioRecord->frameCount());
 }
@@ -498,8 +498,7 @@
     return 0; // TODO implement when AudioRecord supports it
 }
 
-int32_t AudioStreamRecord::getFramesPerBurst() const
-{
+int32_t AudioStreamRecord::getFramesPerBurstFromDevice() const {
     return static_cast<int32_t>(mAudioRecord->getNotificationPeriodInFrames());
 }
 
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index e4ef1c0..ad8dfe4 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -56,14 +56,10 @@
 
     int32_t getBufferSize() const override;
 
-    int32_t getBufferCapacity() const override;
-
     int32_t getXRunCount() const override;
 
     int64_t getFramesWritten() override;
 
-    int32_t getFramesPerBurst() const override;
-
     aaudio_result_t updateStateMachine() override;
 
     aaudio_direction_t getDirection() const override {
@@ -79,6 +75,11 @@
 
     const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) override;
 
+protected:
+
+    int32_t getFramesPerBurstFromDevice() const override;
+    int32_t getBufferCapacityFromDevice() const override;
+
 private:
     android::sp<android::AudioRecord> mAudioRecord;
     // adapts between variable sized blocks and fixed size blocks
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 4ba08fd..307904e 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -192,9 +192,9 @@
     setSamplesPerFrame(mAudioTrack->channelCount());
     setFormat(mAudioTrack->format());
     setDeviceFormat(mAudioTrack->format());
-
-    int32_t actualSampleRate = mAudioTrack->getSampleRate();
-    setSampleRate(actualSampleRate);
+    setSampleRate(mAudioTrack->getSampleRate());
+    setBufferCapacity(getBufferCapacityFromDevice());
+    setFramesPerBurst(getFramesPerBurstFromDevice());
 
     // We may need to pass the data through a block size adapter to guarantee constant size.
     if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) {
@@ -217,9 +217,6 @@
             : (aaudio_session_id_t) mAudioTrack->getSessionId();
     setSessionId(actualSessionId);
 
-    mInitialBufferCapacity = getBufferCapacity();
-    mInitialFramesPerBurst = getFramesPerBurst();
-
     mAudioTrack->addAudioDeviceCallback(this);
 
     // Update performance mode based on the actual stream flags.
@@ -284,8 +281,8 @@
                     || mAudioTrack->format() != getFormat()
                     || mAudioTrack->getSampleRate() != getSampleRate()
                     || mAudioTrack->getRoutedDeviceId() != getDeviceId()
-                    || getBufferCapacity() != mInitialBufferCapacity
-                    || getFramesPerBurst() != mInitialFramesPerBurst) {
+                    || getBufferCapacityFromDevice() != getBufferCapacity()
+                    || getFramesPerBurstFromDevice() != getFramesPerBurst()) {
                 processCallbackCommon(AAUDIO_CALLBACK_OPERATION_DISCONNECTED, info);
             }
             break;
@@ -474,7 +471,7 @@
     return static_cast<int32_t>(mAudioTrack->getBufferSizeInFrames());
 }
 
-int32_t AudioStreamTrack::getBufferCapacity() const
+int32_t AudioStreamTrack::getBufferCapacityFromDevice() const
 {
     return static_cast<int32_t>(mAudioTrack->frameCount());
 }
@@ -484,8 +481,7 @@
     return static_cast<int32_t>(mAudioTrack->getUnderrunCount());
 }
 
-int32_t AudioStreamTrack::getFramesPerBurst() const
-{
+int32_t AudioStreamTrack::getFramesPerBurstFromDevice() const {
     return static_cast<int32_t>(mAudioTrack->getNotificationPeriodInFrames());
 }
 
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 6334f66..5a8fb39 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -69,8 +69,6 @@
 
     aaudio_result_t setBufferSize(int32_t requestedFrames) override;
     int32_t getBufferSize() const override;
-    int32_t getBufferCapacity() const override;
-    int32_t getFramesPerBurst()const  override;
     int32_t getXRunCount() const override;
 
     int64_t getFramesRead() override;
@@ -96,6 +94,11 @@
             const android::media::VolumeShaper::Operation& operation) override;
 #endif
 
+protected:
+
+    int32_t getFramesPerBurstFromDevice() const override;
+    int32_t getBufferCapacityFromDevice() const override;
+
 private:
 
     android::sp<android::AudioTrack> mAudioTrack;
@@ -105,10 +108,6 @@
 
     // TODO add 64-bit position reporting to AudioTrack and use it.
     aaudio_wrapping_frames_t         mPositionWhenPausing = 0;
-
-    // initial AudioTrack frame count and notification period
-    int32_t mInitialBufferCapacity = 0;
-    int32_t mInitialFramesPerBurst = 0;
 };
 
 } /* namespace aaudio */