Merge "AAC decoder: compensate limiter delay"

5 files changed, 973 insertions, 218 deletions

diff --git a/media/libstagefright/codecs/aacdec/Android.mk b/media/libstagefright/codecs/aacdec/Android.mk
index 49ff238..afb00aa 100644
--- a/media/libstagefright/codecs/aacdec/Android.mk
+++ b/media/libstagefright/codecs/aacdec/Android.mk
@@ -3,7 +3,8 @@ LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 
 LOCAL_SRC_FILES := \
-      SoftAAC2.cpp
+      SoftAAC2.cpp \
+      DrcPresModeWrap.cpp
 
 LOCAL_C_INCLUDES := \
       frameworks/av/media/libstagefright/include \
diff --git a/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
new file mode 100644
index 0000000..129ad65
--- /dev/null
+++ b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.cpp
@@ -0,0 +1,372 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include "DrcPresModeWrap.h"
+
+#include <assert.h>
+
+#define LOG_TAG "SoftAAC2_DrcWrapper"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+//#define DRC_PRES_MODE_WRAP_DEBUG
+
+#define GPM_ENCODER_TARGET_LEVEL 64
+#define MAX_TARGET_LEVEL 64
+
+CDrcPresModeWrapper::CDrcPresModeWrapper()
+{
+    mDataUpdate = true;
+
+    /* Data from streamInfo. */
+    /* Initialized to the same values as in the aac decoder */
+    mStreamPRL = -1;
+    mStreamDRCPresMode = -1;
+    mStreamNrAACChan = 0;
+    mStreamNrOutChan = 0;
+
+    /* Desired values (set by user). */
+    /* Initialized to the same values as in the aac decoder */
+    mDesTarget = -1;
+    mDesAttFactor = 0;
+    mDesBoostFactor = 0;
+    mDesHeavy = 0;
+
+    mEncoderTarget = -1;
+
+    /* Values from last time. */
+    /* Initialized to the same values as the desired values */
+    mLastTarget = -1;
+    mLastAttFactor = 0;
+    mLastBoostFactor = 0;
+    mLastHeavy = 0;
+}
+
+CDrcPresModeWrapper::~CDrcPresModeWrapper()
+{
+}
+
+void
+CDrcPresModeWrapper::setDecoderHandle(const HANDLE_AACDECODER handle)
+{
+    mHandleDecoder = handle;
+}
+
+void
+CDrcPresModeWrapper::submitStreamData(CStreamInfo* pStreamInfo)
+{
+    assert(pStreamInfo);
+
+    if (mStreamPRL != pStreamInfo->drcProgRefLev) {
+        mStreamPRL = pStreamInfo->drcProgRefLev;
+        mDataUpdate = true;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+        ALOGV("DRC presentation mode wrapper: drcProgRefLev is %d\n", mStreamPRL);
+#endif
+    }
+
+    if (mStreamDRCPresMode != pStreamInfo->drcPresMode) {
+        mStreamDRCPresMode = pStreamInfo->drcPresMode;
+        mDataUpdate = true;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+        ALOGV("DRC presentation mode wrapper: drcPresMode is %d\n", mStreamDRCPresMode);
+#endif
+    }
+
+    if (mStreamNrAACChan != pStreamInfo->aacNumChannels) {
+        mStreamNrAACChan = pStreamInfo->aacNumChannels;
+        mDataUpdate = true;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+        ALOGV("DRC presentation mode wrapper: aacNumChannels is %d\n", mStreamNrAACChan);
+#endif
+    }
+
+    if (mStreamNrOutChan != pStreamInfo->numChannels) {
+        mStreamNrOutChan = pStreamInfo->numChannels;
+        mDataUpdate = true;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+        ALOGV("DRC presentation mode wrapper: numChannels is %d\n", mStreamNrOutChan);
+#endif
+    }
+
+
+
+    if (mStreamNrOutChan<mStreamNrAACChan) {
+        mIsDownmix = true;
+    } else {
+        mIsDownmix = false;
+    }
+
+    if (mIsDownmix && (mStreamNrOutChan == 1)) {
+        mIsMonoDownmix = true;
+    } else {
+        mIsMonoDownmix = false;
+    }
+
+    if (mIsDownmix && mStreamNrOutChan == 2){
+        mIsStereoDownmix = true;
+    } else {
+        mIsStereoDownmix = false;
+    }
+
+}
+
+void
+CDrcPresModeWrapper::setParam(const DRC_PRES_MODE_WRAP_PARAM param, const int value)
+{
+    switch (param) {
+    case DRC_PRES_MODE_WRAP_DESIRED_TARGET:
+        mDesTarget = value;
+        break;
+    case DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR:
+        mDesAttFactor = value;
+        break;
+    case DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR:
+        mDesBoostFactor = value;
+        break;
+    case DRC_PRES_MODE_WRAP_DESIRED_HEAVY:
+        mDesHeavy = value;
+        break;
+    case DRC_PRES_MODE_WRAP_ENCODER_TARGET:
+        mEncoderTarget = value;
+        break;
+    default:
+        break;
+    }
+    mDataUpdate = true;
+}
+
+void
+CDrcPresModeWrapper::update()
+{
+    // Get Data from Decoder
+    int progRefLevel = mStreamPRL;
+    int drcPresMode = mStreamDRCPresMode;
+
+    // by default, do as desired
+    int newTarget         = mDesTarget;
+    int newAttFactor      = mDesAttFactor;
+    int newBoostFactor    = mDesBoostFactor;
+    int newHeavy          = mDesHeavy;
+
+    if (mDataUpdate) {
+        // sanity check
+        if (mDesTarget < MAX_TARGET_LEVEL){
+            mDesTarget = MAX_TARGET_LEVEL;  // limit target level to -16 dB or below
+            newTarget = MAX_TARGET_LEVEL;
+        }
+
+        if (mEncoderTarget != -1) {
+            if (mDesTarget<124) { // if target level > -31 dB
+                if ((mIsStereoDownmix == false) && (mIsMonoDownmix == false)) {
+                    // no stereo or mono downmixing, calculated scaling of light DRC
+                    /* use as little compression as possible */
+                    newAttFactor = 0;
+                    newBoostFactor = 0;
+                    if (mDesTarget<progRefLevel) { // if target level > PRL
+                        if (mEncoderTarget < mDesTarget) { // if mEncoderTarget > target level
+                            // mEncoderTarget > target level > PRL
+                            int calcFactor;
+                            float calcFactor_norm;
+                            // 0.0f < calcFactor_norm < 1.0f
+                            calcFactor_norm = (float)(mDesTarget - progRefLevel) /
+                                    (float)(mEncoderTarget - progRefLevel);
+                            calcFactor = (int)(calcFactor_norm*127.0f); // 0 <= calcFactor < 127
+                            // calcFactor is the lower limit
+                            newAttFactor = (calcFactor>newAttFactor) ? calcFactor : newAttFactor;
+                            // new AttFactor will be always = calcFactor, as it is set to 0 before.
+                            newBoostFactor = newAttFactor;
+                        } else {
+                            /* target level > mEncoderTarget > PRL */
+                            // newTDLimiterEnable = 1;
+                            // the time domain limiter must always be active in this case.
+                            //     It is assumed that the framework activates it by default
+                            newAttFactor = 127;
+                            newBoostFactor = 127;
+                        }
+                    } else { // target level <= PRL
+                        // no restrictions required
+                        // newAttFactor = newAttFactor;
+                    }
+                } else { // downmixing
+                    // if target level > -23 dB or mono downmix
+                    if ( (mDesTarget<92) || mIsMonoDownmix ) {
+                        newHeavy = 1;
+                    } else {
+                        // we perform a downmix, so, we need at least full light DRC
+                        newAttFactor = 127;
+                    }
+                }
+            } else { // target level <= -31 dB
+                // playback -31 dB: light DRC only needed if we perform downmixing
+                if (mIsDownmix) {   // we do downmixing
+                    newAttFactor = 127;
+                }
+            }
+        }
+        else { // handle other used encoder target levels
+
+            // Sanity check: DRC presentation mode is only specified for max. 5.1 channels
+            if (mStreamNrAACChan > 6) {
+                drcPresMode = 0;
+            }
+
+            switch (drcPresMode) {
+            case 0:
+            default: // presentation mode not indicated
+            {
+
+                if (mDesTarget<124) { // if target level > -31 dB
+                    // no stereo or mono downmixing
