diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 613435d..7703058 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -38,7 +38,7 @@
 static const int32_t kAudioBitRate = 12200;
 static const int64_t kDurationUs = 10000000LL;  // 10 seconds
 
-#if 1
+#if 0
 class DummySource : public MediaSource {
 
 public:
@@ -318,7 +318,7 @@
 
     sp<MetaData> encMeta = new MetaData;
     encMeta->setCString(kKeyMIMEType,
-            1 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
+            0 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
     encMeta->setInt32(kKeySampleRate, kSampleRate);
     encMeta->setInt32(kKeyChannelCount, kNumChannels);
     encMeta->setInt32(kKeyMaxInputSize, 8192);
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 84f8282..4c30e04 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -172,6 +172,7 @@
     uint32_t mFlags;
 
     bool mIsEncoder;
+    bool mIsVideo;
     char *mMIME;
     char *mComponentName;
     sp<MetaData> mOutputFormat;
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 0aeb515..4d61067 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -77,7 +77,6 @@
 
 LOCAL_STATIC_LIBRARIES := \
         libstagefright_color_conversion \
-        libstagefright_aacenc \
         libstagefright_amrnbenc \
         libstagefright_amrwbenc \
         libstagefright_avcenc \
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 60d9bb7..7597f64 100755
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -18,7 +18,6 @@
 #define LOG_TAG "OMXCodec"
 #include <utils/Log.h>
 
-#include "include/AACEncoder.h"
 #include "include/AMRNBEncoder.h"
 #include "include/AMRWBEncoder.h"
 #include "include/AVCEncoder.h"
@@ -73,7 +72,6 @@
 
 FACTORY_CREATE_ENCODER(AMRNBEncoder)
 FACTORY_CREATE_ENCODER(AMRWBEncoder)
-FACTORY_CREATE_ENCODER(AACEncoder)
 FACTORY_CREATE_ENCODER(AVCEncoder)
 FACTORY_CREATE_ENCODER(M4vH263Encoder)
 
@@ -88,7 +86,6 @@
     static const FactoryInfo kFactoryInfo[] = {
         FACTORY_REF(AMRNBEncoder)
         FACTORY_REF(AMRWBEncoder)
-        FACTORY_REF(AACEncoder)
         FACTORY_REF(AVCEncoder)
         FACTORY_REF(M4vH263Encoder)
     };
@@ -153,7 +150,7 @@
     { MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.TI.WBAMR.encode" },
     { MEDIA_MIMETYPE_AUDIO_AMR_WB, "AMRWBEncoder" },
     { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.TI.AAC.encode" },
-    { MEDIA_MIMETYPE_AUDIO_AAC, "AACEncoder" },
+    { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.google.aac.encoder" },
     { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.DUCATI1.VIDEO.MPEG4E" },
     { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.7x30.video.encoder.mpeg4" },
     { MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.video.encoder.mpeg4" },
@@ -1487,6 +1484,7 @@
       mQuirks(quirks),
       mFlags(flags),
       mIsEncoder(isEncoder),
+      mIsVideo(!strncasecmp("video/", mime, 6)),
       mMIME(strdup(mime)),
       mComponentName(strdup(componentName)),
       mSource(source),
@@ -2192,7 +2190,7 @@
 }
 
 int64_t OMXCodec::retrieveDecodingTimeUs(bool isCodecSpecific) {
-    CHECK(mIsEncoder);
+    CHECK(mIsEncoder && mIsVideo);
 
     if (mDecodingTimeList.empty()) {
         CHECK(mSignalledEOS || mNoMoreOutputData);
@@ -2387,7 +2385,7 @@
                     mNoMoreOutputData = true;
                 }
 
