diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index 4e8ad52..06e8487 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -1186,9 +1186,9 @@
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
-    Status status = mPlugin->setCipherAlgorithm(toHidlVec(sessionId),
+    Return<Status> status = mPlugin->setCipherAlgorithm(toHidlVec(sessionId),
             toHidlString(algorithm));
-    return toStatusT(status);
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::setMacAlgorithm(Vector<uint8_t> const &sessionId,
@@ -1198,9 +1198,9 @@
 
     DrmSessionManager::Instance()->useSession(sessionId);
 
-    Status status = mPlugin->setMacAlgorithm(toHidlVec(sessionId),
+    Return<Status> status = mPlugin->setMacAlgorithm(toHidlVec(sessionId),
             toHidlString(algorithm));
-    return toStatusT(status);
+    return status.isOk() ? toStatusT(status) : DEAD_OBJECT;
 }
 
 status_t DrmHal::encrypt(Vector<uint8_t> const &sessionId,
diff --git a/media/img_utils/include/img_utils/DngUtils.h b/media/img_utils/include/img_utils/DngUtils.h
index 1d8df9c..de8f120 100644
--- a/media/img_utils/include/img_utils/DngUtils.h
+++ b/media/img_utils/include/img_utils/DngUtils.h
@@ -39,11 +39,16 @@
  */
 class ANDROID_API OpcodeListBuilder : public LightRefBase<OpcodeListBuilder> {
     public:
+        // Note that the Adobe DNG 1.4 spec for Bayer phase (defined for the
+        // FixBadPixelsConstant and FixBadPixelsList opcodes) is incorrect. It's
+        // inconsistent with the DNG SDK (cf. dng_negative::SetBayerMosaic and
+        // dng_opcode_FixBadPixelsList::IsGreen), and Adobe confirms that the
+        // spec should be updated to match the SDK.
         enum CfaLayout {
-            CFA_RGGB = 0,
-            CFA_GRBG,
-            CFA_GBRG,
+            CFA_GRBG = 0,
+            CFA_RGGB,
             CFA_BGGR,
+            CFA_GBRG,
         };
 
         OpcodeListBuilder();
diff --git a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
index 2623697..f618d3d 100644
--- a/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
+++ b/media/libaaudio/examples/loopback/src/LoopbackAnalyzer.h
@@ -495,7 +495,7 @@
     }
 
     void printStatus() override {
-        printf("state = %d, echo gain = %f ", mState, mEchoGain);
+        printf("st = %d, echo gain = %f ", mState, mEchoGain);
     }
 
     static void sendImpulse(float *outputData, int outputChannelCount) {
@@ -670,7 +670,7 @@
         printf(LOOPBACK_RESULT_TAG "phase.offset       = %7.5f\n", mPhaseOffset);
         printf(LOOPBACK_RESULT_TAG "ref.phase          = %7.5f\n", mPhase);
         printf(LOOPBACK_RESULT_TAG "frames.accumulated = %6d\n", mFramesAccumulated);
-        printf(LOOPBACK_RESULT_TAG "sine.period        = %6d\n", mPeriod);
+        printf(LOOPBACK_RESULT_TAG "sine.period        = %6d\n", mSinePeriod);
         printf(LOOPBACK_RESULT_TAG "test.state         = %6d\n", mState);
         printf(LOOPBACK_RESULT_TAG "frame.count        = %6d\n", mFrameCounter);
         // Did we ever get a lock?
@@ -684,7 +684,7 @@
     }
 
     void printStatus() override {
-        printf("  state = %d, glitches = %3d,", mState, mGlitchCount);
+        printf("st = %d, #gl = %3d,", mState, mGlitchCount);
     }
 
     double calculateMagnitude(double *phasePtr = NULL) {
@@ -709,6 +709,8 @@
     void process(float *inputData, int inputChannelCount,
                  float *outputData, int outputChannelCount,
                  int numFrames) override {
+        mProcessCount++;
+
         float peak = measurePeakAmplitude(inputData, inputChannelCount, numFrames);
         if (peak > mPeakAmplitude) {
             mPeakAmplitude = peak;
@@ -720,6 +722,7 @@
             float sinOut = sinf(mPhase);
 