+                    if ((mIsStereoDownmix == false) && (mIsMonoDownmix == false)) {
+                        if (mDesTarget<progRefLevel) { // if target level > PRL
+                            // newTDLimiterEnable = 1;
+                            // the time domain limiter must always be active in this case.
+                            //    It is assumed that the framework activates it by default
+                            newAttFactor = 127; // at least, use light compression
+                        } else { // target level <= PRL
+                            // no restrictions required
+                            // newAttFactor = newAttFactor;
+                        }
+                    } else { // downmixing
+                        // newTDLimiterEnable = 1;
+                        // the time domain limiter must always be active in this case.
+                        //    It is assumed that the framework activates it by default
+
+                        // if target level > -23 dB or mono downmix
+                        if ( (mDesTarget < 92) || mIsMonoDownmix ) {
+                            newHeavy = 1;
+                        } else{
+                            // we perform a downmix, so, we need at least full light DRC
+                            newAttFactor = 127;
+                        }
+                    }
+                } else { // target level <= -31 dB
+                    if (mIsDownmix) {   // we do downmixing.
+                        // newTDLimiterEnable = 1;
+                        // the time domain limiter must always be active in this case.
+                        //    It is assumed that the framework activates it by default
+                        newAttFactor = 127;
+                    }
+                }
+            }
+            break;
+
+            // Presentation mode 1 and 2 according to ETSI TS 101 154:
+            // Digital Video Broadcasting (DVB); Specification for the use of Video and Audio Coding
+            // in Broadcasting Applications based on the MPEG-2 Transport Stream,
+            // section C.5.4., "Decoding", and Table C.33
+            // ISO DRC            -> newHeavy = 0  (Use light compression, MPEG-style)
+            // Compression_value  -> newHeavy = 1  (Use heavy compression, DVB-style)
+            // scaling restricted -> newAttFactor = 127
+
+            case 1: // presentation mode 1, Light:-31/Heavy:-23
+            {
+                if (mDesTarget < 124) { // if target level > -31 dB
+                    // playback up to -23 dB
+                    newHeavy = 1;
+                } else { // target level <= -31 dB
+                    // playback -31 dB
+                    if (mIsDownmix) {   // we do downmixing.
+                        newAttFactor = 127;
+                    }
+                }
+            }
+            break;
+
+            case 2: // presentation mode 2, Light:-23/Heavy:-23
+            {
+                if (mDesTarget < 124) { // if target level > -31 dB
+                    // playback up to -23 dB
+                    if (mIsMonoDownmix) { // if mono downmix
+                        newHeavy = 1;
+                    } else {
+                        newHeavy = 0;
+                        newAttFactor = 127;
+                    }
+                } else { // target level <= -31 dB
+                    // playback -31 dB
+                    newHeavy = 0;
+                    if (mIsDownmix) {   // we do downmixing.
+                        newAttFactor = 127;
+                    }
+                }
+            }
+            break;
+
+            } // switch()
+        } // if (mEncoderTarget  == GPM_ENCODER_TARGET_LEVEL)
+
+        // sanity again
+        if (newHeavy == 1) {
+            newBoostFactor=127; // not really needed as the same would be done by the decoder anyway
+            newAttFactor = 127;
+        }
+
+        // update the decoder
+        if (newTarget != mLastTarget) {
+            aacDecoder_SetParam(mHandleDecoder, AAC_DRC_REFERENCE_LEVEL, newTarget);
+            mLastTarget = newTarget;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+            if (newTarget != mDesTarget)
+                ALOGV("DRC presentation mode wrapper: forced target level to %d (from %d)\n", newTarget, mDesTarget);
+            else
+                ALOGV("DRC presentation mode wrapper: set target level to %d\n", newTarget);
+#endif
+        }
+
+        if (newAttFactor != mLastAttFactor) {
+            aacDecoder_SetParam(mHandleDecoder, AAC_DRC_ATTENUATION_FACTOR, newAttFactor);
+            mLastAttFactor = newAttFactor;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+            if (newAttFactor != mDesAttFactor)
+                ALOGV("DRC presentation mode wrapper: forced attenuation factor to %d (from %d)\n", newAttFactor, mDesAttFactor);
+            else
+                ALOGV("DRC presentation mode wrapper: set attenuation factor to %d\n", newAttFactor);
+#endif
+        }
+
+        if (newBoostFactor != mLastBoostFactor) {
+            aacDecoder_SetParam(mHandleDecoder, AAC_DRC_BOOST_FACTOR, newBoostFactor);
+            mLastBoostFactor = newBoostFactor;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+            if (newBoostFactor != mDesBoostFactor)
+                ALOGV("DRC presentation mode wrapper: forced boost factor to %d (from %d)\n",
+                        newBoostFactor, mDesBoostFactor);
+            else
+                ALOGV("DRC presentation mode wrapper: set boost factor to %d\n", newBoostFactor);
+#endif
+        }
+
+        if (newHeavy != mLastHeavy) {
+            aacDecoder_SetParam(mHandleDecoder, AAC_DRC_HEAVY_COMPRESSION, newHeavy);
+            mLastHeavy = newHeavy;
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+            if (newHeavy != mDesHeavy)
+                ALOGV("DRC presentation mode wrapper: forced heavy compression to %d (from %d)\n",
+                        newHeavy, mDesHeavy);
+            else
+                ALOGV("DRC presentation mode wrapper: set heavy compression to %d\n", newHeavy);
+#endif
+        }
+
+#ifdef DRC_PRES_MODE_WRAP_DEBUG
+        ALOGV("DRC config: tgt_lev: %3d, cut: %3d, boost: %3d, heavy: %d\n", newTarget,
+                newAttFactor, newBoostFactor, newHeavy);
+#endif
+        mDataUpdate = false;
+
+    } // if (mDataUpdate)
+}
diff --git a/media/libstagefright/codecs/aacdec/DrcPresModeWrap.h b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.h
new file mode 100644
index 0000000..f0b6cf2
--- /dev/null
+++ b/media/libstagefright/codecs/aacdec/DrcPresModeWrap.h
@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2014 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#pragma once
+#include "aacdecoder_lib.h"
+
+typedef enum
+{
+    DRC_PRES_MODE_WRAP_DESIRED_TARGET         = 0x0000,
+    DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR     = 0x0001,
+    DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR   = 0x0002,
+    DRC_PRES_MODE_WRAP_DESIRED_HEAVY          = 0x0003,
+    DRC_PRES_MODE_WRAP_ENCODER_TARGET         = 0x0004
+} DRC_PRES_MODE_WRAP_PARAM;
+
+
+class CDrcPresModeWrapper {
+public:
+    CDrcPresModeWrapper();
+    ~CDrcPresModeWrapper();
+    void setDecoderHandle(const HANDLE_AACDECODER handle);
+    void setParam(const DRC_PRES_MODE_WRAP_PARAM param, const int value);
+    void submitStreamData(CStreamInfo*);
+    void update();
+
+protected:
+    HANDLE_AACDECODER mHandleDecoder;
+    int mDesTarget;
+    int mDesAttFactor;
+    int mDesBoostFactor;
+    int mDesHeavy;
+
+    int mEncoderTarget;
+
+    int mLastTarget;
+    int mLastAttFactor;
+    int mLastBoostFactor;
+    int mLastHeavy;
+
+    SCHAR mStreamPRL;
+    SCHAR mStreamDRCPresMode;
+    INT mStreamNrAACChan;
+    INT mStreamNrOutChan;
+
+    bool mIsDownmix;
+    bool mIsMonoDownmix;
+    bool mIsStereoDownmix;
+
+    bool mDataUpdate;
+};
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 532e36f..a0e3265 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -25,16 +25,22 @@
 #include <media/stagefright/foundation/hexdump.h>
 #include <media/stagefright/MediaErrors.h>
 