-                if (mIsEncoder) {
+                if (mIsEncoder && mIsVideo) {
                     int64_t decodingTimeUs = retrieveDecodingTimeUs(isCodecSpecific);
                     buffer->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
                 }
@@ -3249,7 +3247,7 @@
         int64_t lastBufferTimeUs;
         CHECK(srcBuffer->meta_data()->findInt64(kKeyTime, &lastBufferTimeUs));
         CHECK(lastBufferTimeUs >= 0);
-        if (mIsEncoder) {
+        if (mIsEncoder && mIsVideo) {
             mDecodingTimeList.push_back(lastBufferTimeUs);
         }
 
diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index 8318ba4..34a2796 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -85,3 +85,29 @@
 endif
 
 include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+        SoftAACEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+        frameworks/base/media/libstagefright/include \
+        frameworks/base/include/media/stagefright/openmax \
+	frameworks/base/media/libstagefright/codecs/common/include \
+
+LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
+
+LOCAL_STATIC_LIBRARIES := \
+        libstagefright_aacenc
+
+LOCAL_SHARED_LIBRARIES := \
+        libstagefright_omx libstagefright_foundation libutils \
+        libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_aacenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
new file mode 100644
index 0000000..c6724c2
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAACEncoder"
+#include <utils/Log.h>
+
+#include "SoftAACEncoder.h"
+
+#include "voAAC.h"
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+template<class T>
+static void InitOMXParams(T *params) {
+    params->nSize = sizeof(T);
+    params->nVersion.s.nVersionMajor = 1;
+    params->nVersion.s.nVersionMinor = 0;
+    params->nVersion.s.nRevision = 0;
+    params->nVersion.s.nStep = 0;
+}
+
+SoftAACEncoder::SoftAACEncoder(
+        const char *name,
+        const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData,
+        OMX_COMPONENTTYPE **component)
+    : SimpleSoftOMXComponent(name, callbacks, appData, component),
+      mEncoderHandle(NULL),
+      mApiHandle(NULL),
+      mMemOperator(NULL),
+      mNumChannels(1),
+      mSampleRate(44100),
+      mBitRate(0),
+      mSentCodecSpecificData(false),
+      mInputSize(0),
+      mInputFrame(NULL),
+      mInputTimeUs(-1ll),
+      mSawInputEOS(false),
+      mSignalledError(false) {
+    initPorts();
+    CHECK_EQ(initEncoder(), (status_t)OK);
+
+    setAudioParams();
+}
+
+SoftAACEncoder::~SoftAACEncoder() {
+    delete[] mInputFrame;
+    mInputFrame = NULL;
+
+    if (mEncoderHandle) {
+        CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+        mEncoderHandle = NULL;
+    }
+
+    delete mApiHandle;
+    mApiHandle = NULL;
+
+    delete mMemOperator;
+    mMemOperator = NULL;
+}
+
+void SoftAACEncoder::initPorts() {
+    OMX_PARAM_PORTDEFINITIONTYPE def;
+    InitOMXParams(&def);
+
+    def.nPortIndex = 0;
+    def.eDir = OMX_DirInput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t) * 2;
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 1;
+
+    def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+    addPort(def);
+
+    def.nPortIndex = 1;
+    def.eDir = OMX_DirOutput;
+    def.nBufferCountMin = kNumBuffers;
+    def.nBufferCountActual = def.nBufferCountMin;
+    def.nBufferSize = 8192;
+    def.bEnabled = OMX_TRUE;
+    def.bPopulated = OMX_FALSE;
+    def.eDomain = OMX_PortDomainAudio;
+    def.bBuffersContiguous = OMX_FALSE;
+    def.nBufferAlignment = 2;
+
+    def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
+    def.format.audio.pNativeRender = NULL;
+    def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+    def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+    addPort(def);
+}
+
+status_t SoftAACEncoder::initEncoder() {
+    mApiHandle = new VO_AUDIO_CODECAPI;
+
+    if (VO_ERR_NONE != voGetAACEncAPI(mApiHandle)) {
+        ALOGE("Failed to get api handle");
+        return UNKNOWN_ERROR;
+    }
+
+    mMemOperator = new VO_MEM_OPERATOR;
+    mMemOperator->Alloc = cmnMemAlloc;
+    mMemOperator->Copy = cmnMemCopy;
+    mMemOperator->Free = cmnMemFree;
+    mMemOperator->Set = cmnMemSet;
+    mMemOperator->Check = cmnMemCheck;
+
+    VO_CODEC_INIT_USERDATA userData;
+    memset(&userData, 0, sizeof(userData));
+    userData.memflag = VO_IMF_USERMEMOPERATOR;
+    userData.memData = (VO_PTR) mMemOperator;
+    if (VO_ERR_NONE !=
+            mApiHandle->Init(&mEncoderHandle, VO_AUDIO_CodingAAC, &userData)) {
+        ALOGE("Failed to init AAC encoder");
+        return UNKNOWN_ERROR;
+    }
+
+    return OK;
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalGetParameter(
+        OMX_INDEXTYPE index, OMX_PTR params) {
+    switch (index) {
+        case OMX_IndexParamAudioPortFormat:
+        {
