diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
new file mode 100644
index 0000000..c29c7e5
--- /dev/null
+++ b/include/media/AudioTimestamp.h
@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2013 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 ANDROID_AUDIO_TIMESTAMP_H
+#define ANDROID_AUDIO_TIMESTAMP_H
+
+#include <time.h>
+
+class AudioTimestamp {
+public:
+    AudioTimestamp() : mPosition(0) {
+        mTime.tv_sec = 0;
+        mTime.tv_nsec = 0;
+    }
+    // FIXME change type to match android.media.AudioTrack
+    uint32_t        mPosition; // a frame position in AudioTrack::getPosition() units
+    struct timespec mTime;     // corresponding CLOCK_MONOTONIC when frame is expected to present
+};
+
+#endif  // ANDROID_AUDIO_TIMESTAMP_H
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index ae92cdd..453c106 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -19,6 +19,7 @@
 
 #include <cutils/sched_policy.h>
 #include <media/AudioSystem.h>
+#include <media/AudioTimestamp.h>
 #include <media/IAudioTrack.h>
 #include <utils/threads.h>
 
@@ -62,6 +63,9 @@
                                     // voluntary invalidation by mediaserver, or mediaserver crash.
         EVENT_STREAM_END = 7,       // Sent after all the buffers queued in AF and HW are played
                                     // back (after stop is called)
+        EVENT_NEW_TIMESTAMP = 8,    // Delivered periodically and when there's a significant change
+                                    // in the mapping from frame position to presentation time.
+                                    // See AudioTimestamp for the information included with event.
     };
 
     /* Client should declare Buffer on the stack and pass address to obtainBuffer()
@@ -107,6 +111,8 @@
      *          - EVENT_NEW_POS: pointer to const uint32_t containing the new position in frames.
      *          - EVENT_BUFFER_END: unused.
      *          - EVENT_NEW_IAUDIOTRACK: unused.
+     *          - EVENT_STREAM_END: unused.
+     *          - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
      */
 
     typedef void (*callback_t)(int event, void* user, void *info);
@@ -221,6 +227,7 @@
      *  - INVALID_OPERATION: AudioTrack is already initialized
      *  - BAD_VALUE: invalid parameter (channelMask, format, sampleRate...)
      *  - NO_INIT: audio server or audio hardware not initialized
+     * If status is not equal to NO_ERROR, don't call any other APIs on this AudioTrack.
      * If sharedBuffer is non-0, the frameCount parameter is ignored and
      * replaced by the shared buffer's total allocated size in frame units.
      *
@@ -243,7 +250,7 @@
                             transfer_type transferType = TRANSFER_DEFAULT,
                             const audio_offload_info_t *offloadInfo = NULL);
 
-    /* Result of constructing the AudioTrack. This must be checked
+    /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
      * an uninitialized AudioTrack produces undefined results.
      * See set() method above for possible return codes.
@@ -564,6 +571,16 @@
     /* Get parameters */
             String8     getParameters(const String8& keys);
 
+    /* Poll for a timestamp on demand.
+     * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs,
+     * or if you need to get the most recent timestamp outside of the event callback handler.
+     * Caution: calling this method too often may be inefficient;
+     * if you need a high resolution mapping between frame position and presentation time,
+     * consider implementing that at application level, based on the low resolution timestamps.
+     * Returns NO_ERROR if timestamp is valid.
+     */
+            status_t    getTimestamp(AudioTimestamp& timestamp);
+
 protected:
     /* copying audio tracks is not allowed */
                         AudioTrack(const AudioTrack& other);
@@ -630,7 +647,7 @@
             bool     isOffloaded() const
                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
 