+#include <math.h>
+
 #define FILEREAD_MAX_LAYERS 2
 
 #define DRC_DEFAULT_MOBILE_REF_LEVEL 64  /* 64*-0.25dB = -16 dB below full scale for mobile conf */
 #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_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
 #define PROP_DRC_OVERRIDE_REF_LEVEL  "aac_drc_reference_level"
 #define PROP_DRC_OVERRIDE_CUT        "aac_drc_cut"
 #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"
 
 namespace android {
 
@@ -57,18 +63,19 @@ SoftAAC2::SoftAAC2(
       mStreamInfo(NULL),
       mIsADTS(false),
       mInputBufferCount(0),
+      mOutputBufferCount(0),
       mSignalledError(false),
-      mSawInputEos(false),
-      mSignalledOutputEos(false),
-      mAnchorTimeUs(0),
-      mNumSamplesOutput(0),
       mOutputPortSettingsChange(NONE) {
+    for (unsigned int i = 0; i < kNumDelayBlocksMax; i++) {
+        mAnchorTimeUs[i] = 0;
+    }
     initPorts();
     CHECK_EQ(initDecoder(), (status_t)OK);
 }
 
 SoftAAC2::~SoftAAC2() {
     aacDecoder_Close(mAACDecoder);
+    delete mOutputDelayRingBuffer;
 }
 
 void SoftAAC2::initPorts() {
@@ -121,36 +128,72 @@ status_t SoftAAC2::initDecoder() {
             status = OK;
         }
     }
-    mDecoderHasData = false;
 