+            OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+                (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+            if (formatParams->nPortIndex > 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            if (formatParams->nIndex > 0) {
+                return OMX_ErrorNoMore;
+            }
+
+            formatParams->eEncoding =
+                (formatParams->nPortIndex == 0)
+                    ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAAC;
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioAac:
+        {
+            OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+                (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+            if (aacParams->nPortIndex != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            aacParams->nBitRate = mBitRate;
+            aacParams->nAudioBandWidth = 0;
+            aacParams->nAACtools = 0;
+            aacParams->nAACERtools = 0;
+            aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
+            aacParams->eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4FF;
+            aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
+
+            aacParams->nChannels = mNumChannels;
+            aacParams->nSampleRate = mSampleRate;
+            aacParams->nFrameLength = 0;
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioPcm:
+        {
+            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex != 0) {
+                return OMX_ErrorUndefined;
+            }
+
+            pcmParams->eNumData = OMX_NumericalDataSigned;
+            pcmParams->eEndian = OMX_EndianBig;
+            pcmParams->bInterleaved = OMX_TRUE;
+            pcmParams->nBitPerSample = 16;
+            pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+            pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+            pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+
+            pcmParams->nChannels = mNumChannels;
+            pcmParams->nSamplingRate = mSampleRate;
+
+            return OMX_ErrorNone;
+        }
+
+        default:
+            return SimpleSoftOMXComponent::internalGetParameter(index, params);
+    }
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalSetParameter(
+        OMX_INDEXTYPE index, const OMX_PTR params) {
+    switch (index) {
+        case OMX_IndexParamStandardComponentRole:
+        {
+            const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+                (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+            if (strncmp((const char *)roleParams->cRole,
+                        "audio_encoder.aac",
+                        OMX_MAX_STRINGNAME_SIZE - 1)) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioPortFormat:
+        {
+            const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+                (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+            if (formatParams->nPortIndex > 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            if (formatParams->nIndex > 0) {
+                return OMX_ErrorNoMore;
+            }
+
+            if ((formatParams->nPortIndex == 0
+                        && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+                || (formatParams->nPortIndex == 1
+                        && formatParams->eEncoding != OMX_AUDIO_CodingAAC)) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioAac:
+        {
+            OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+                (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+            if (aacParams->nPortIndex != 1) {
+                return OMX_ErrorUndefined;
+            }
+
+            mBitRate = aacParams->nBitRate;
+            mNumChannels = aacParams->nChannels;
+            mSampleRate = aacParams->nSampleRate;
+
+            if (setAudioParams() != OK) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+        case OMX_IndexParamAudioPcm:
+        {
+            OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+                (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+            if (pcmParams->nPortIndex != 0) {
+                return OMX_ErrorUndefined;
+            }
+
+            mNumChannels = pcmParams->nChannels;
+            mSampleRate = pcmParams->nSamplingRate;
+
+            if (setAudioParams() != OK) {
+                return OMX_ErrorUndefined;
+            }
+
+            return OMX_ErrorNone;
+        }
+
+
+        default:
+            return SimpleSoftOMXComponent::internalSetParameter(index, params);
+    }
+}
+
+status_t SoftAACEncoder::setAudioParams() {
+    // We call this whenever sample rate, number of channels or bitrate change
+    // in reponse to setParameter calls.
+
+    ALOGV("setAudioParams: %lu Hz, %lu channels, %lu bps",
+         mSampleRate, mNumChannels, mBitRate);
+
