diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index e796ab3..f1636e6 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -116,6 +116,7 @@
         kWhatStart                   = 'star',
         kWhatRequestIDRFrame         = 'ridr',
         kWhatSetParameters           = 'setP',
+        kWhatSubmitOutputMetaDataBufferIfEOS = 'subm',
     };
 
     enum {
@@ -212,6 +213,7 @@
             OMX_U32 *nMinUndequeuedBuffers);
     status_t allocateOutputMetaDataBuffers();
     status_t submitOutputMetaDataBuffer();
+    void signalSubmitOutputMetaDataBufferIfEOS_workaround();
     status_t allocateOutputBuffersFromNativeWindow();
     status_t cancelBufferToNativeWindow(BufferInfo *info);
     status_t freeOutputBuffersNotOwnedByComponent();
@@ -247,6 +249,8 @@
             int32_t numChannels, int32_t sampleRate, int32_t bitRate,
             int32_t aacProfile, bool isADTS);
 
+    status_t setupAC3Codec(bool encoder, int32_t numChannels, int32_t sampleRate);
+
     status_t selectAudioPortFormat(
             OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat);
 
diff --git a/include/media/stagefright/MediaDefs.h b/include/media/stagefright/MediaDefs.h
index 85693d4..cf5beda 100644
--- a/include/media/stagefright/MediaDefs.h
+++ b/include/media/stagefright/MediaDefs.h
@@ -44,6 +44,7 @@
 extern const char *MEDIA_MIMETYPE_AUDIO_FLAC;
 extern const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS;
 extern const char *MEDIA_MIMETYPE_AUDIO_MSGSM;
+extern const char *MEDIA_MIMETYPE_AUDIO_AC3;
 
 extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
 extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index daaf20f..5121c17 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -248,6 +248,8 @@
             int32_t numChannels, int32_t sampleRate, int32_t bitRate,
             int32_t aacProfile, bool isADTS);
 
+    status_t setAC3Format(int32_t numChannels, int32_t sampleRate);
+
     void setG711Format(int32_t numChannels);
 
     status_t setVideoPortFormatType(
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 76a3358..c68dfc1 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -35,7 +35,9 @@
 
 #include <media/hardware/HardwareAPI.h>
 
+#include <OMX_AudioExt.h>
 #include <OMX_Component.h>
+#include <OMX_IndexExt.h>
 
 #include "include/avc_utils.h"
 
@@ -452,6 +454,18 @@
     (new AMessage(kWhatRequestIDRFrame, id()))->post();
 }
 
+// *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+// Some codecs may return input buffers before having them processed.
+// This causes a halt if we already signaled an EOS on the input
+// port.  For now keep submitting an output buffer if there was an
+// EOS on the input port, but not yet on the output port.
+void ACodec::signalSubmitOutputMetaDataBufferIfEOS_workaround() {
+    if (mPortEOS[kPortIndexInput] && !mPortEOS[kPortIndexOutput] &&
+            mMetaDataBuffersToSubmit > 0) {
+        (new AMessage(kWhatSubmitOutputMetaDataBufferIfEOS, id()))->post();
+    }
+}
+
 status_t ACodec::allocateBuffersOnPort(OMX_U32 portIndex) {
     CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
 
@@ -965,6 +979,10 @@
             "audio_decoder.flac", "audio_encoder.flac" },
         { MEDIA_MIMETYPE_AUDIO_MSGSM,
             "audio_decoder.gsm", "audio_encoder.gsm" },
+        { MEDIA_MIMETYPE_VIDEO_MPEG2,
+            "video_decoder.mpeg2", "video_encoder.mpeg2" },
+        { MEDIA_MIMETYPE_AUDIO_AC3,
+            "audio_decoder.ac3", "audio_encoder.ac3" },
     };
 
     static const size_t kNumMimeToRole =
@@ -1256,6 +1274,15 @@
         } else {
             err = setupRawAudioFormat(kPortIndexInput, sampleRate, numChannels);
         }
+    } else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AC3)) {
+        int32_t numChannels;
+        int32_t sampleRate;
+        if (!msg->findInt32("channel-count", &numChannels)
+                || !msg->findInt32("sample-rate", &sampleRate)) {
+            err = INVALID_OPERATION;
+        } else {
+            err = setupAC3Codec(encoder, numChannels, sampleRate);
+        }
     }
 
     if (err != OK) {
@@ -1452,6 +1479,44 @@
             mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
 }
 