-    // for streams that contain metadata, use the mobile profile DRC settings unless overridden
-    // by platform properties:
+    mEndOfInput = false;
+    mEndOfOutput = false;
+    mOutputDelayCompensated = 0;
+    mOutputDelayRingBufferSize = 2048 * MAX_CHANNEL_COUNT * kNumDelayBlocksMax;
+    mOutputDelayRingBuffer = new short[mOutputDelayRingBufferSize];
+    mOutputDelayRingBufferWritePos = 0;
+    mOutputDelayRingBufferReadPos = 0;
+
+    if (mAACDecoder == NULL) {
+        ALOGE("AAC decoder is null. TODO: Can not call aacDecoder_SetParam in the following code");
+    }
+
+    //aacDecoder_SetParam(mAACDecoder, AAC_PCM_LIMITER_ENABLE, 0);
+
+    //init DRC wrapper
+    mDrcWrap.setDecoderHandle(mAACDecoder);
+    mDrcWrap.submitStreamData(mStreamInfo);
+
+    // for streams that contain metadata, use the mobile profile DRC settings unless overridden by platform properties
+    // TODO: change the DRC settings depending on audio output device type (HDMI, loadspeaker, headphone)
     char value[PROPERTY_VALUE_MAX];
-    //  * AAC_DRC_REFERENCE_LEVEL
+    //  DRC_PRES_MODE_WRAP_DESIRED_TARGET
     if (property_get(PROP_DRC_OVERRIDE_REF_LEVEL, value, NULL)) {
         unsigned refLevel = atoi(value);
-        ALOGV("AAC decoder using AAC_DRC_REFERENCE_LEVEL of %d instead of %d",
-                refLevel, DRC_DEFAULT_MOBILE_REF_LEVEL);
-        aacDecoder_SetParam(mAACDecoder, AAC_DRC_REFERENCE_LEVEL, refLevel);
+        ALOGV("AAC decoder using desired DRC target reference level of %d instead of %d", refLevel,
+                DRC_DEFAULT_MOBILE_REF_LEVEL);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, refLevel);
     } else {
-        aacDecoder_SetParam(mAACDecoder, AAC_DRC_REFERENCE_LEVEL, DRC_DEFAULT_MOBILE_REF_LEVEL);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_TARGET, DRC_DEFAULT_MOBILE_REF_LEVEL);
     }
-    //  * AAC_DRC_ATTENUATION_FACTOR
+    //  DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR
     if (property_get(PROP_DRC_OVERRIDE_CUT, value, NULL)) {
         unsigned cut = atoi(value);
-        ALOGV("AAC decoder using AAC_DRC_ATTENUATION_FACTOR of %d instead of %d",
-                        cut, DRC_DEFAULT_MOBILE_DRC_CUT);
-        aacDecoder_SetParam(mAACDecoder, AAC_DRC_ATTENUATION_FACTOR, cut);
+        ALOGV("AAC decoder using desired DRC attenuation factor of %d instead of %d", cut,
+                DRC_DEFAULT_MOBILE_DRC_CUT);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, cut);
     } else {
-        aacDecoder_SetParam(mAACDecoder, AAC_DRC_ATTENUATION_FACTOR, DRC_DEFAULT_MOBILE_DRC_CUT);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_ATT_FACTOR, DRC_DEFAULT_MOBILE_DRC_CUT);
     }
-    //  * AAC_DRC_BOOST_FACTOR (note: no default, using cut)
+    //  DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR
     if (property_get(PROP_DRC_OVERRIDE_BOOST, value, NULL)) {
         unsigned boost = atoi(value);
-        ALOGV("AAC decoder using AAC_DRC_BOOST_FACTOR of %d", boost);
-        aacDecoder_SetParam(mAACDecoder, AAC_DRC_BOOST_FACTOR, boost);
+        ALOGV("AAC decoder using desired DRC boost factor of %d instead of %d", boost,
+                DRC_DEFAULT_MOBILE_DRC_BOOST);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, boost);
+    } else {
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_BOOST_FACTOR, DRC_DEFAULT_MOBILE_DRC_BOOST);
+    }
+    //  DRC_PRES_MODE_WRAP_DESIRED_HEAVY
+    if (property_get(PROP_DRC_OVERRIDE_HEAVY, value, NULL)) {
+        unsigned heavy = atoi(value);
+        ALOGV("AAC decoder using desried DRC heavy compression switch of %d instead of %d", heavy,
+                DRC_DEFAULT_MOBILE_DRC_HEAVY);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, heavy);
     } else {
-        aacDecoder_SetParam(mAACDecoder, AAC_DRC_BOOST_FACTOR, DRC_DEFAULT_MOBILE_DRC_BOOST);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_DESIRED_HEAVY, DRC_DEFAULT_MOBILE_DRC_HEAVY);
+    }
+    // DRC_PRES_MODE_WRAP_ENCODER_TARGET
+    if (property_get(PROP_DRC_OVERRIDE_ENC_LEVEL, value, NULL)) {
+        unsigned encoderRefLevel = atoi(value);
+        ALOGV("AAC decoder using encoder-side DRC reference level of %d instead of %d",
+                encoderRefLevel, DRC_DEFAULT_MOBILE_ENC_LEVEL);
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, encoderRefLevel);
+    } else {
+        mDrcWrap.setParam(DRC_PRES_MODE_WRAP_ENCODER_TARGET, DRC_DEFAULT_MOBILE_ENC_LEVEL);
     }
 
     return status;
@@ -290,19 +333,101 @@ bool SoftAAC2::isConfigured() const {
     return mInputBufferCount > 0;
 }
 
-void SoftAAC2::maybeConfigureDownmix() const {
-    if (mStreamInfo->numChannels > 2) {
-        char value[PROPERTY_VALUE_MAX];
-        if (!(property_get("media.aac_51_output_enabled", value, NULL) &&
-                (!strcmp(value, "1") || !strcasecmp(value, "true")))) {
-            ALOGI("Downmixing multichannel AAC to stereo");
-            aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
-            mStreamInfo->numChannels = 2;
-            // By default, the decoder creates a 5.1 channel downmix signal
-            // for seven and eight channel input streams. To enable 6.1 and 7.1 channel output
-            // use aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1)
+void SoftAAC2::configureDownmix() const {
+    char value[PROPERTY_VALUE_MAX];
+    if (!(property_get("media.aac_51_output_enabled", value, NULL)
+            && (!strcmp(value, "1") || !strcasecmp(value, "true")))) {
+        ALOGI("limiting to stereo output");
+        aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
+        // By default, the decoder creates a 5.1 channel downmix signal
+        // for seven and eight channel input streams. To enable 6.1 and 7.1 channel output
+        // use aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1)
+    }
+}
+
+bool SoftAAC2::outputDelayRingBufferPutSamples(INT_PCM *samples, int32_t numSamples) {
+    if (mOutputDelayRingBufferWritePos + numSamples <= mOutputDelayRingBufferSize
+            && (mOutputDelayRingBufferReadPos <= mOutputDelayRingBufferWritePos
+                    || mOutputDelayRingBufferReadPos > mOutputDelayRingBufferWritePos + numSamples)) {
+        // faster memcopy loop without checks, if the preconditions allow this
+        for (int32_t i = 0; i < numSamples; i++) {
+            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos++] = samples[i];
+        }
+
+        if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
+            mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
+        }
+        if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
+            ALOGE("RING BUFFER OVERFLOW");
+            return false;
+        }
+    } else {
+        ALOGV("slow SoftAAC2::outputDelayRingBufferPutSamples()");
+
+        for (int32_t i = 0; i < numSamples; i++) {
+            mOutputDelayRingBuffer[mOutputDelayRingBufferWritePos] = samples[i];
+            mOutputDelayRingBufferWritePos++;
+            if (mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferSize) {
+                mOutputDelayRingBufferWritePos -= mOutputDelayRingBufferSize;
+            }
+            if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
+                ALOGE("RING BUFFER OVERFLOW");
+                return false;
+            }
+        }
+    }
+    return true;
+}
+
+int32_t SoftAAC2::outputDelayRingBufferGetSamples(INT_PCM *samples, int32_t numSamples) {
+    if (mOutputDelayRingBufferReadPos + numSamples <= mOutputDelayRingBufferSize
+            && (mOutputDelayRingBufferWritePos < mOutputDelayRingBufferReadPos
+                    || mOutputDelayRingBufferWritePos >= mOutputDelayRingBufferReadPos + numSamples)) {
+        // faster memcopy loop without checks, if the preconditions allow this
+        if (samples != 0) {
+            for (int32_t i = 0; i < numSamples; i++) {
+                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos++];
+            }
+        } else {
+            mOutputDelayRingBufferReadPos += numSamples;
+        }
+        if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
+            mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
+        }
+    } else {
+        ALOGV("slow SoftAAC2::outputDelayRingBufferGetSamples()");
+
+        for (int32_t i = 0; i < numSamples; i++) {
+            if (mOutputDelayRingBufferWritePos == mOutputDelayRingBufferReadPos) {
+                ALOGE("RING BUFFER UNDERRUN");
+                return -1;
+            }
+            if (samples != 0) {
+                samples[i] = mOutputDelayRingBuffer[mOutputDelayRingBufferReadPos];
+            }
+            mOutputDelayRingBufferReadPos++;
+            if (mOutputDelayRingBufferReadPos >= mOutputDelayRingBufferSize) {
+                mOutputDelayRingBufferReadPos -= mOutputDelayRingBufferSize;
+            }
         }
     }
+    return numSamples;
+}
+
+int32_t SoftAAC2::outputDelayRingBufferSamplesAvailable() {
+    int32_t available = mOutputDelayRingBufferWritePos - mOutputDelayRingBufferReadPos;
+    if (available < 0) {
+        available += mOutputDelayRingBufferSize;
+    }
+    if (available < 0) {
+        ALOGE("FATAL RING BUFFER ERROR");
+        return 0;
+    }
+    return available;
+}
+
+int32_t SoftAAC2::outputDelayRingBufferSamplesLeft() {
+    return mOutputDelayRingBufferSize - outputDelayRingBufferSamplesAvailable();
 }
 