+    status_t err = setAudioSpecificConfigData();
+
+    if (err != OK) {
+        return err;
+    }
+
+    AACENC_PARAM params;
+    memset(&params, 0, sizeof(params));
+    params.sampleRate = mSampleRate;
+    params.bitRate = mBitRate;
+    params.nChannels = mNumChannels;
+    params.adtsUsed = 0;  // We add adts header in the file writer if needed.
+    if (VO_ERR_NONE != mApiHandle->SetParam(
+                mEncoderHandle, VO_PID_AAC_ENCPARAM,  &params)) {
+        ALOGE("Failed to set AAC encoder parameters");
+        return UNKNOWN_ERROR;
+    }
+
+    return OK;
+}
+
+static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) {
+    static const int32_t kSampleRateTable[] = {
+        96000, 88200, 64000, 48000, 44100, 32000,
+        24000, 22050, 16000, 12000, 11025, 8000
+    };
+    const int32_t tableSize =
+        sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);
+
+    for (int32_t i = 0; i < tableSize; ++i) {
+        if (sampleRate == kSampleRateTable[i]) {
+            index = i;
+            return OK;
+        }
+    }
+
+    return UNKNOWN_ERROR;
+}
+
+status_t SoftAACEncoder::setAudioSpecificConfigData() {
+    // The AAC encoder's audio specific config really only encodes
+    // number of channels and the sample rate (mapped to an index into
+    // a fixed sample rate table).
+
+    int32_t index;
+    status_t err = getSampleRateTableIndex(mSampleRate, index);
+    if (err != OK) {
+        ALOGE("Unsupported sample rate (%lu Hz)", mSampleRate);
+        return err;
+    }
+
+    if (mNumChannels > 2 || mNumChannels <= 0) {
+        ALOGE("Unsupported number of channels(%lu)", mNumChannels);
+        return UNKNOWN_ERROR;
+    }
+
+    // OMX_AUDIO_AACObjectLC
+    mAudioSpecificConfigData[0] = ((0x02 << 3) | (index >> 1));
+    mAudioSpecificConfigData[1] = ((index & 0x01) << 7) | (mNumChannels << 3);
+
+    return OK;
+}
+
+void SoftAACEncoder::onQueueFilled(OMX_U32 portIndex) {
+    if (mSignalledError) {
+        return;
+    }
+
+    List<BufferInfo *> &inQueue = getPortQueue(0);
+    List<BufferInfo *> &outQueue = getPortQueue(1);
+
+    if (!mSentCodecSpecificData) {
+        // The very first thing we want to output is the codec specific
+        // data. It does not require any input data but we will need an
+        // output buffer to store it in.
+
+        if (outQueue.empty()) {
+            return;
+        }
+
+        BufferInfo *outInfo = *outQueue.begin();
+        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+        outHeader->nFilledLen = sizeof(mAudioSpecificConfigData);
+        outHeader->nFlags = OMX_BUFFERFLAG_CODECCONFIG;
+
+        uint8_t *out = outHeader->pBuffer + outHeader->nOffset;
+        memcpy(out, mAudioSpecificConfigData, sizeof(mAudioSpecificConfigData));
+
+#if 0
+        ALOGI("sending codec specific data.");
+        hexdump(out, sizeof(mAudioSpecificConfigData));
+#endif
+
+        outQueue.erase(outQueue.begin());
+        outInfo->mOwnedByUs = false;
+        notifyFillBufferDone(outHeader);
+
+        mSentCodecSpecificData = true;
+    }
+
+    size_t numBytesPerInputFrame =
+        mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);
+
+    for (;;) {
+        // We do the following until we run out of buffers.
+
+        while (mInputSize < numBytesPerInputFrame) {
+            // As long as there's still input data to be read we
+            // will drain "kNumSamplesPerFrame * mNumChannels" samples
+            // into the "mInputFrame" buffer and then encode those
+            // as a unit into an output buffer.
+
+            if (mSawInputEOS || inQueue.empty()) {
+                return;
+            }
+
+            BufferInfo *inInfo = *inQueue.begin();
+            OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+            const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+            size_t copy = numBytesPerInputFrame - mInputSize;
+            if (copy > inHeader->nFilledLen) {
+                copy = inHeader->nFilledLen;
+            }
+
+            if (mInputFrame == NULL) {
+                mInputFrame = new int16_t[kNumSamplesPerFrame * mNumChannels];
+            }
+
+            if (mInputSize == 0) {
+                mInputTimeUs = inHeader->nTimeStamp;
+            }
+
+            memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+            mInputSize += copy;
+
+            inHeader->nOffset += copy;
+            inHeader->nFilledLen -= copy;
+
+            // "Time" on the input buffer has in effect advanced by the
+            // number of audio frames we just advanced nOffset by.
+            inHeader->nTimeStamp +=
+                (copy * 1000000ll / mSampleRate)
+                    / (mNumChannels * sizeof(int16_t));
+
+            if (inHeader->nFilledLen == 0) {
+                if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+                    ALOGV("saw input EOS");