+status_t ACodec::setupAC3Codec(
+        bool encoder, int32_t numChannels, int32_t sampleRate) {
+    status_t err = setupRawAudioFormat(
+            encoder ? kPortIndexInput : kPortIndexOutput, sampleRate, numChannels);
+
+    if (err != OK) {
+        return err;
+    }
+
+    if (encoder) {
+        ALOGW("AC3 encoding is not supported.");
+        return INVALID_OPERATION;
+    }
+
+    OMX_AUDIO_PARAM_ANDROID_AC3TYPE def;
+    InitOMXParams(&def);
+    def.nPortIndex = kPortIndexInput;
+
+    err = mOMX->getParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+
+    if (err != OK) {
+        return err;
+    }
+
+    def.nChannels = numChannels;
+    def.nSampleRate = sampleRate;
+
+    return mOMX->setParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+}
+
 static OMX_AUDIO_AMRBANDMODETYPE pickModeFromBitRate(
         bool isAMRWB, int32_t bps) {
     if (isAMRWB) {
@@ -2546,7 +2611,7 @@
         {
             OMX_AUDIO_PORTDEFINITIONTYPE *audioDef = &def.format.audio;
 
-            switch (audioDef->eEncoding) {
+            switch ((int)audioDef->eEncoding) {
                 case OMX_AUDIO_CodingPCM:
                 {
                     OMX_AUDIO_PARAM_PCMMODETYPE params;
@@ -2652,6 +2717,24 @@
                     break;
                 }
 
+                case OMX_AUDIO_CodingAndroidAC3:
+                {
+                    OMX_AUDIO_PARAM_ANDROID_AC3TYPE params;
+                    InitOMXParams(&params);
+                    params.nPortIndex = kPortIndexOutput;
+
+                    CHECK_EQ((status_t)OK, mOMX->getParameter(
+                            mNode,
+                            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+                            &params,
+                            sizeof(params)));
+
+                    notify->setString("mime", MEDIA_MIMETYPE_AUDIO_AC3);
+                    notify->setInt32("channel-count", params.nChannels);
+                    notify->setInt32("sample-rate", params.nSampleRate);
+                    break;
+                }
+
                 default:
                     TRESPASS();
             }
@@ -4036,6 +4119,9 @@
                 break;
         }
     }
+
+    // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+    mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
 }
 
 void ACodec::ExecutingState::submitRegularOutputBuffers() {
@@ -4184,6 +4270,19 @@
             break;
         }
 
+        // *** NOTE: THE FOLLOWING WORKAROUND WILL BE REMOVED ***
+        case kWhatSubmitOutputMetaDataBufferIfEOS:
+        {
+            if (mCodec->mPortEOS[kPortIndexInput] &&
+                    !mCodec->mPortEOS[kPortIndexOutput]) {
+                status_t err = mCodec->submitOutputMetaDataBuffer();
+                if (err == OK) {
+                    mCodec->signalSubmitOutputMetaDataBufferIfEOS_workaround();
+                }
+            }
+            return true;
+        }
+
         default:
             handled = BaseState::onMessageReceived(msg);
             break;
diff --git a/media/libstagefright/MediaDefs.cpp b/media/libstagefright/MediaDefs.cpp
index b5d4e44..340cba7 100644
--- a/media/libstagefright/MediaDefs.cpp
+++ b/media/libstagefright/MediaDefs.cpp
@@ -42,6 +42,7 @@
 const char *MEDIA_MIMETYPE_AUDIO_FLAC = "audio/flac";
 const char *MEDIA_MIMETYPE_AUDIO_AAC_ADTS = "audio/aac-adts";
 const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm";
+const char *MEDIA_MIMETYPE_AUDIO_AC3 = "audio/ac3";
 
 const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
 const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav";
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 43736ad..625922f 100644
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -40,7 +40,9 @@
 #include <utils/Vector.h>
 