             switch (mState) {
+                case STATE_IDLE:
                 case STATE_IMMUNE:
                 case STATE_WAITING_FOR_SIGNAL:
                     break;
@@ -728,7 +731,7 @@
                     mCosAccumulator += sample * cosf(mPhase);
                     mFramesAccumulated++;
                     // Must be a multiple of the period or the calculation will not be accurate.
-                    if (mFramesAccumulated == mPeriod * 4) {
+                    if (mFramesAccumulated == mSinePeriod * PERIODS_NEEDED_FOR_LOCK) {
                         mPhaseOffset = 0.0;
                         mMagnitude = calculateMagnitude(&mPhaseOffset);
                         if (mMagnitude > mThreshold) {
@@ -754,7 +757,22 @@
                         //       mFrameCounter, mGlitchCount, predicted, sample);
                         mState = STATE_IMMUNE;
                         //printf("%5d: switch to STATE_IMMUNE\n", mFrameCounter);
-                        mDownCounter = mPeriod;  // Set duration of IMMUNE state.
+                        mDownCounter = mSinePeriod;  // Set duration of IMMUNE state.
+                    }
+
+                    // Track incoming signal and slowly adjust magnitude to account
+                    // for drift in the DRC or AGC.
+                    mSinAccumulator += sample * sinOut;
+                    mCosAccumulator += sample * cosf(mPhase);
+                    mFramesAccumulated++;
+                    // Must be a multiple of the period or the calculation will not be accurate.
+                    if (mFramesAccumulated == mSinePeriod) {
+                        const double coefficient = 0.1;
+                        double phaseOffset = 0.0;
+                        double magnitude = calculateMagnitude(&phaseOffset);
+                        // One pole averaging filter.
+                        mMagnitude = (mMagnitude * (1.0 - coefficient)) + (magnitude * coefficient);
+                        resetAccumulator();
                     }
                 } break;
             }
@@ -775,6 +793,9 @@
 
         // Do these once per buffer.
         switch (mState) {
+            case STATE_IDLE:
+                mState = STATE_IMMUNE; // so we can tell when
+                break;
             case STATE_IMMUNE:
                 mDownCounter -= numFrames;
                 if (mDownCounter <= 0) {
@@ -805,21 +826,29 @@
     void reset() override {
         mGlitchCount = 0;
         mState = STATE_IMMUNE;
-        mPhaseIncrement = 2.0 * M_PI / mPeriod;
-        printf("phaseInc = %f for period %d\n", mPhaseIncrement, mPeriod);
+        mDownCounter = IMMUNE_FRAME_COUNT;
+        mPhaseIncrement = 2.0 * M_PI / mSinePeriod;
+        printf("phaseInc = %f for period %d\n", mPhaseIncrement, mSinePeriod);
         resetAccumulator();
+        mProcessCount = 0;
     }
 
 private:
 
     enum sine_state_t {
+        STATE_IDLE,
         STATE_IMMUNE,
         STATE_WAITING_FOR_SIGNAL,
         STATE_WAITING_FOR_LOCK,
         STATE_LOCKED
     };
 
-    int     mPeriod = 79;
+    enum constants {
+        IMMUNE_FRAME_COUNT = 48 * 500,
+        PERIODS_NEEDED_FOR_LOCK = 8
+    };
+
+    int     mSinePeriod = 79;
     double  mPhaseIncrement = 0.0;
     double  mPhase = 0.0;
     double  mPhaseOffset = 0.0;
@@ -828,18 +857,19 @@
     double  mThreshold = 0.005;
     double  mTolerance = 0.01;
     int32_t mFramesAccumulated = 0;
+    int32_t mProcessCount = 0;
     double  mSinAccumulator = 0.0;
     double  mCosAccumulator = 0.0;
     int32_t mGlitchCount = 0;
     double  mPeakAmplitude = 0.0;
-    int     mDownCounter = 4000;
+    int     mDownCounter = IMMUNE_FRAME_COUNT;
     int32_t mFrameCounter = 0;
     float   mOutputAmplitude = 0.75;
 
     PseudoRandom  mWhiteNoise;
     float   mNoiseAmplitude = 0.00; // Used to experiment with warbling caused by DRC.
 