-    // may be changed if IAudioTrack is re-created
+    // Next 3 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
     audio_track_cblk_t*     mCblk;                  // re-load after mLock.unlock()
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 1014403..afac4ae 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -26,6 +26,7 @@
 #include <binder/IMemory.h>
 #include <utils/LinearTransform.h>
 #include <utils/String8.h>
+#include <media/AudioTimestamp.h>
 
 namespace android {
 
@@ -86,6 +87,9 @@
 
     /* Send parameters to the audio hardware */
     virtual status_t    setParameters(const String8& keyValuePairs) = 0;
+
+    /* Return NO_ERROR if timestamp is valid */
+    virtual status_t    getTimestamp(AudioTimestamp& timestamp) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 1379379..ad7409d 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -422,6 +422,9 @@
     // Return the total number of frames which AudioFlinger desired but were unavailable,
     // and thus which resulted in an underrun.
     virtual uint32_t    getUnderrunFrames() const { return mCblk->u.mStreaming.mUnderrunFrames; }
+
+    // Return the total number of frames that AudioFlinger has obtained and released
+    virtual size_t      framesReleased() const { return mCblk->mServer; }
 };
 
 class StaticAudioTrackServerProxy : public AudioTrackServerProxy {
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 7aa9e35..176197c 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -148,10 +148,8 @@
             mAudioTrackThread->requestExitAndWait();
             mAudioTrackThread.clear();
         }
-        if (mAudioTrack != 0) {
-            mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
-            mAudioTrack.clear();
-        }
+        mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
+        mAudioTrack.clear();
         IPCThreadState::self()->flushCommands();
         AudioSystem::releaseAudioSessionId(mSessionId);
     }
@@ -222,6 +220,7 @@
 
     AutoMutex lock(mLock);
 
+    // invariant that mAudioTrack != 0 is true only after set() returns successfully
     if (mAudioTrack != 0) {
         ALOGE("Track already in use");
         return INVALID_OPERATION;
@@ -968,6 +967,7 @@
         ALOGE("Could not get control block");
         return NO_INIT;
     }
+    // invariant that mAudioTrack != 0 is true only after set() returns successfully
     if (mAudioTrack != 0) {
         mAudioTrack->asBinder()->unlinkToDeath(mDeathNotifier, this);
         mDeathNotifier.clear();
@@ -1708,11 +1708,13 @@
 status_t AudioTrack::setParameters(const String8& keyValuePairs)
 {
     AutoMutex lock(mLock);
-    if (mAudioTrack != 0) {
-        return mAudioTrack->setParameters(keyValuePairs);
-    } else {
-        return NO_INIT;
-    }
+    return mAudioTrack->setParameters(keyValuePairs);
+}
+
+status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
+{
+    AutoMutex lock(mLock);
+    return mAudioTrack->getTimestamp(timestamp);
 }
 
 String8 AudioTrack::getParameters(const String8& keys)
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index a2b49a3..f0d75ba 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -39,7 +39,8 @@
     ALLOCATE_TIMED_BUFFER,
     QUEUE_TIMED_BUFFER,
     SET_MEDIA_TIME_TRANSFORM,
-    SET_PARAMETERS
+    SET_PARAMETERS,
+    GET_TIMESTAMP,
 };
 
 class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -166,6 +167,21 @@
         }
         return status;
     }
+
+    virtual status_t getTimestamp(AudioTimestamp& timestamp) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        status_t status = remote()->transact(GET_TIMESTAMP, data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+            if (status == NO_ERROR) {
+                timestamp.mPosition = reply.readInt32();
+                timestamp.mTime.tv_sec = reply.readInt32();
+                timestamp.mTime.tv_nsec = reply.readInt32();
+            }
+        }
+        return status;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -241,6 +257,18 @@
             reply->writeInt32(setParameters(keyValuePairs));
             return NO_ERROR;
         } break;
+        case GET_TIMESTAMP: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            AudioTimestamp timestamp;
+            status_t status = getTimestamp(timestamp);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->writeInt32(timestamp.mPosition);
+                reply->writeInt32(timestamp.mTime.tv_sec);
+                reply->writeInt32(timestamp.mTime.tv_nsec);
+            }
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 2418aab..e38e261 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -680,6 +680,14 @@
 
 int64_t AudioPlayer::getRealTimeUs() {
     Mutex::Autolock autoLock(mLock);
+    if (useOffload()) {
+        if (mSeeking) {
+            return mSeekTimeUs;
+        }
+        mPositionTimeRealUs = getOutputPlayPositionUs_l();
+        return mPositionTimeRealUs;
+    }
+
     return getRealTimeUsLocked();
 }
 