+                    mSawInputEOS = true;
+
+                    // Pad any remaining data with zeroes.
+                    memset((uint8_t *)mInputFrame + mInputSize,
+                           0,
+                           numBytesPerInputFrame - mInputSize);
+
+                    mInputSize = numBytesPerInputFrame;
+                }
+
+                inQueue.erase(inQueue.begin());
+                inInfo->mOwnedByUs = false;
+                notifyEmptyBufferDone(inHeader);
+
+                inData = NULL;
+                inHeader = NULL;
+                inInfo = NULL;
+            }
+        }
+
+        // At this  point we have all the input data necessary to encode
+        // a single frame, all we need is an output buffer to store the result
+        // in.
+
+        if (outQueue.empty()) {
+            return;
+        }
+
+        BufferInfo *outInfo = *outQueue.begin();
+        OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+        VO_CODECBUFFER inputData;
+        memset(&inputData, 0, sizeof(inputData));
+        inputData.Buffer = (unsigned char *)mInputFrame;
+        inputData.Length = numBytesPerInputFrame;
+        CHECK(VO_ERR_NONE ==
+                mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+        VO_CODECBUFFER outputData;
+        memset(&outputData, 0, sizeof(outputData));
+        VO_AUDIO_OUTPUTINFO outputInfo;
+        memset(&outputInfo, 0, sizeof(outputInfo));
+
+        uint8_t *outPtr = (uint8_t *)outHeader->pBuffer + outHeader->nOffset;
+        size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+        VO_U32 ret = VO_ERR_NONE;
+        size_t nOutputBytes = 0;
+        do {
+            outputData.Buffer = outPtr;
+            outputData.Length = outAvailable - nOutputBytes;
+            ret = mApiHandle->GetOutputData(
+                    mEncoderHandle, &outputData, &outputInfo);
+            if (ret == VO_ERR_NONE) {
+                outPtr += outputData.Length;
+                nOutputBytes += outputData.Length;
+            }
+        } while (ret != VO_ERR_INPUT_BUFFER_SMALL);
+
+        outHeader->nFilledLen = nOutputBytes;
+
+        outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+        if (mSawInputEOS) {
+            // We also tag this output buffer with EOS if it corresponds
+            // to the final input buffer.
+            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+        }
+
+        outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+        ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+              nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+        hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+        outQueue.erase(outQueue.begin());
+        outInfo->mOwnedByUs = false;
+        notifyFillBufferDone(outHeader);
+
+        outHeader = NULL;
+        outInfo = NULL;
+
+        mInputSize = 0;
+    }
+}
+
+}  // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+        const char *name, const OMX_CALLBACKTYPE *callbacks,
+        OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+    return new android::SoftAACEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.h b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
new file mode 100644
index 0000000..d148eb7
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+#ifndef SOFT_AAC_ENCODER_H_
+
+#define SOFT_AAC_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAACEncoder : public SimpleSoftOMXComponent {
+    SoftAACEncoder(
+            const char *name,
+            const OMX_CALLBACKTYPE *callbacks,
+            OMX_PTR appData,
+            OMX_COMPONENTTYPE **component);
+
+protected:
+    virtual ~SoftAACEncoder();
+
+    virtual OMX_ERRORTYPE internalGetParameter(
+            OMX_INDEXTYPE index, OMX_PTR params);
+
+    virtual OMX_ERRORTYPE internalSetParameter(
+            OMX_INDEXTYPE index, const OMX_PTR params);
+
+    virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+    enum {
+        kNumBuffers             = 4,
+        kNumSamplesPerFrame     = 1024,
+    };
+
+    void *mEncoderHandle;
+    VO_AUDIO_CODECAPI *mApiHandle;
+    VO_MEM_OPERATOR  *mMemOperator;
+
+    OMX_U32 mNumChannels;
+    OMX_U32 mSampleRate;
+    OMX_U32 mBitRate;
+
+    bool mSentCodecSpecificData;
+    size_t mInputSize;
+    int16_t *mInputFrame;
+    int64_t mInputTimeUs;
+
+    bool mSawInputEOS;
+
+    uint8_t mAudioSpecificConfigData[2];
+
+    bool mSignalledError;
+
+    void initPorts();
+    status_t initEncoder();
+
+    status_t setAudioSpecificConfigData();
+    status_t setAudioParams();
+
+    DISALLOW_EVIL_CONSTRUCTORS(SoftAACEncoder);
+};
+
+}  // namespace android
+
+#endif  // SOFT_AAC_ENCODER_H_
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index da3ae42..cf9e8c9 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -35,6 +35,7 @@
 