-    sine_state_t  mState = STATE_IMMUNE;
+    sine_state_t  mState = STATE_IDLE;
 };
 
 
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 0ebcdbd..26d1e4b 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -38,21 +38,25 @@
 // Tag for machine readable results as property = value pairs
 #define RESULT_TAG              "RESULT: "
 #define NUM_SECONDS             5
+#define PERIOD_MILLIS           1000
 #define NUM_INPUT_CHANNELS      1
 #define FILENAME_ALL            "/data/loopback_all.wav"
 #define FILENAME_ECHOS          "/data/loopback_echos.wav"
-#define APP_VERSION             "0.2.01"
+#define APP_VERSION             "0.2.03"
+
+constexpr int kNumCallbacksToDrain   = 20;
+constexpr int kNumCallbacksToDiscard = 20;
 
 struct LoopbackData {
     AAudioStream      *inputStream = nullptr;
     int32_t            inputFramesMaximum = 0;
     int16_t           *inputShortData = nullptr;
     float             *inputFloatData = nullptr;
-    int16_t            peakShort = 0;
     aaudio_format_t    actualInputFormat = AAUDIO_FORMAT_INVALID;
     int32_t            actualInputChannelCount = 0;
     int32_t            actualOutputChannelCount = 0;
-    int32_t            inputBuffersToDiscard = 10;
+    int32_t            numCallbacksToDrain = kNumCallbacksToDrain;
+    int32_t            numCallbacksToDiscard = kNumCallbacksToDiscard;
     int32_t            minNumFrames = INT32_MAX;
     int32_t            maxNumFrames = 0;
     int32_t            insufficientReadCount = 0;
@@ -131,23 +135,32 @@
         myData->minNumFrames = numFrames;
     }
 
-    if (myData->inputBuffersToDiscard > 0) {
+    // Silence the output.
+    int32_t numBytes = numFrames * myData->actualOutputChannelCount * sizeof(float);
+    memset(audioData, 0 /* value */, numBytes);
+
+    if (myData->numCallbacksToDrain > 0) {
         // Drain the input.
         do {
             framesRead = readFormattedData(myData, numFrames);
-            if (framesRead < 0) {
-                result = AAUDIO_CALLBACK_RESULT_STOP;
-            } else if (framesRead > 0) {
-                myData->inputBuffersToDiscard--;
-            }
+            // Ignore errors because input stream may not be started yet.
         } while (framesRead > 0);
+        myData->numCallbacksToDrain--;
 
-        // Silence the output.
-        int32_t numBytes = numFrames * myData->actualOutputChannelCount * sizeof(float);
-        memset(audioData, 0 /* value */, numBytes);
+    } else if (myData->numCallbacksToDiscard > 0) {
+        // Ignore. Allow the input to fill back up to equilibrium with the output.
+        framesRead = readFormattedData(myData, numFrames);
+        if (framesRead < 0) {
+            result = AAUDIO_CALLBACK_RESULT_STOP;
+        }
+        myData->numCallbacksToDiscard--;
 
     } else {
 
+        int32_t numInputBytes = numFrames * myData->actualInputChannelCount * sizeof(float);
+        memset(myData->inputFloatData, 0 /* value */, numInputBytes);
+
+        // Process data after equilibrium.
         int64_t inputFramesWritten = AAudioStream_getFramesWritten(myData->inputStream);
         int64_t inputFramesRead = AAudioStream_getFramesRead(myData->inputStream);
         int64_t framesAvailable = inputFramesWritten - inputFramesRead;
@@ -155,6 +168,7 @@
         if (framesRead < 0) {
             result = AAUDIO_CALLBACK_RESULT_STOP;
         } else {
+
             if (framesRead < numFrames) {
                 if(framesRead < (int32_t) framesAvailable) {
                     printf("insufficient but numFrames = %d, framesRead = %d, available = %d\n",
@@ -172,13 +186,13 @@
             // Save for later.
             myData->audioRecording.write(myData->inputFloatData,
                                          myData->actualInputChannelCount,
-                                         framesRead);
+                                         numFrames);
             // Analyze the data.
             myData->loopbackProcessor->process(myData->inputFloatData,
                                                myData->actualInputChannelCount,
                                                outputData,
                                                myData->actualOutputChannelCount,
-                                               framesRead);
+                                               numFrames);
             myData->isDone = myData->loopbackProcessor->isDone();
             if (myData->isDone) {
                 result = AAUDIO_CALLBACK_RESULT_STOP;
@@ -366,7 +380,9 @@
     }
 