@@ -741,11 +749,6 @@
         return 0;
     }
 
-    if (useOffload()) {
-        mPositionTimeRealUs = getOutputPlayPositionUs_l();
-        return mPositionTimeRealUs;
-    }
-
     int64_t realTimeOffset = getRealTimeUsLocked() - mPositionTimeRealUs;
     if (realTimeOffset < 0) {
         realTimeOffset = 0;
@@ -758,8 +761,14 @@
         int64_t *realtime_us, int64_t *mediatime_us) {
     Mutex::Autolock autoLock(mLock);
 
-    *realtime_us = mPositionTimeRealUs;
-    *mediatime_us = mPositionTimeMediaUs;
+    if (useOffload()) {
+        mPositionTimeRealUs = getOutputPlayPositionUs_l();
+        *realtime_us = mPositionTimeRealUs;
+        *mediatime_us = mPositionTimeRealUs;
+    } else {
+        *realtime_us = mPositionTimeRealUs;
+        *mediatime_us = mPositionTimeMediaUs;
+    }
 
     return mPositionTimeRealUs != -1 && mPositionTimeMediaUs != -1;
 }
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
index 1b20cbb..c9b5d26 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.cpp
@@ -58,6 +58,8 @@
       mIsADTS(false),
       mInputBufferCount(0),
       mSignalledError(false),
+      mSawInputEos(false),
+      mSignalledOutputEos(false),
       mAnchorTimeUs(0),
       mNumSamplesOutput(0),
       mOutputPortSettingsChange(NONE) {
@@ -350,115 +352,83 @@
         return;
     }
 
-    while (!inQueue.empty() && !outQueue.empty()) {
-        BufferInfo *inInfo = *inQueue.begin();
-        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+    while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+        BufferInfo *inInfo = NULL;
+        OMX_BUFFERHEADERTYPE *inHeader = NULL;
+        if (!inQueue.empty()) {
+            inInfo = *inQueue.begin();
+            inHeader = inInfo->mHeader;
+        }
 
         BufferInfo *outInfo = *outQueue.begin();
         OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+        outHeader->nFlags = 0;
 
-        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
-            inQueue.erase(inQueue.begin());
-            inInfo->mOwnedByUs = false;
-            notifyEmptyBufferDone(inHeader);
+        if (inHeader) {
+            if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+                mSawInputEos = true;
+            }
 
-            if (mDecoderHasData) {
-                // flush out the decoder's delayed data by calling DecodeFrame
-                // one more time, with the AACDEC_FLUSH flag set
-                INT_PCM *outBuffer =
-                        reinterpret_cast<INT_PCM *>(
-                                outHeader->pBuffer + outHeader->nOffset);
+            if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+                mAnchorTimeUs = inHeader->nTimeStamp;
+                mNumSamplesOutput = 0;
+            }
 
-                AAC_DECODER_ERROR decoderErr =
-                    aacDecoder_DecodeFrame(mAACDecoder,
-                                           outBuffer,
-                                           outHeader->nAllocLen,
-                                           AACDEC_FLUSH);
-                mDecoderHasData = false;
+            if (mIsADTS) {
+                size_t adtsHeaderSize = 0;
+                // skip 30 bits, aac_frame_length follows.
+                // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
 
-                if (decoderErr != AAC_DEC_OK) {
+                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 %ld 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 %ld 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;
+                    }
+                }
+
+                if (signalError) {
                     mSignalledError = true;
 
-                    notify(OMX_EventError, OMX_ErrorUndefined, decoderErr,
+                    notify(OMX_EventError,
+                           OMX_ErrorStreamCorrupt,
+                           ERROR_MALFORMED,
                            NULL);
 