 } kComponents[] = {
     { "OMX.google.aac.decoder", "aacdec", "audio_decoder.aac" },
+    { "OMX.google.aac.encoder", "aacenc", "audio_encoder.aac" },
     { "OMX.google.amrnb.decoder", "amrdec", "audio_decoder.amrnb" },
     { "OMX.google.amrwb.decoder", "amrdec", "audio_decoder.amrwb" },
     { "OMX.google.h264.decoder", "h264dec", "video_decoder.avc" },
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 96bc552..2d856ad 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1866,7 +1866,7 @@
 bool AudioFlinger::MixerThread::threadLoop()
 {
     Vector< sp<Track> > tracksToRemove;
-    uint32_t mixerStatus = MIXER_IDLE;
+    mixer_state mixerStatus = MIXER_IDLE;
     nsecs_t standbyTime = systemTime();
     size_t mixBufferSize = mFrameCount * mFrameSize;
     // FIXME: Relaxed timing because of a certain device that can't meet latency
@@ -2085,10 +2085,11 @@
 }
 
 // prepareTracks_l() must be called with ThreadBase::mLock held
-uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+        const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
 {
 
-    uint32_t mixerStatus = MIXER_IDLE;
+    mixer_state mixerStatus = MIXER_IDLE;
     // find out which tracks need to be processed
     size_t count = activeTracks.size();
     size_t mixedTracks = 0;
@@ -2612,7 +2613,7 @@
 
 bool AudioFlinger::DirectOutputThread::threadLoop()
 {
-    uint32_t mixerStatus = MIXER_IDLE;
+    mixer_state mixerStatus = MIXER_IDLE;
     sp<Track> trackToRemove;
     sp<Track> activeTrack;
     nsecs_t standbyTime = systemTime();
@@ -3009,7 +3010,7 @@
 bool AudioFlinger::DuplicatingThread::threadLoop()
 {
     Vector< sp<Track> > tracksToRemove;
-    uint32_t mixerStatus = MIXER_IDLE;
+    mixer_state mixerStatus = MIXER_IDLE;
     nsecs_t standbyTime = systemTime();
     size_t mixBufferSize = mFrameCount*mFrameSize;
     SortedVector< sp<OutputTrack> > outputTracks;
@@ -3549,7 +3550,7 @@
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
         Mutex::Autolock _l(thread->mLock);
-        int state = mState;
+        track_state state = mState;
         // here the track could be either new, or restarted
         // in both cases "unstop" the track
         if (mState == PAUSED) {
@@ -3590,7 +3591,7 @@
     sp<ThreadBase> thread = mThread.promote();
     if (thread != 0) {
         Mutex::Autolock _l(thread->mLock);
-        int state = mState;
+        track_state state = mState;
         if (mState > STOPPED) {
             mState = STOPPED;
             // If the track is not active (PAUSED and buffers full), flush buffers
@@ -4107,6 +4108,10 @@
     mTrack->destroy();
 }
 