 #include <OMX_Audio.h>
+#include <OMX_AudioExt.h>
 #include <OMX_Component.h>
+#include <OMX_IndexExt.h>
 
 #include "include/avc_utils.h"
 
@@ -528,6 +530,17 @@
                     sampleRate,
                     numChannels);
         }
+    } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_AC3, mMIME)) {
+        int32_t numChannels;
+        int32_t sampleRate;
+        CHECK(meta->findInt32(kKeyChannelCount, &numChannels));
+        CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
+
+        status_t err = setAC3Format(numChannels, sampleRate);
+        if (err != OK) {
+            CODEC_LOGE("setAC3Format() failed (err = %d)", err);
+            return err;
+        }
     } else if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_ALAW, mMIME)
             || !strcasecmp(MEDIA_MIMETYPE_AUDIO_G711_MLAW, mMIME)) {
         // These are PCM-like formats with a fixed sample rate but
@@ -1394,6 +1407,10 @@
             "audio_decoder.flac", "audio_encoder.flac" },
         { MEDIA_MIMETYPE_AUDIO_MSGSM,
             "audio_decoder.gsm", "audio_encoder.gsm" },
+        { MEDIA_MIMETYPE_VIDEO_MPEG2,
+            "video_decoder.mpeg2", "video_encoder.mpeg2" },
+        { MEDIA_MIMETYPE_AUDIO_AC3,
+            "audio_decoder.ac3", "audio_encoder.ac3" },
     };
 
     static const size_t kNumMimeToRole =
@@ -3489,6 +3506,31 @@
     return OK;
 }
 
+status_t OMXCodec::setAC3Format(int32_t numChannels, int32_t sampleRate) {
+    OMX_AUDIO_PARAM_ANDROID_AC3TYPE def;
+    InitOMXParams(&def);
+    def.nPortIndex = kPortIndexInput;
+
+    status_t err = mOMX->getParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+
+    if (err != OK) {
+        return err;
+    }
+
+    def.nChannels = numChannels;
+    def.nSampleRate = sampleRate;
+
+    return mOMX->setParameter(
+            mNode,
+            (OMX_INDEXTYPE)OMX_IndexParamAudioAndroidAc3,
+            &def,
+            sizeof(def));
+}
+
 void OMXCodec::setG711Format(int32_t numChannels) {
     CHECK(!mIsEncoder);
     setRawAudioFormat(kPortIndexInput, 8000, numChannels);
@@ -4422,6 +4464,17 @@
                 mOutputFormat->setInt32(kKeyChannelCount, numChannels);
                 mOutputFormat->setInt32(kKeySampleRate, sampleRate);
                 mOutputFormat->setInt32(kKeyBitRate, bitRate);
+            } else if (audio_def->eEncoding ==
+                    (OMX_AUDIO_CODINGTYPE)OMX_AUDIO_CodingAndroidAC3) {
+                mOutputFormat->setCString(
+                        kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+                int32_t numChannels, sampleRate, bitRate;
+                inputFormat->findInt32(kKeyChannelCount, &numChannels);
+                inputFormat->findInt32(kKeySampleRate, &sampleRate);
+                inputFormat->findInt32(kKeyBitRate, &bitRate);
+                mOutputFormat->setInt32(kKeyChannelCount, numChannels);
+                mOutputFormat->setInt32(kKeySampleRate, sampleRate);
+                mOutputFormat->setInt32(kKeyBitRate, bitRate);
             } else {
                 CHECK(!"Should not be here. Unknown audio encoding.");
             }
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 175a263..cb57a2f 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -506,6 +506,11 @@
                     ElementaryStreamQueue::PCM_AUDIO);
             break;
 