                     return;
                 }
-
-                outHeader->nFilledLen =
-                        mStreamInfo->frameSize
-                            * sizeof(int16_t)
-                            * mStreamInfo->numChannels;
             } else {
-                // we never submitted any data to the decoder, so there's nothing to flush out
-                outHeader->nFilledLen = 0;
-            }
-
-            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
-
-            outQueue.erase(outQueue.begin());
-            outInfo->mOwnedByUs = false;
-            notifyFillBufferDone(outHeader);
-            return;
-        }
-
-        if (inHeader->nOffset == 0) {
-            mAnchorTimeUs = inHeader->nTimeStamp;
-            mNumSamplesOutput = 0;
-        }
-
-        size_t adtsHeaderSize = 0;
-        if (mIsADTS) {
-            // skip 30 bits, aac_frame_length follows.
-            // ssssssss ssssiiip ppffffPc ccohCCll llllllll lll?????
-
-            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 %ld 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 %ld 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;
-                }
-            }
-
-            if (signalError) {
-                mSignalledError = true;
-
-                notify(OMX_EventError,
-                       OMX_ErrorStreamCorrupt,
-                       ERROR_MALFORMED,
-                       NULL);
-
-                return;
+                inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
+                inBufferLength[0] = inHeader->nFilledLen;
             }
         } else {
-            inBuffer[0] = inHeader->pBuffer + inHeader->nOffset;
-            inBufferLength[0] = inHeader->nFilledLen;
+            inBufferLength[0] = 0;
         }
 
         // Fill and decode
@@ -471,50 +441,66 @@
         int prevNumChannels = mStreamInfo->numChannels;
 
         AAC_DECODER_ERROR decoderErr = AAC_DEC_NOT_ENOUGH_BITS;
-        while (bytesValid[0] > 0 && 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);
-            mDecoderHasData = true;
 
             decoderErr = aacDecoder_DecodeFrame(mAACDecoder,
                                                 outBuffer,
                                                 outHeader->nAllocLen,
                                                 0 /* flags */);
-
             if (decoderErr == AAC_DEC_NOT_ENOUGH_BITS) {
-                ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+                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;
+                    }
+                    outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+                    mSignalledOutputEos = true;
+                    break;
+                } else {
+                    ALOGW("Not enough bits, bytesValid %d", bytesValid[0]);
+                }
             }
         }
 
         size_t numOutBytes =
             mStreamInfo->frameSize * sizeof(int16_t) * mStreamInfo->numChannels;
 
-        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 (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);
 
-            memset(outHeader->pBuffer + outHeader->nOffset, 0, numOutBytes);
+                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
+            }
 
-        if (inHeader->nFilledLen == 0) {
-            inInfo->mOwnedByUs = false;
-            inQueue.erase(inQueue.begin());
-            inInfo = NULL;
-            notifyEmptyBufferDone(inHeader);
-            inHeader = NULL;
+            if (inHeader->nFilledLen == 0) {
+                inInfo->mOwnedByUs = false;
+                inQueue.erase(inQueue.begin());
+                inInfo = NULL;
+                notifyEmptyBufferDone(inHeader);
+                inHeader = NULL;
+            }
         }
 
         /*
@@ -555,7 +541,6 @@
             // we've previously decoded valid data, in the latter case
             // (decode failed) we'll output a silent frame.
             outHeader->nFilledLen = numOutBytes;
-            outHeader->nFlags = 0;
 
             outHeader->nTimeStamp =
                 mAnchorTimeUs
@@ -606,6 +591,8 @@
     mStreamInfo->sampleRate = 0;
 
     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 2d960ab..a7ea1e2 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC2.h
+++ b/media/libstagefright/codecs/aacdec/SoftAAC2.h
@@ -55,6 +55,8 @@
     bool mDecoderHasData;
     size_t mInputBufferCount;
     bool mSignalledError;
+    bool mSawInputEos;
+    bool mSignalledOutputEos;
     int64_t mAnchorTimeUs;
     int64_t mNumSamplesOutput;
 
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
index 7c382fb..877e3cb 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.cpp
@@ -49,6 +49,8 @@
       mNumChannels(2),
       mSamplingRate(44100),
       mSignalledError(false),
+      mSawInputEos(false),
+      mSignalledOutputEos(false),
       mOutputPortSettingsChange(NONE) {
     initPorts();
     initDecoder();
@@ -194,48 +196,36 @@
     List<BufferInfo *> &inQueue = getPortQueue(0);
     List<BufferInfo *> &outQueue = getPortQueue(1);
 