 void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
@@ -318,12 +443,11 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
     List<BufferInfo *> &outQueue = getPortQueue(1);
 
     if (portIndex == 0 && mInputBufferCount == 0) {
-        ++mInputBufferCount;
-        BufferInfo *info = *inQueue.begin();
-        OMX_BUFFERHEADERTYPE *header = info->mHeader;
+        BufferInfo *inInfo = *inQueue.begin();
+        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
 
-        inBuffer[0] = header->pBuffer + header->nOffset;
-        inBufferLength[0] = header->nFilledLen;
+        inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+        inBufferLength[0] = inHeader->nFilledLen;
 
         AAC_DECODER_ERROR decoderErr =
             aacDecoder_ConfigRaw(mAACDecoder,
@@ -331,19 +455,25 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
                                  inBufferLength);
 
         if (decoderErr != AAC_DEC_OK) {
+            ALOGW("aacDecoder_ConfigRaw decoderErr = 0x%4.4x", decoderErr);
             mSignalledError = true;
             notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
             return;
         }
 
+        mInputBufferCount++;
+        mOutputBufferCount++; // fake increase of outputBufferCount to keep the counters aligned
+
+        inInfo->mOwnedByUs = false;
         inQueue.erase(inQueue.begin());
-        info->mOwnedByUs = false;
-        notifyEmptyBufferDone(header);
+        inInfo = NULL;
+        notifyEmptyBufferDone(inHeader);
+        inHeader = NULL;
 
+        configureDownmix();
         // Only send out port settings changed event if both sample rate
         // and numChannels are valid.
         if (mStreamInfo->sampleRate && mStreamInfo->numChannels) {
-            maybeConfigureDownmix();
             ALOGI("Initially configuring decoder: %d Hz, %d channels",
                 mStreamInfo->sampleRate,
                 mStreamInfo->numChannels);
@@ -355,202 +485,304 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
         return;
     }
 
-    while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
-        BufferInfo *inInfo = NULL;
-        OMX_BUFFERHEADERTYPE *inHeader = NULL;
+    while ((!inQueue.empty() || mEndOfInput) && !outQueue.empty()) {
         if (!inQueue.empty()) {
-            inInfo = *inQueue.begin();
-            inHeader = inInfo->mHeader;
-        }
+            INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
+            BufferInfo *inInfo = *inQueue.begin();
+            OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
 
-        BufferInfo *outInfo = *outQueue.begin();
-        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
-        outHeader->nFlags = 0;
-
-        if (inHeader) {
             if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
-                mSawInputEos = true;
+                mEndOfInput = true;
+            } else {
+                mEndOfInput = false;
             }
-
-            if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
-                mAnchorTimeUs = inHeader->nTimeStamp;
-                mNumSamplesOutput = 0;
+            if (inHeader->nOffset == 0) { // TODO: does nOffset != 0 happen?
+                mAnchorTimeUs[mInputBufferCount % kNumDelayBlocksMax] =
+                        inHeader->nTimeStamp;
             }
 
-            if (mIsADTS && inHeader->nFilledLen) {
-                size_t adtsHeaderSize = 0;
-                // skip 30 bits, aac_frame_length follows.
-                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
+            if (inHeader->nFilledLen == 0) {
+                inInfo->mOwnedByUs = false;
+                inQueue.erase(inQueue.begin());
+                inInfo = NULL;
+                notifyEmptyBufferDone(inHeader);
+                inHeader = NULL;
+            } else {
+                if (mIsADTS) {
+                    size_t adtsHeaderSize = 0;
+                    // skip 30 bits, aac_frame_length follows.
+                    // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
 
-                const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
+                    const uint8_t *adtsHeader = inHeader->pBuffer + inHeader->nOffset;
 
-                bool signalError = false;
-                if (inHeader->nFilledLen < 7) {
-                    ALOGE("Audio data too short to contain even the ADTS header. "
-                          "Got %d bytes.", inHeader->nFilledLen);
-                    hexdump(adtsHeader, inHeader->nFilledLen);
-                    signalError = true;
-                } else {
-                    bool protectionAbsent = (adtsHeader[1] & 1);
-
-                    unsigned aac_frame_length =
-                        ((adtsHeader[3] & 3) << 11)
-                        | (adtsHeader[4] << 3)
-                        | (adtsHeader[5] >> 5);
-
-                    if (inHeader->nFilledLen < aac_frame_length) {
-                        ALOGE("Not enough audio data for the complete frame. "
-                              "Got %d bytes, frame size according to the ADTS "
-                              "header is %u bytes.",
-                              inHeader->nFilledLen, aac_frame_length);
+                    bool signalError = false;
+                    if (inHeader->nFilledLen < 7) {
+                        ALOGE("Audio data too short to contain even the ADTS header. "
+                                "Got %d bytes.", inHeader->nFilledLen);
                         hexdump(adtsHeader, inHeader->nFilledLen);
                         signalError = true;
                     } else {
-                        adtsHeaderSize = (protectionAbsent ? 7 : 9);
+                        bool protectionAbsent = (adtsHeader[1] & 1);
+
+                        unsigned aac_frame_length =
+                            ((adtsHeader[3] & 3) << 11)
+                            | (adtsHeader[4] << 3)
+                            | (adtsHeader[5] >> 5);
+
+                        if (inHeader->nFilledLen < aac_frame_length) {
+                            ALOGE("Not enough audio data for the complete frame. "
+                                    "Got %d bytes, frame size according to the ADTS "
+                                    "header is %u bytes.",
+                                    inHeader->nFilledLen, aac_frame_length);
+                            hexdump(adtsHeader, inHeader->nFilledLen);
+                            signalError = true;
+                        } else {
+                            adtsHeaderSize = (protectionAbsent ? 7 : 9);
+
+                            inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
+                            inBufferLength[0] = aac_frame_length - adtsHeaderSize;
+
+                            inHeader->nOffset += adtsHeaderSize;
+                            inHeader->nFilledLen -= adtsHeaderSize;
+                        }
+                    }
 