     int32_t requestedDuration = argParser.getDurationSeconds();
-    int32_t recordingDuration = std::min(60, requestedDuration);
+    int32_t requestedDurationMillis = requestedDuration * MILLIS_PER_SECOND;
+    int32_t timeMillis = 0;
+    int32_t recordingDuration = std::min(60 * 5, requestedDuration);
 
     switch(testMode) {
         case TEST_SINE_MAGNITUDE:
@@ -449,7 +465,6 @@
 
     // Allocate a buffer for the audio data.
     loopbackData.inputFramesMaximum = 32 * AAudioStream_getFramesPerBurst(inputStream);
-    loopbackData.inputBuffersToDiscard = 200;
 
     if (loopbackData.actualInputFormat == AAUDIO_FORMAT_PCM_I16) {
         loopbackData.inputShortData = new int16_t[loopbackData.inputFramesMaximum
@@ -460,13 +475,7 @@
 
     loopbackData.loopbackProcessor->reset();
 
-    result = recorder.start();
-    if (result != AAUDIO_OK) {
-        printf("ERROR - AAudioStream_requestStart(input) returned %d = %s\n",
-               result, AAudio_convertResultToText(result));
-        goto finish;
-    }
-
+    // Start OUTPUT first so INPUT does not overflow.
     result = player.start();
     if (result != AAUDIO_OK) {
         printf("ERROR - AAudioStream_requestStart(output) returned %d = %s\n",
@@ -474,9 +483,15 @@
         goto finish;
     }
 
-    printf("------- sleep while the callback runs --------------\n");
-    fflush(stdout);
-    for (int i = requestedDuration; i > 0 ; i--) {
+    result = recorder.start();
+    if (result != AAUDIO_OK) {
+        printf("ERROR - AAudioStream_requestStart(input) returned %d = %s\n",
+               result, AAudio_convertResultToText(result));
+        goto finish;
+    }
+
+    printf("------- sleep and log while the callback runs --------------\n");
+    while (timeMillis <= requestedDurationMillis) {
         if (loopbackData.inputError != AAUDIO_OK) {
             printf("  ERROR on input stream\n");
             break;
@@ -487,10 +502,9 @@
                 printf("  test says it is done!\n");
                 break;
         } else {
-            sleep(1);
-            printf("%4d: ", i);
+            // Log a line of stream data.
+            printf("%7.3f: ", 0.001 * timeMillis); // display in seconds
             loopbackData.loopbackProcessor->printStatus();
-
             printf(" insf %3d,", (int) loopbackData.insufficientReadCount);
 
             int64_t inputFramesWritten = AAudioStream_getFramesWritten(inputStream);
@@ -498,7 +512,7 @@
             int64_t outputFramesWritten = AAudioStream_getFramesWritten(outputStream);
             int64_t outputFramesRead = AAudioStream_getFramesRead(outputStream);
             static const int textOffset = strlen("AAUDIO_STREAM_STATE_"); // strip this off
-            printf(" INPUT: wr %7lld - rd %7lld = %5lld, state %s, oruns %3d | ",
+            printf(" | INPUT: wr %7lld - rd %7lld = %5lld, st %8s, oruns %3d",
                    (long long) inputFramesWritten,
                    (long long) inputFramesRead,
                    (long long) (inputFramesWritten - inputFramesRead),
@@ -506,7 +520,7 @@
                            AAudioStream_getState(inputStream))[textOffset],
                    AAudioStream_getXRunCount(inputStream));
 