-    while (!inQueue.empty() && !outQueue.empty()) {
-        BufferInfo *inInfo = *inQueue.begin();
-        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+    while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+        BufferInfo *inInfo = NULL;
+        OMX_BUFFERHEADERTYPE *inHeader = NULL;
+        if (!inQueue.empty()) {
+            inInfo = *inQueue.begin();
+            inHeader = inInfo->mHeader;
+        }
 
         BufferInfo *outInfo = *outQueue.begin();
         OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+        outHeader->nFlags = 0;
 
-        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
-            inQueue.erase(inQueue.begin());
-            inInfo->mOwnedByUs = false;
-            notifyEmptyBufferDone(inHeader);
-
-            if (!mIsFirst) {
-                // pad the end of the stream with 529 samples, since that many samples
-                // were trimmed off the beginning when decoding started
-                outHeader->nFilledLen =
-                    kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
-
-                memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
-            } else {
-                // Since we never discarded frames from the start, we won't have
-                // to add any padding at the end either.
-                outHeader->nFilledLen = 0;
+        if (inHeader) {
+            if (inHeader->nOffset == 0 && inHeader->nFilledLen) {
+                mAnchorTimeUs = inHeader->nTimeStamp;
+                mNumFramesOutput = 0;
             }
 
-            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+            if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+                mSawInputEos = true;
+            }
 
-            outQueue.erase(outQueue.begin());
-            outInfo->mOwnedByUs = false;
-            notifyFillBufferDone(outHeader);
-            return;
+            mConfig->pInputBuffer =
+                inHeader->pBuffer + inHeader->nOffset;
+
+            mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
+        } else {
+            mConfig->pInputBuffer = NULL;
+            mConfig->inputBufferCurrentLength = 0;
         }
-
-        if (inHeader->nOffset == 0) {
-            mAnchorTimeUs = inHeader->nTimeStamp;
-            mNumFramesOutput = 0;
-        }
-
-        mConfig->pInputBuffer =
-            inHeader->pBuffer + inHeader->nOffset;
-
-        mConfig->inputBufferCurrentLength = inHeader->nFilledLen;
         mConfig->inputBufferMaxLength = 0;
         mConfig->inputBufferUsedLength = 0;
 
@@ -262,13 +252,28 @@
                 mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t);
             }
 
-            // This is recoverable, just ignore the current frame and
-            // play silence instead.
-            memset(outHeader->pBuffer,
-                   0,
-                   mConfig->outputFrameSize * sizeof(int16_t));
+            if (decoderErr == NO_ENOUGH_MAIN_DATA_ERROR && mSawInputEos) {
+                if (!mIsFirst) {
+                    // pad the end of the stream with 529 samples, since that many samples
+                    // were trimmed off the beginning when decoding started
+                    outHeader->nOffset = 0;
+                    outHeader->nFilledLen = kPVMP3DecoderDelay * mNumChannels * sizeof(int16_t);
 
-            mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+                    memset(outHeader->pBuffer, 0, outHeader->nFilledLen);
+                }
+                outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+                mSignalledOutputEos = true;
+            } else {
+                // This is recoverable, just ignore the current frame and
+                // play silence instead.
+                memset(outHeader->pBuffer,
+                       0,
+                       mConfig->outputFrameSize * sizeof(int16_t));
+
+                if (inHeader) {
+                    mConfig->inputBufferUsedLength = inHeader->nFilledLen;
+                }
+            }
         } else if (mConfig->samplingRate != mSamplingRate
                 || mConfig->num_channels != mNumChannels) {
             mSamplingRate = mConfig->samplingRate;
@@ -289,7 +294,7 @@
 
             outHeader->nFilledLen =
                 mConfig->outputFrameSize * sizeof(int16_t) - outHeader->nOffset;
-        } else {
+        } else if (!mSignalledOutputEos) {
             outHeader->nOffset = 0;
             outHeader->nFilledLen = mConfig->outputFrameSize * sizeof(int16_t);
         }
@@ -298,23 +303,24 @@
             mAnchorTimeUs
                 + (mNumFramesOutput * 1000000ll) / mConfig->samplingRate;
 