-                        inBuffer[0] = (UCHAR *)adtsHeader + adtsHeaderSize;
-                        inBufferLength[0] = aac_frame_length - adtsHeaderSize;
+                    if (signalError) {
+                        mSignalledError = true;
 
-                        inHeader->nOffset += adtsHeaderSize;
-                        inHeader->nFilledLen -= adtsHeaderSize;
+                        notify(OMX_EventError,
+                               OMX_ErrorStreamCorrupt,
+                               ERROR_MALFORMED,
+                               NULL);
+
+                        return;
                     }
+                } else {
+                    inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+                    inBufferLength[0] = inHeader->nFilledLen;
                 }
 
-                if (signalError) {
-                    mSignalledError = true;
+                // Fill and decode
+                bytesValid[0] = inBufferLength[0];
+
+                INT prevSampleRate = mStreamInfo->sampleRate;
+                INT prevNumChannels = mStreamInfo->numChannels;
+
+                aacDecoder_Fill(mAACDecoder,
+                                inBuffer,
+                                inBufferLength,
+                                bytesValid);
+
+                 // run DRC check
+                 mDrcWrap.submitStreamData(mStreamInfo);
+                 mDrcWrap.update();
 
-                    notify(OMX_EventError,
-                           OMX_ErrorStreamCorrupt,
-                           ERROR_MALFORMED,
-                           NULL);
+                AAC_DECODER_ERROR decoderErr =
+                    aacDecoder_DecodeFrame(mAACDecoder,
+                                           tmpOutBuffer,
+                                           2048 * MAX_CHANNEL_COUNT,
+                                           0 /* flags */);
 
+                if (decoderErr != AAC_DEC_OK) {
+                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
+                }
+
+                if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
+                    ALOGE("AAC_DEC_NOT_ENOUGH_BITS should never happen");
+                    mSignalledError = true;
+                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
                     return;
                 }
-            } else {
-                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
-                inBufferLength[0] = inHeader->nFilledLen;
-            }
-        } else {
-            inBufferLength[0] = 0;
-        }
 
-        // Fill and decode
-        INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(
-                outHeader->pBuffer + outHeader->nOffset);
-
-        bytesValid[0] = inBufferLength[0];
-
-        int prevSampleRate = mStreamInfo->sampleRate;
-        int prevNumChannels = mStreamInfo->numChannels;
-
-        AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;
-        while ((bytesValid[0] > 0 || mSawInputEos) && decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
-            mDecoderHasData |= (bytesValid[0] > 0);
-            aacDecoder_Fill(mAACDecoder,
-                            inBuffer,
-                            inBufferLength,
-                            bytesValid);
-
-            decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
-                                                outBuffer,
-                                                outHeader->nAllocLen,
-                                                0 /* flags */);
-            if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
-                if (mSawInputEos && bytesValid[0] <= 0) {
-                    if (mDecoderHasData) {
-                        // flush out the decoder's delayed data by calling DecodeFrame
-                        // one more time, with the AACDEC_FLUSH flag set
-                        decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
-                                                            outBuffer,
-                                                            outHeader->nAllocLen,
-                                                            AACDEC_FLUSH);
-                        mDecoderHasData = false;
+                if (bytesValid[0] != 0) {
+                    ALOGE("bytesValid[0] != 0 should never happen");
+                    mSignalledError = true;
+                    notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                    return;
+                }
+
+                size_t numOutBytes =
+                    mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
+
+                if (decoderErr == AAC_DEC_OK) {
+                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
+                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
+                        mSignalledError = true;
+                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
+                        return;
                     }
-                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-                    mSignalledOutputEos = true;
-                    break;
+                    UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
+                    inHeader->nFilledLen -= inBufferUsedLength;
+                    inHeader->nOffset += inBufferUsedLength;
                 } else {
-                    ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
-                }
-            }
-        }
+                    ALOGW("AAC decoder returned error 0x%4.4x, substituting silence", decoderErr);
 
-        size_t numOutBytes =
-            mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
+                    memset(tmpOutBuffer, 0, numOutBytes); // TODO: check for overflow
 
-        if (inHeader) {
-            if (decoderErr == AAC_DEC_OK) {
-                UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
-                inHeader->nFilledLen -= inBufferUsedLength;
-                inHeader->nOffset += inBufferUsedLength;
-            } else {
-                ALOGW("AAC decoder returned error %d, substituting silence",
-                      decoderErr);
+                    if (!outputDelayRingBufferPutSamples(tmpOutBuffer,
+                            mStreamInfo->frameSize * mStreamInfo->numChannels)) {
+                        mSignalledError = true;
+                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
+                        return;
+                    }
 
-                memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
+                    // Discard input buffer.
+                    inHeader->nFilledLen = 0;
 
-                // Discard input buffer.
-                inHeader->nFilledLen = 0;
+                    aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
 
-                aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
+                    // fall through
+                }
 
-                // fall through
+                /*
+                 * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
+                 * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
+                 * rate system and the sampling rate in the final output is actually
+                 * doubled compared with the core AAC decoder sampling rate.
+                 *
+                 * Explicit signalling is done by explicitly defining SBR audio object
+                 * type in the bitstream. Implicit signalling is done by embedding
+                 * SBR content in AAC extension payload specific to SBR, and hence
+                 * requires an AAC decoder to perform pre-checks on actual audio frames.
+                 *
+                 * Thus, we could not say for sure whether a stream is
+                 * AAC+/eAAC+ until the first data frame is decoded.
+                 */
+                if (mOutputBufferCount > 1) {
+                    if (mStreamInfo->sampleRate != prevSampleRate ||
+                        mStreamInfo->numChannels != prevNumChannels) {
+                        ALOGE("can not reconfigure AAC output");
+                        mSignalledError = true;
+                        notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
+                        return;
+                    }
+                }
+                if (mInputBufferCount <= 2) { // TODO: <= 1
+                    if (mStreamInfo->sampleRate != prevSampleRate ||
+                        mStreamInfo->numChannels != prevNumChannels) {
+                        ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
+                              prevSampleRate, mStreamInfo->sampleRate,
+                              prevNumChannels, mStreamInfo->numChannels);
+
+                        notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
+                        mOutputPortSettingsChange = AWAITING_DISABLED;
+
+                        if (inHeader->nFilledLen == 0) {
+                            inInfo->mOwnedByUs = false;
+                            mInputBufferCount++;
+                            inQueue.erase(inQueue.begin());
+                            inInfo = NULL;
+                            notifyEmptyBufferDone(inHeader);
+                            inHeader = NULL;
+                        }
+                        return;
+                    }
+                } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
+                    ALOGW("Invalid AAC stream");
+                    mSignalledError = true;
+                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
+                    return;
+                }
+                if (inHeader->nFilledLen == 0) {
+                    inInfo->mOwnedByUs = false;
+                    mInputBufferCount++;
+                    inQueue.erase(inQueue.begin());
+                    inInfo = NULL;
+                    notifyEmptyBufferDone(inHeader);
+                    inHeader = NULL;
+                } else {
+                    ALOGW("inHeader->nFilledLen = %d", inHeader->nFilledLen);
+                }
             }
+        }
 