-            printf(" OUTPUT: wr %7lld - rd %7lld = %5lld, state %s, uruns %3d\n",
+            printf(" | OUTPUT: wr %7lld - rd %7lld = %5lld, st %8s, uruns %3d\n",
                    (long long) outputFramesWritten,
                    (long long) outputFramesRead,
                     (long long) (outputFramesWritten - outputFramesRead),
@@ -515,6 +529,9 @@
                    AAudioStream_getXRunCount(outputStream)
             );
         }
+        int32_t periodMillis = (timeMillis < 2000) ? PERIOD_MILLIS / 4 : PERIOD_MILLIS;
+        usleep(periodMillis * 1000);
+        timeMillis += periodMillis;
     }
 
     result = player.stop();
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ad81f04..2abea9e 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2144,6 +2144,10 @@
                 // value is unknown
                 drc.targetRefLevel = -1;
             }
+            if (!msg->findInt32("aac-drc-effect-type", &drc.effectType)) {
+                // value is unknown
+                drc.effectType = -2; // valid values are -1 and over
+            }
 
             err = setupAACCodec(
                     encoder, numChannels, sampleRate, bitrate, aacProfile,
@@ -2778,7 +2782,7 @@
             ? OMX_AUDIO_AACStreamFormatMP4ADTS
             : OMX_AUDIO_AACStreamFormatMP4FF;
 
-    OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE presentation;
+    OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE presentation;
     InitOMXParams(&presentation);
     presentation.nMaxOutputChannels = maxOutputChannelCount;
     presentation.nDrcCut = drc.drcCut;
@@ -2787,14 +2791,29 @@
     presentation.nTargetReferenceLevel = drc.targetRefLevel;
     presentation.nEncodedTargetLevel = drc.encodedTargetLevel;
     presentation.nPCMLimiterEnable = pcmLimiterEnable;
+    presentation.nDrcEffectType = drc.effectType;
 
     status_t res = mOMXNode->setParameter(
             OMX_IndexParamAudioAac, &profile, sizeof(profile));
     if (res == OK) {
         // optional parameters, will not cause configuration failure
-        mOMXNode->setParameter(
+        if (mOMXNode->setParameter(
+                (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacDrcPresentation,
+                &presentation, sizeof(presentation)) == ERROR_UNSUPPORTED) {
+            // prior to 9.0 we used a different config structure and index
+            OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE presentation8;
+            InitOMXParams(&presentation8);
+            presentation8.nMaxOutputChannels = presentation.nMaxOutputChannels;
+            presentation8.nDrcCut = presentation.nDrcCut;
+            presentation8.nDrcBoost = presentation.nDrcBoost;
+            presentation8.nHeavyCompression = presentation.nHeavyCompression;
+            presentation8.nTargetReferenceLevel = presentation.nTargetReferenceLevel;
+            presentation8.nEncodedTargetLevel = presentation.nEncodedTargetLevel;
+            presentation8.nPCMLimiterEnable = presentation.nPCMLimiterEnable;
+            (void)mOMXNode->setParameter(
                 (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAacPresentation,
-                &presentation, sizeof(presentation));
+                &presentation8, sizeof(presentation8));
+        }
     } else {
         ALOGW("did not set AudioAndroidAacPresentation due to error %d when setting AudioAac", res);
     }
diff --git a/media/libstagefright/StagefrightMediaScanner.cpp b/media/libstagefright/StagefrightMediaScanner.cpp
index 4ff2bfe..e010b3e 100644
--- a/media/libstagefright/StagefrightMediaScanner.cpp
+++ b/media/libstagefright/StagefrightMediaScanner.cpp
@@ -40,7 +40,8 @@
         ".mpeg", ".ogg", ".mid", ".smf", ".imy", ".wma", ".aac",
         ".wav", ".amr", ".midi", ".xmf", ".rtttl", ".rtx", ".ota",
         ".mkv", ".mka", ".webm", ".ts", ".fl", ".flac", ".mxmf",
-        ".avi", ".mpeg", ".mpg", ".awb", ".mpga", ".mov"
+        ".avi", ".mpeg", ".mpg", ".awb", ".mpga", ".mov",
+        ".m4v", ".oga"
     };
     static const size_t kNumValidExtensions =
         sizeof(kValidExtensions) / sizeof(kValidExtensions[0]);
@@ -62,6 +63,11 @@
     client.setLocale(locale());
     client.beginFile();
     MediaScanResult result = processFileInternal(path, mimeType, client);
+    ALOGV("result: %d", result);
+    if (mimeType == NULL && result != MEDIA_SCAN_RESULT_OK) {
+        ALOGW("media scan failed for %s", path);
+        client.setMimeType("application/octet-stream");
+    }
     client.endFile();
     return result;
 }
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index c61f4b5..cf5e91e 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -154,6 +154,7 @@
         { 23, OMX_AUDIO_AACObjectLD       },
         { 29, OMX_AUDIO_AACObjectHE_PS    },
         { 39, OMX_AUDIO_AACObjectELD      },
+        { 42, OMX_AUDIO_AACObjectXHE      },
     };
 