-        outHeader->nFlags = 0;
+        if (inHeader) {
+            CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
 
-        CHECK_GE(inHeader->nFilledLen, mConfig->inputBufferUsedLength);
+            inHeader->nOffset += mConfig->inputBufferUsedLength;
+            inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
 
-        inHeader->nOffset += mConfig->inputBufferUsedLength;
-        inHeader->nFilledLen -= mConfig->inputBufferUsedLength;
+
+            if (inHeader->nFilledLen == 0) {
+                inInfo->mOwnedByUs = false;
+                inQueue.erase(inQueue.begin());
+                inInfo = NULL;
+                notifyEmptyBufferDone(inHeader);
+                inHeader = NULL;
+            }
+        }
 
         mNumFramesOutput += mConfig->outputFrameSize / mNumChannels;
 
-        if (inHeader->nFilledLen == 0) {
-            inInfo->mOwnedByUs = false;
-            inQueue.erase(inQueue.begin());
-            inInfo = NULL;
-            notifyEmptyBufferDone(inHeader);
-            inHeader = NULL;
-        }
-
         outInfo->mOwnedByUs = false;
         outQueue.erase(outQueue.begin());
         outInfo = NULL;
@@ -362,6 +368,8 @@
     pvmp3_InitDecoder(mConfig, mDecoderBuf);
     mIsFirst = true;
     mSignalledError = false;
+    mSawInputEos = false;
+    mSignalledOutputEos = false;
     mOutputPortSettingsChange = NONE;
 }
 
diff --git a/media/libstagefright/codecs/mp3dec/SoftMP3.h b/media/libstagefright/codecs/mp3dec/SoftMP3.h
index 4af91ea..f9e7b53 100644
--- a/media/libstagefright/codecs/mp3dec/SoftMP3.h
+++ b/media/libstagefright/codecs/mp3dec/SoftMP3.h
@@ -61,6 +61,8 @@
 
     bool mIsFirst;
     bool mSignalledError;
+    bool mSawInputEos;
+    bool mSignalledOutputEos;
 
     enum {
         NONE,
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 51bb958..a377b23 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -54,6 +54,8 @@
       mAnchorTimeUs(0),
       mNumFramesOutput(0),
       mNumFramesLeftOnPage(-1),
+      mSawInputEos(false),
+      mSignalledOutputEos(false),
       mOutputPortSettingsChange(NONE) {
     initPorts();
     CHECK_EQ(initDecoder(), (status_t)OK);
@@ -290,48 +292,47 @@
         return;
     }
 
-    while (!inQueue.empty() && !outQueue.empty()) {
-        BufferInfo *inInfo = *inQueue.begin();
-        OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+    while ((!inQueue.empty() || (mSawInputEos && !mSignalledOutputEos)) && !outQueue.empty()) {
+        BufferInfo *inInfo = NULL;
+        OMX_BUFFERHEADERTYPE *inHeader = NULL;
+        if (!inQueue.empty()) {
+            inInfo = *inQueue.begin();
+            inHeader = inInfo->mHeader;
+        }
 
         BufferInfo *outInfo = *outQueue.begin();
         OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
 
-        if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
-            inQueue.erase(inQueue.begin());
-            inInfo->mOwnedByUs = false;
-            notifyEmptyBufferDone(inHeader);
+        int32_t numPageSamples = 0;
 
-            outHeader->nFilledLen = 0;
-            outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+        if (inHeader) {
+            if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+                mSawInputEos = true;
+            }
 
-            outQueue.erase(outQueue.begin());
-            outInfo->mOwnedByUs = false;
-            notifyFillBufferDone(outHeader);
-            return;
+            if (inHeader->nFilledLen || !mSawInputEos) {
+                CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
+                memcpy(&numPageSamples,
+                       inHeader->pBuffer
+                        + inHeader->nOffset + inHeader->nFilledLen - 4,
+                       sizeof(numPageSamples));
+
+                if (inHeader->nOffset == 0) {
+                    mAnchorTimeUs = inHeader->nTimeStamp;
+                    mNumFramesOutput = 0;
+                }
+
+                inHeader->nFilledLen -= sizeof(numPageSamples);;
+            }
         }
 