-            if (inHeader->nFilledLen == 0) {
-                inInfo->mOwnedByUs = false;
-                inQueue.erase(inQueue.begin());
-                inInfo = NULL;
-                notifyEmptyBufferDone(inHeader);
-                inHeader = NULL;
+        int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
+
+        if (!mEndOfInput && mOutputDelayCompensated < outputDelay) {
+            // discard outputDelay at the beginning
+            int32_t toCompensate = outputDelay - mOutputDelayCompensated;
+            int32_t discard = outputDelayRingBufferSamplesAvailable();
+            if (discard > toCompensate) {
+                discard = toCompensate;
             }
+            int32_t discarded = outputDelayRingBufferGetSamples(0, discard);
+            mOutputDelayCompensated += discarded;
+            continue;
         }
 
-        /*
-         * AAC+/eAAC+ streams can be signalled in two ways: either explicitly
-         * or implicitly, according to MPEG4 spec. AAC+/eAAC+ is a dual
-         * rate system and the sampling rate in the final output is actually
-         * doubled compared with the core AAC decoder sampling rate.
-         *
-         * Explicit signalling is done by explicitly defining SBR audio object
-         * type in the bitstream. Implicit signalling is done by embedding
-         * SBR content in AAC extension payload specific to SBR, and hence
-         * requires an AAC decoder to perform pre-checks on actual audio frames.
-         *
-         * Thus, we could not say for sure whether a stream is
-         * AAC+/eAAC+ until the first data frame is decoded.
-         */
-        if (mInputBufferCount <= 2) {
-            if (mStreamInfo->sampleRate != prevSampleRate ||
-                mStreamInfo->numChannels != prevNumChannels) {
-                maybeConfigureDownmix();
-                ALOGI("Reconfiguring decoder: %d->%d Hz, %d->%d channels",
-                      prevSampleRate, mStreamInfo->sampleRate,
-                      prevNumChannels, mStreamInfo->numChannels);
-
-                notify(OMX_EventPortSettingsChanged, 1, 0, NULL);
-                mOutputPortSettingsChange = AWAITING_DISABLED;
-                return;
+        if (mEndOfInput) {
+            while (mOutputDelayCompensated > 0) {
+                // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
+                INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
+ 
+                 // run DRC check
+                 mDrcWrap.submitStreamData(mStreamInfo);
+                 mDrcWrap.update();
+
+                AAC_DECODER_ERROR decoderErr =
+                    aacDecoder_DecodeFrame(mAACDecoder,
+                                           tmpOutBuffer,
+                                           2048 * MAX_CHANNEL_COUNT,
+                                           AACDEC_FLUSH);
+                if (decoderErr != AAC_DEC_OK) {
+                    ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
+                }
+
+                int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
+                if (tmpOutBufferSamples > mOutputDelayCompensated) {
+                    tmpOutBufferSamples = mOutputDelayCompensated;
+                }
+                outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
+                mOutputDelayCompensated -= tmpOutBufferSamples;
             }
-        } else if (!mStreamInfo->sampleRate || !mStreamInfo->numChannels) {
-            ALOGW("Invalid AAC stream");
-            mSignalledError = true;
-            notify(OMX_EventError, OMX_ErrorUndefined, decoderErr, NULL);
-            return;
         }
 
-        if (decoderErr == AAC_DEC_OK || mNumSamplesOutput > 0) {
-            // We'll only output data if we successfully decoded it or
-            // we've previously decoded valid data, in the latter case
-            // (decode failed) we'll output a silent frame.
-            outHeader->nFilledLen = numOutBytes;
+        while (!outQueue.empty()
+                && outputDelayRingBufferSamplesAvailable()
+                        >= mStreamInfo->frameSize * mStreamInfo->numChannels) {
+            BufferInfo *outInfo = *outQueue.begin();
+            OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
 
-            outHeader->nTimeStamp =
-                mAnchorTimeUs
-                    + (mNumSamplesOutput * 1000000ll) / mStreamInfo->sampleRate;
+            if (outHeader->nOffset != 0) {
+                ALOGE("outHeader->nOffset != 0 is not handled");
+                mSignalledError = true;
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
+            }
+
+            INT_PCM *outBuffer =
+                    reinterpret_cast<INT_PCM *>(outHeader->pBuffer + outHeader->nOffset);
+            if (outHeader->nOffset
+                    + mStreamInfo->frameSize * mStreamInfo->numChannels * sizeof(int16_t)
+                    > outHeader->nAllocLen) {
+                ALOGE("buffer overflow");
+                mSignalledError = true;
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
 
-            mNumSamplesOutput += mStreamInfo->frameSize;
+            }
+            int32_t ns = outputDelayRingBufferGetSamples(outBuffer,
+                    mStreamInfo->frameSize * mStreamInfo->numChannels); // TODO: check for overflow
+            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
+                ALOGE("not a complete frame of samples available");
+                mSignalledError = true;
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
+            }
+
+            outHeader->nFilledLen = mStreamInfo->frameSize * mStreamInfo->numChannels
+                    * sizeof(int16_t);
+            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
+                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+                mEndOfOutput = true;
+            } else {
+                outHeader->nFlags = 0;
+            }
 
+            outHeader->nTimeStamp = mAnchorTimeUs[mOutputBufferCount
+                    % kNumDelayBlocksMax];
+
+            mOutputBufferCount++;
             outInfo->mOwnedByUs = false;
             outQueue.erase(outQueue.begin());
             outInfo = NULL;
@@ -558,8 +790,48 @@ void SoftAAC2::onQueueFilled(OMX_U32 portIndex) {
             outHeader = NULL;
         }
 