     OMX_AUDIO_AACPROFILETYPE profile;
@@ -1610,6 +1611,7 @@
     { OMX_AUDIO_AACObjectLD,          AUDIO_FORMAT_AAC_LD},
     { OMX_AUDIO_AACObjectHE_PS,       AUDIO_FORMAT_AAC_HE_V2},
     { OMX_AUDIO_AACObjectELD,         AUDIO_FORMAT_AAC_ELD},
+    { OMX_AUDIO_AACObjectXHE,         AUDIO_FORMAT_AAC_XHE},
     { OMX_AUDIO_AACObjectNull,        AUDIO_FORMAT_AAC},
 };
 
diff --git a/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
index 129ad65..95d3724 100644
--- a/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
+++ b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
@@ -24,7 +24,7 @@
 //#define DRC_PRES_MODE_WRAP_DEBUG
 
 #define GPM_ENCODER_TARGET_LEVEL 64
-#define MAX_TARGET_LEVEL 64
+#define MAX_TARGET_LEVEL 40
 
 CDrcPresModeWrapper::CDrcPresModeWrapper()
 {
@@ -164,7 +164,7 @@
     if (mDataUpdate) {
         // sanity check
         if (mDesTarget < MAX_TARGET_LEVEL){
-            mDesTarget = MAX_TARGET_LEVEL;  // limit target level to -16 dB or below
+            mDesTarget = MAX_TARGET_LEVEL;  // limit target level to -10 dB or below
             newTarget = MAX_TARGET_LEVEL;
         }
 
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index e0c0c32..ecd2512 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -36,6 +36,7 @@
 #define DRC_DEFAULT_MOBILE_DRC_CUT   127 /* maximum compression of dynamic range for mobile conf */
 #define DRC_DEFAULT_MOBILE_DRC_BOOST 127 /* maximum compression of dynamic range for mobile conf */
 #define DRC_DEFAULT_MOBILE_DRC_HEAVY 1   /* switch for heavy compression for mobile conf */
+#define DRC_DEFAULT_MOBILE_DRC_EFFECT 3  /* MPEG-D DRC effect type; 3 => Limited playback range */
 #define DRC_DEFAULT_MOBILE_ENC_LEVEL (-1) /* encoder target level; -1 => the value is unknown, otherwise dB step value (e.g. 64 for -16 dB) */
 #define MAX_CHANNEL_COUNT            8  /* maximum number of audio channels that can be decoded */
 // names of properties that can be used to override the default DRC settings
@@ -44,6 +45,7 @@
 #define PROP_DRC_OVERRIDE_BOOST      "aac_drc_boost"
 #define PROP_DRC_OVERRIDE_HEAVY      "aac_drc_heavy"
 #define PROP_DRC_OVERRIDE_ENC_LEVEL "aac_drc_enc_target_level"
+#define PROP_DRC_OVERRIDE_EFFECT     "aac_drc_effect_type"
 
 namespace android {
 
@@ -207,6 +209,17 @@
     } else {
         mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, DRC_DEFAULT_MOBILE_ENC_LEVEL);
     }
+    // AAC_UNIDRC_SET_EFFECT
+    int32_t effectType = DRC_DEFAULT_MOBILE_DRC_EFFECT;
+    // FIXME can't read default property for DRC effect type
+    //int32_t effectType =
+    //        property_get_int32(PROP_DRC_OVERRIDE_EFFECT, DRC_DEFAULT_MOBILE_DRC_EFFECT);
+    if (effectType < -1 || effectType > 8) {
+        effectType = DRC_DEFAULT_MOBILE_DRC_EFFECT;
+    }
+    ALOGV("AAC decoder using MPEG-D DRC effect type %d (default=%d)",
+            effectType, DRC_DEFAULT_MOBILE_DRC_EFFECT);
+    aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_SET_EFFECT, effectType);
 