-        int32_t numPageSamples;
-        CHECK_GE(inHeader->nFilledLen, sizeof(numPageSamples));
-        memcpy(&numPageSamples,
-               inHeader->pBuffer
-                + inHeader->nOffset + inHeader->nFilledLen - 4,
-               sizeof(numPageSamples));
-
         if (numPageSamples >= 0) {
             mNumFramesLeftOnPage = numPageSamples;
         }
 
-        if (inHeader->nOffset == 0) {
-            mAnchorTimeUs = inHeader->nTimeStamp;
-            mNumFramesOutput = 0;
-        }
-
-        inHeader->nFilledLen -= sizeof(numPageSamples);;
-
         ogg_buffer buf;
-        buf.data = inHeader->pBuffer + inHeader->nOffset;
-        buf.size = inHeader->nFilledLen;
+        buf.data = inHeader ? inHeader->pBuffer + inHeader->nOffset : NULL;
+        buf.size = inHeader ? inHeader->nFilledLen : 0;
         buf.refcount = 1;
         buf.ptr.owner = NULL;
 
@@ -384,11 +385,13 @@
 
         mNumFramesOutput += numFrames;
 
-        inInfo->mOwnedByUs = false;
-        inQueue.erase(inQueue.begin());
-        inInfo = NULL;
-        notifyEmptyBufferDone(inHeader);
-        inHeader = NULL;
+        if (inHeader) {
+            inInfo->mOwnedByUs = false;
+            inQueue.erase(inQueue.begin());
+            inInfo = NULL;
+            notifyEmptyBufferDone(inHeader);
+            inHeader = NULL;
+        }
 
         outInfo->mOwnedByUs = false;
         outQueue.erase(outQueue.begin());
@@ -425,6 +428,8 @@
         mVi = NULL;
     }
 
+    mSawInputEos = false;
+    mSignalledOutputEos = false;
     mOutputPortSettingsChange = NONE;
 }
 
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
index cb628a0..1d00816 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.h
@@ -59,6 +59,8 @@
     int64_t mAnchorTimeUs;
     int64_t mNumFramesOutput;
     int32_t mNumFramesLeftOnPage;
+    bool mSawInputEos;
+    bool mSignalledOutputEos;
 
     enum {
         NONE,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index e5e4113..5df04f4 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -411,6 +411,7 @@
         virtual status_t    setMediaTimeTransform(const LinearTransform& xform,
                                                   int target);
         virtual status_t    setParameters(const String8& keyValuePairs);
+        virtual status_t    getTimestamp(AudioTimestamp& timestamp);
 
         virtual status_t onTransact(
             uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 5600411c..d34833f 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -59,6 +59,7 @@
             void        setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
             int16_t     *mainBuffer() const { return mMainBuffer; }
             int         auxEffectId() const { return mAuxEffectId; }
+    virtual status_t    getTimestamp(AudioTimestamp& timestamp);
 
 // implement FastMixerState::VolumeProvider interface
     virtual uint32_t    getVolumeLR();
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2c2931f..bc01ede 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -943,7 +943,9 @@
         mDraining(false),
         mScreenState(AudioFlinger::mScreenState),
         // index 0 is reserved for normal mixer's submix
-        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1)
+        mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1),
+        // mLatchD, mLatchQ,
+        mLatchDValid(false), mLatchQValid(false)
 {
     snprintf(mName, kNameLength, "AudioOut_%X", id);
     mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName);
@@ -1818,6 +1820,14 @@
         } else {
             bytesWritten = framesWritten;
         }
+        status_t status = INVALID_OPERATION;    // mLatchD.mTimestamp is invalid
+        if (status == NO_ERROR) {
+            size_t totalFramesWritten = mNormalSink->framesWritten();
+            if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) {
+                mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition;
+                mLatchDValid = true;
+            }
+        }
     // otherwise use the HAL / AudioStreamOut directly
     } else {
         // Direct output and offload threads
@@ -2096,6 +2106,12 @@
                 logString = NULL;
             }
 