+        case STREAMTYPE_AC3:
+            mQueue = new ElementaryStreamQueue(
+                    ElementaryStreamQueue::AC3);
+            break;
+
         default:
             break;
     }
@@ -614,6 +619,7 @@
         case STREAMTYPE_MPEG2_AUDIO:
         case STREAMTYPE_MPEG2_AUDIO_ADTS:
         case STREAMTYPE_PCM_AUDIO:
+        case STREAMTYPE_AC3:
             return true;
 
         default:
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index a10edc9..d4e30b4 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -88,6 +88,10 @@
         STREAMTYPE_MPEG2_AUDIO_ADTS     = 0x0f,
         STREAMTYPE_MPEG4_VIDEO          = 0x10,
         STREAMTYPE_H264                 = 0x1b,
+
+        // From ATSC A/53 Part 3:2009, 6.7.1
+        STREAMTYPE_AC3                  = 0x81,
+
         STREAMTYPE_PCM_AUDIO            = 0x83,
     };
 
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index e0ff0d1..c0c9717 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -56,6 +56,122 @@
     }
 }
 
+// Parse AC3 header assuming the current ptr is start position of syncframe,
+// update metadata only applicable, and return the payload size
+static unsigned parseAC3SyncFrame(
+        const uint8_t *ptr, size_t size, sp<MetaData> *metaData) {
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 3, 4, 4, 5};
+    static const unsigned samplingRateTable[] = {48000, 44100, 32000};
+    static const unsigned rates[] = {32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256,
+            320, 384, 448, 512, 576, 640};
+
+    static const unsigned frameSizeTable[19][3] = {
+        { 64, 69, 96 },
+        { 80, 87, 120 },
+        { 96, 104, 144 },
+        { 112, 121, 168 },
+        { 128, 139, 192 },
+        { 160, 174, 240 },
+        { 192, 208, 288 },
+        { 224, 243, 336 },
+        { 256, 278, 384 },
+        { 320, 348, 480 },
+        { 384, 417, 576 },
+        { 448, 487, 672 },
+        { 512, 557, 768 },
+        { 640, 696, 960 },
+        { 768, 835, 1152 },
+        { 896, 975, 1344 },
+        { 1024, 1114, 1536 },
+        { 1152, 1253, 1728 },
+        { 1280, 1393, 1920 },
+    };
+
+    ABitReader bits(ptr, size);
+    unsigned syncStartPos = 0;  // in bytes
+    if (bits.numBitsLeft() < 16) {
+        return 0;
+    }
+    if (bits.getBits(16) != 0x0B77) {
+        return 0;
+    }
+
+    if (bits.numBitsLeft() < 16 + 2 + 6 + 5 + 3 + 3) {
+        ALOGV("Not enough bits left for further parsing");
+        return 0;
+    }
+    bits.skipBits(16);  // crc1
+
+    unsigned fscod = bits.getBits(2);
+    if (fscod == 3) {
+        ALOGW("Incorrect fscod in AC3 header");
+        return 0;
+    }
+
+    unsigned frmsizecod = bits.getBits(6);
+    if (frmsizecod > 37) {
+        ALOGW("Incorrect frmsizecod in AC3 header");
+        return 0;
+    }
+
+    unsigned bsid = bits.getBits(5);
+    if (bsid > 8) {
+        ALOGW("Incorrect bsid in AC3 header. Possibly E-AC-3?");
+        return 0;
+    }
+
+    unsigned bsmod = bits.getBits(3);
+    unsigned acmod = bits.getBits(3);
+    unsigned cmixlev = 0;
+    unsigned surmixlev = 0;
+    unsigned dsurmod = 0;
+
+    if ((acmod & 1) > 0 && acmod != 1) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        cmixlev = bits.getBits(2);
+    }
+    if ((acmod & 4) > 0) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        surmixlev = bits.getBits(2);
+    }
+    if (acmod == 2) {
+        if (bits.numBitsLeft() < 2) {
+            return 0;
+        }
+        dsurmod = bits.getBits(2);
+    }
+
+    if (bits.numBitsLeft() < 1) {
+        return 0;
+    }
+    unsigned lfeon = bits.getBits(1);
+
+    unsigned samplingRate = samplingRateTable[fscod];
+    unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod];
+    if (fscod == 1) {
+        payloadSize += frmsizecod & 1;
+    }
+    payloadSize <<= 1;  // convert from 16-bit words to bytes
+
+    unsigned channelCount = channelCountTable[acmod] + lfeon;
+
+    if (metaData != NULL) {
+        (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
+        (*metaData)->setInt32(kKeyChannelCount, channelCount);
+        (*metaData)->setInt32(kKeySampleRate, samplingRate);
+    }
+
+    return payloadSize;
+}
+
+static bool IsSeeminglyValidAC3Header(const uint8_t *ptr, size_t size) {
+    return parseAC3SyncFrame(ptr, size, NULL) > 0;
+}
+
 static bool IsSeeminglyValidADTSHeader(const uint8_t *ptr, size_t size) {
     if (size < 3) {
         // Not enough data to verify header.
@@ -224,6 +340,33 @@
                 break;
             }
 