     // By default, the decoder creates a 5.1 channel downmix signal.
     // For seven and eight channel input streams, enable 6.1 and 7.1 channel output
@@ -414,10 +427,10 @@
             return OMX_ErrorNone;
         }
 
-        case OMX_IndexParamAudioAndroidAacPresentation:
+        case OMX_IndexParamAudioAndroidAacDrcPresentation:
         {
-            const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *aacPresParams =
-                    (const OMX_AUDIO_PARAM_ANDROID_AACPRESENTATIONTYPE *)params;
+            const OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *aacPresParams =
+                    (const OMX_AUDIO_PARAM_ANDROID_AACDRCPRESENTATIONTYPE *)params;
 
             if (!isValidOMXParam(aacPresParams)) {
                 return OMX_ErrorBadParameter;
@@ -443,6 +456,10 @@
                 ALOGV("set nMaxOutputChannels=%d", max);
                 aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, max);
             }
+            if (aacPresParams->nDrcEffectType >= -1) {
+                ALOGV("set nDrcEffectType=%d", aacPresParams->nDrcEffectType);
+                aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_SET_EFFECT, aacPresParams->nDrcEffectType);
+            }
             bool updateDrcWrapper = false;
             if (aacPresParams->nDrcBoost >= 0) {
                 ALOGV("set nDrcBoost=%d", aacPresParams->nDrcBoost);
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 64caeed..97d15a7 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -446,6 +446,7 @@
         int32_t heavyCompression;
         int32_t targetRefLevel;
         int32_t encodedTargetLevel;
+        int32_t effectType;
     } drcParams_t;
 
     status_t setupAACCodec(
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 20de97c..c47aa01 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6603,8 +6603,17 @@
         if (mPipeSource != 0) {
             size_t framesToRead = mBufferSize / mFrameSize;
             framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2);
-            framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
-                    framesToRead);
+
+            // The audio fifo read() returns OVERRUN on overflow, and advances the read pointer
+            // to the full buffer point (clearing the overflow condition).  Upon OVERRUN error,
+            // we immediately retry the read() to get data and prevent another overflow.
+            for (int retries = 0; retries <= 2; ++retries) {
+                ALOGW_IF(retries > 0, "overrun on read from pipe, retry #%d", retries);
+                framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
+                        framesToRead);
+                if (framesRead != OVERRUN) break;
+            }
+
             // since pipe is non-blocking, simulate blocking input by waiting for 1/2 of
             // buffer size or at least for 20ms.
             size_t sleepFrames = max(
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 3736129..264e709 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -915,7 +915,7 @@
                 // reuse direct output if currently open by the same client
                 // and configured with same parameters
                 if ((config->sample_rate == desc->mSamplingRate) &&
-                    audio_formats_match(config->format, desc->mFormat) &&
+                    (config->format == desc->mFormat) &&
                     (config->channel_mask == desc->mChannelMask) &&
                     (session == desc->mDirectClientSession)) {
                     desc->mDirectOpenCount++;
@@ -942,8 +942,7 @@
         // only accept an output with the requested parameters
         if (status != NO_ERROR ||
             (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
-            (config->format != AUDIO_FORMAT_DEFAULT &&
-                    !audio_formats_match(config->format, outputDesc->mFormat)) ||
+            (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
             (config->channel_mask != 0 && config->channel_mask != outputDesc->mChannelMask)) {
             ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
                     "format %d %d, channel mask %04x %04x", output, config->sample_rate,
@@ -1032,7 +1031,7 @@
             // if a valid format is specified, skip output if not compatible
             if (format != AUDIO_FORMAT_INVALID) {
                 if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
-                    if (!audio_formats_match(format, outputDesc->mFormat)) {
+                    if (format != outputDesc->mFormat) {
                         continue;
                     }
                 } else if (!audio_is_linear_pcm(format)) {