+            if (mLatchDValid) {
+                mLatchQ = mLatchD;
+                mLatchDValid = false;
+                mLatchQValid = true;
+            }
+
             if (checkForNewParameters_l()) {
                 cacheParameters_l();
             }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 31d5323..1333de2 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -606,6 +606,17 @@
                 // accessed by both binder threads and within threadLoop(), lock on mutex needed
                 unsigned    mFastTrackAvailMask;    // bit i set if fast track [i] is available
     virtual     void        flushOutput_l();
+
+private:
+    // timestamp latch:
+    //  D input is written by threadLoop_write while mutex is unlocked, and read while locked
+    //  Q output is written while locked, and read while locked
+    struct {
+        AudioTimestamp  mTimestamp;
+        uint32_t        mUnpresentedFrames;
+    } mLatchD, mLatchQ;
+    bool mLatchDValid;  // true means mLatchD is valid, and clock it into latch at next opportunity
+    bool mLatchQValid;  // true means mLatchQ is valid
 };
 
 class MixerThread : public PlaybackThread {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e676365..9622709 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -283,6 +283,11 @@
     return mTrack->setParameters(keyValuePairs);
 }
 
+status_t AudioFlinger::TrackHandle::getTimestamp(AudioTimestamp& timestamp)
+{
+    return mTrack->getTimestamp(timestamp);
+}
+
 status_t AudioFlinger::TrackHandle::onTransact(
     uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
 {
@@ -711,6 +716,29 @@
     }
 }
 
+status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
+{
+    sp<ThreadBase> thread = mThread.promote();
+    if (thread == 0) {
+        return false;
+    }
+    Mutex::Autolock _l(thread->mLock);
+    PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+    if (!playbackThread->mLatchQValid) {
+        return INVALID_OPERATION;
+    }
+    uint32_t unpresentedFrames =
+            ((int64_t) playbackThread->mLatchQ.mUnpresentedFrames * mSampleRate) /
+            playbackThread->mSampleRate;
+    uint32_t framesWritten = mAudioTrackServerProxy->framesReleased();
+    if (framesWritten < unpresentedFrames) {
+        return INVALID_OPERATION;
+    }
+    timestamp.mPosition = framesWritten - unpresentedFrames;
+    timestamp.mTime = playbackThread->mLatchQ.mTimestamp.mTime;
+    return NO_ERROR;
+}
+
 status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
 {
     status_t status = DEAD_OBJECT;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 7f2ec7a..47321e0 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -1424,6 +1424,8 @@
     Mutex::Autolock l(mRequestLock);
     mRequestQueue.push_back(request);
 
+    unpauseForNewRequests();
+
     return OK;
 }
 
@@ -1489,6 +1491,9 @@
     mRepeatingRequests.clear();
     mRepeatingRequests.insert(mRepeatingRequests.begin(),
             requests.begin(), requests.end());
+
+    unpauseForNewRequests();
+
     return OK;
 }
 
@@ -1791,7 +1796,9 @@
         mRequestQueue.erase(firstRequest);
     }
 
-    // Not paused
+    // In case we've been unpaused by setPaused clearing mDoPause, need to
+    // update internal pause state (capture/setRepeatingRequest unpause
+    // directly).
     Mutex::Autolock pl(mPauseLock);
     mPaused = false;
 
@@ -1824,6 +1831,16 @@
     return false;
 }
 
+void Camera3Device::RequestThread::unpauseForNewRequests() {
+    // With work to do, mark thread as unpaused.
+    // If paused by request (setPaused), don't resume, to avoid
+    // extra signaling/waiting overhead to waitUntilPaused
+    Mutex::Autolock p(mPauseLock);
+    if (!mDoPause) {
+        mPaused = false;
+    }
+}
+
 void Camera3Device::RequestThread::setErrorState(const char *fmt, ...) {
     sp<Camera3Device> parent = mParent.promote();
     if (parent != NULL) {
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 99e1cc8..6565048 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -314,6 +314,7 @@
 
         // Pause handling
         bool               waitIfPaused();
+        void               unpauseForNewRequests();
 
         // Relay error to parent device object setErrorState
         void               setErrorState(const char *fmt, ...);