+            case AC3:
+            {
+                uint8_t *ptr = (uint8_t *)data;
+
+                ssize_t startOffset = -1;
+                for (size_t i = 0; i < size; ++i) {
+                    if (IsSeeminglyValidAC3Header(&ptr[i], size - i)) {
+                        startOffset = i;
+                        break;
+                    }
+                }
+
+                if (startOffset < 0) {
+                    return ERROR_MALFORMED;
+                }
+
+                if (startOffset > 0) {
+                    ALOGI("found something resembling an AC3 syncword at "
+                          "offset %d",
+                          startOffset);
+                }
+
+                data = &ptr[startOffset];
+                size -= startOffset;
+                break;
+            }
+
             case MPEG_AUDIO:
             {
                 uint8_t *ptr = (uint8_t *)data;
@@ -328,6 +471,8 @@
             return dequeueAccessUnitH264();
         case AAC:
             return dequeueAccessUnitAAC();
+        case AC3:
+            return dequeueAccessUnitAC3();
         case MPEG_VIDEO:
             return dequeueAccessUnitMPEGVideo();
         case MPEG4_VIDEO:
@@ -340,6 +485,51 @@
     }
 }
 
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC3() {
+    unsigned syncStartPos = 0;  // in bytes
+    unsigned payloadSize = 0;
+    sp<MetaData> format = new MetaData;
+    while (true) {
+        if (syncStartPos + 2 >= mBuffer->size()) {
+            return NULL;
+        }
+
+        payloadSize = parseAC3SyncFrame(
+                mBuffer->data() + syncStartPos,
+                mBuffer->size() - syncStartPos,
+                &format);
+        if (payloadSize > 0) {
+            break;
+        }
+        ++syncStartPos;
+    }
+
+    if (mBuffer->size() < syncStartPos + payloadSize) {
+        ALOGV("Not enough buffer size for AC3");
+        return NULL;
+    }
+
+    if (mFormat == NULL) {
+        mFormat = format;
+    }
+
+    sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize);
+    memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize);
+
+    int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
+    CHECK_GE(timeUs, 0ll);
+    accessUnit->meta()->setInt64("timeUs", timeUs);
+
+    memmove(
+            mBuffer->data(),
+            mBuffer->data() + syncStartPos + payloadSize,
+            mBuffer->size() - syncStartPos - payloadSize);
+
+    mBuffer->setRange(0, mBuffer->size() - syncStartPos - payloadSize);
+
+    return accessUnit;
+}
+
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() {
     if (mBuffer->size() < 4) {
         return NULL;
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index 66a8087..a2cca77 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -32,6 +32,7 @@
     enum Mode {
         H264,
         AAC,
+        AC3,
         MPEG_AUDIO,
         MPEG_VIDEO,
         MPEG4_VIDEO,
@@ -67,6 +68,7 @@
 
     sp<ABuffer> dequeueAccessUnitH264();
     sp<ABuffer> dequeueAccessUnitAAC();
+    sp<ABuffer> dequeueAccessUnitAC3();
     sp<ABuffer> dequeueAccessUnitMPEGAudio();
     sp<ABuffer> dequeueAccessUnitMPEGVideo();
     sp<ABuffer> dequeueAccessUnitMPEG4Video();