-        if (decoderErr == AAC_DEC_OK) {
-            ++mInputBufferCount;
+        if (mEndOfInput) {
+            if (outputDelayRingBufferSamplesAvailable() > 0
+                    && outputDelayRingBufferSamplesAvailable()
+                            < mStreamInfo->frameSize * mStreamInfo->numChannels) {
+                ALOGE("not a complete frame of samples available");
+                mSignalledError = true;
+                notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                return;
+            }
+
+            if (mEndOfInput && !outQueue.empty() && outputDelayRingBufferSamplesAvailable() == 0) {
+                if (!mEndOfOutput) {
+                    // send empty block signaling EOS
+                    mEndOfOutput = true;
+                    BufferInfo *outInfo = *outQueue.begin();
+                    OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+                    if (outHeader->nOffset != 0) {
+                        ALOGE("outHeader->nOffset != 0 is not handled");
+                        mSignalledError = true;
+                        notify(OMX_EventError, OMX_ErrorUndefined, 0, NULL);
+                        return;
+                    }
+
+                    INT_PCM *outBuffer = reinterpret_cast<INT_PCM *>(outHeader->pBuffer
+                            + outHeader->nOffset);
+                    int32_t ns = 0;
+                    outHeader->nFilledLen = 0;
+                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+
+                    outHeader->nTimeStamp = mAnchorTimeUs[mOutputBufferCount
+                            % kNumDelayBlocksMax];
+
+                    mOutputBufferCount++;
+                    outInfo->mOwnedByUs = false;
+                    outQueue.erase(outQueue.begin());
+                    outInfo = NULL;
+                    notifyFillBufferDone(outHeader);
+                    outHeader = NULL;
+                }
+                break; // if outQueue not empty but no more output
+            }
         }
     }
 }
@@ -574,34 +846,67 @@ void SoftAAC2::onPortFlushCompleted(OMX_U32 portIndex) {
         // but only if initialization has already happened.
         if (mInputBufferCount != 0) {
             mInputBufferCount = 1;
-            mStreamInfo->sampleRate = 0;
         }
+    } else {
+        while (outputDelayRingBufferSamplesAvailable() > 0) {
+            int32_t ns = outputDelayRingBufferGetSamples(0,
+                    mStreamInfo->frameSize * mStreamInfo->numChannels);
+            if (ns != mStreamInfo->frameSize * mStreamInfo->numChannels) {
+                ALOGE("not a complete frame of samples available");
+            }
+            mOutputBufferCount++;
+        }
+        mOutputDelayRingBufferReadPos = mOutputDelayRingBufferWritePos;
     }
 }
 
 void SoftAAC2::drainDecoder() {
-    // a buffer big enough for 6 channels of decoded HE-AAC
-    short buf [2048*6];
-    aacDecoder_DecodeFrame(mAACDecoder,
-            buf, sizeof(buf), AACDEC_FLUSH | AACDEC_CLRHIST | AACDEC_INTR);
-    aacDecoder_DecodeFrame(mAACDecoder,
-            buf, sizeof(buf), AACDEC_FLUSH | AACDEC_CLRHIST | AACDEC_INTR);
-    aacDecoder_SetParam(mAACDecoder, AAC_TPDEC_CLEAR_BUFFER, 1);
-    mDecoderHasData = false;
+    int32_t outputDelay = mStreamInfo->outputDelay * mStreamInfo->numChannels;
+
+    // flush decoder until outputDelay is compensated
+    while (mOutputDelayCompensated > 0) {
+        // a buffer big enough for MAX_CHANNEL_COUNT channels of decoded HE-AAC
+        INT_PCM tmpOutBuffer[2048 * MAX_CHANNEL_COUNT];
+
+        // run DRC check
+        mDrcWrap.submitStreamData(mStreamInfo);
+        mDrcWrap.update();
+
+        AAC_DECODER_ERROR decoderErr =
+            aacDecoder_DecodeFrame(mAACDecoder,
+                                   tmpOutBuffer,
+                                   2048 * MAX_CHANNEL_COUNT,
+                                   AACDEC_FLUSH);
+        if (decoderErr != AAC_DEC_OK) {
+            ALOGW("aacDecoder_DecodeFrame decoderErr = 0x%4.4x", decoderErr);
+        }
+
+        int32_t tmpOutBufferSamples = mStreamInfo->frameSize * mStreamInfo->numChannels;
+        if (tmpOutBufferSamples > mOutputDelayCompensated) {
+            tmpOutBufferSamples = mOutputDelayCompensated;
+        }
+        outputDelayRingBufferPutSamples(tmpOutBuffer, tmpOutBufferSamples);
+
+        mOutputDelayCompensated -= tmpOutBufferSamples;
+    }
 }
 
 void SoftAAC2::onReset() {
     drainDecoder();
     // reset the "configured" state
     mInputBufferCount = 0;
-    mNumSamplesOutput = 0;
+    mOutputBufferCount = 0;
+    mOutputDelayCompensated = 0;
+    mOutputDelayRingBufferWritePos = 0;
+    mOutputDelayRingBufferReadPos = 0;
+    mEndOfInput = false;
+    mEndOfOutput = false;
+
     // To make the codec behave the same before and after a reset, we need to invalidate the
     // streaminfo struct. This does that:
-    mStreamInfo->sampleRate = 0;
+    mStreamInfo->sampleRate = 0; // TODO: mStreamInfo is read only
 
     mSignalledError = false;
-    mSawInputEos = false;
-    mSignalledOutputEos = false;
     mOutputPortSettingsChange = NONE;
 }
 
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.h b/media/libstagefright/codecs/aacdec/SoftAAC2.h
index a7ea1e2..5cde03a 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -20,6 +20,7 @@
 #include "SimpleSoftOMXComponent.h"
 
 #include "aacdecoder_lib.h"
+#include "DrcPresModeWrap.h"
 
 namespace android {
 
@@ -47,18 +48,19 @@ private:
     enum {
         kNumInputBuffers        = 4,
         kNumOutputBuffers       = 4,
+        kNumDelayBlocksMax      = 8,
     };
 
     HANDLE_AACDECODER mAACDecoder;
     CStreamInfo *mStreamInfo;
     bool mIsADTS;
-    bool mDecoderHasData;
+    bool mIsFirst;
     size_t mInputBufferCount;
+    size_t mOutputBufferCount;
     bool mSignalledError;
-    bool mSawInputEos;
-    bool mSignalledOutputEos;
-    int64_t mAnchorTimeUs;
-    int64_t mNumSamplesOutput;
+    int64_t mAnchorTimeUs[kNumDelayBlocksMax];
+
+    CDrcPresModeWrapper mDrcWrap;
 
     enum {
         NONE,
@@ -69,9 +71,22 @@ private:
     void initPorts();
     status_t initDecoder();
     bool isConfigured() const;
-    void maybeConfigureDownmix() const;
+    void configureDownmix() const;
     void drainDecoder();
 
+//      delay compensation
+    bool mEndOfInput;
+    bool mEndOfOutput;
+    int32_t mOutputDelayCompensated;
+    int32_t mOutputDelayRingBufferSize;
+    short *mOutputDelayRingBuffer;
+    int32_t mOutputDelayRingBufferWritePos;
+    int32_t mOutputDelayRingBufferReadPos;
+    bool outputDelayRingBufferPutSamples(INT_PCM *samples, int numSamples);
+    int32_t outputDelayRingBufferGetSamples(INT_PCM *samples, int numSamples);
+    int32_t outputDelayRingBufferSamplesAvailable();
+    int32_t outputDelayRingBufferSamplesLeft();
+
     DISALLOW_EVIL_CONSTRUCTORS(SoftAAC2);
 };