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+    return mTrack->getCblk();
+}
+
 status_t AudioFlinger::TrackHandle::start() {
     return mTrack->start();
 }
@@ -4127,10 +4132,6 @@
     mTrack->pause();
 }
 
-sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
-    return mTrack->getCblk();
-}
-
 status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
 {
     return mTrack->attachAuxEffect(EffectId);
@@ -4237,6 +4238,10 @@
     stop();
 }
 
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+    return mRecordTrack->getCblk();
+}
+
 status_t AudioFlinger::RecordHandle::start() {
     ALOGV("RecordHandle::start()");
     return mRecordTrack->start();
@@ -4247,10 +4252,6 @@
     mRecordTrack->stop();
 }
 
-sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
-    return mRecordTrack->getCblk();
-}
-
 status_t AudioFlinger::RecordHandle::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 262c890..766ba44 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -375,7 +375,7 @@
             void*               mBufferEnd;
             uint32_t            mFrameCount;
             // we don't really need a lock for these
-            int                 mState;
+            track_state         mState;
             int                 mClientTid;
             audio_format_t      mFormat;
             uint32_t            mFlags;
@@ -828,7 +828,7 @@
         virtual     status_t    dumpInternals(int fd, const Vector<String16>& args);
 
     protected:
-                    uint32_t    prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
+                    mixer_state prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
                                                 Vector< sp<Track> > *tracksToRemove);
         virtual     int         getTrackName_l();
         virtual     void        deleteTrackName_l(int name);
@@ -836,8 +836,7 @@
         virtual     uint32_t    suspendSleepTimeUs();
 
                     AudioMixer* mAudioMixer;
-                    uint32_t    mPrevMixerStatus; // previous status (mixer_state) returned by
-                                                  // prepareTracks_l()
+                    mixer_state mPrevMixerStatus; // previous status returned by prepareTracks_l()
     };
 
     class DirectOutputThread : public PlaybackThread {
@@ -908,12 +907,12 @@
     public:
                             TrackHandle(const sp<PlaybackThread::Track>& track);
         virtual             ~TrackHandle();
+        virtual sp<IMemory> getCblk() const;
         virtual status_t    start();
         virtual void        stop();
         virtual void        flush();
         virtual void        mute(bool);
         virtual void        pause();
-        virtual sp<IMemory> getCblk() const;
         virtual status_t    attachAuxEffect(int effectId);
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
@@ -1031,9 +1030,9 @@
     public:
         RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
         virtual             ~RecordHandle();
+        virtual sp<IMemory> getCblk() const;
         virtual status_t    start();
         virtual void        stop();
-        virtual sp<IMemory> getCblk() const;
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
     private:
@@ -1087,7 +1086,7 @@
         void reset_l();
         status_t configure();
         status_t init();
-        uint32_t state() {
+        effect_state state() const {
             return mState;
         }
         uint32_t status() {
@@ -1152,8 +1151,8 @@
         effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
         effect_config_t     mConfig;    // input and output audio configuration
         effect_handle_t  mEffectInterface; // Effect module C API
-        status_t mStatus;               // initialization status
-        uint32_t mState;                // current activation state (effect_state)
+        status_t            mStatus;    // initialization status
+        effect_state        mState;     // current activation state
         Vector< wp<EffectHandle> > mHandles;    // list of client handles
         uint32_t mMaxDisableWaitCnt;    // maximum grace period before forcing an effect off after
                                         // sending disable command.