diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 62f0c64..b426798 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -60,7 +60,7 @@
         size_t      frameCount;     // number of sample frames corresponding to size;
                                     // on input it is the number of frames available,
                                     // on output is the number of frames actually drained
-                                    // (currently ignored, but will make the primary field in future)
+                                    // (currently ignored but will make the primary field in future)
 
         size_t      size;           // input/output in bytes == frameCount * frameSize
                                     // FIXME this is redundant with respect to frameCount,
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 006af08..77f7d9a 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -155,7 +155,8 @@
     class OutputDescriptor {
     public:
         OutputDescriptor()
-        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)  {}
+        : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0)
+            {}
 
         uint32_t samplingRate;
         audio_format_t format;
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index be818c6..22ad453 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -132,7 +132,7 @@
             lStatus = reply.readInt32();
             track = interface_cast<IAudioTrack>(reply.readStrongBinder());
         }
-        if (status) {
+        if (status != NULL) {
             *status = lStatus;
         }
         return track;
@@ -180,7 +180,7 @@
             lStatus = reply.readInt32();
             record = interface_cast<IAudioRecord>(reply.readStrongBinder());
         }
-        if (status) {
+        if (status != NULL) {
             *status = lStatus;
         }
         return record;
@@ -397,15 +397,25 @@
         audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
         ALOGV("openOutput() returned output, %d", output);
         devices = (audio_devices_t)reply.readInt32();
-        if (pDevices != NULL) *pDevices = devices;
+        if (pDevices != NULL) {
+            *pDevices = devices;
+        }
         samplingRate = reply.readInt32();
-        if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+        if (pSamplingRate != NULL) {
+            *pSamplingRate = samplingRate;
+        }
         format = (audio_format_t) reply.readInt32();
-        if (pFormat != NULL) *pFormat = format;
+        if (pFormat != NULL) {
+            *pFormat = format;
+        }
         channelMask = (audio_channel_mask_t)reply.readInt32();
-        if (pChannelMask != NULL) *pChannelMask = channelMask;
+        if (pChannelMask != NULL) {
+            *pChannelMask = channelMask;
+        }
         latency = reply.readInt32();
-        if (pLatencyMs != NULL) *pLatencyMs = latency;
+        if (pLatencyMs != NULL) {
+            *pLatencyMs = latency;
+        }
         return output;
     }
 
@@ -469,13 +479,21 @@
         remote()->transact(OPEN_INPUT, data, &reply);
         audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
         devices = (audio_devices_t)reply.readInt32();
-        if (pDevices != NULL) *pDevices = devices;
+        if (pDevices != NULL) {
+            *pDevices = devices;
+        }
         samplingRate = reply.readInt32();
-        if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+        if (pSamplingRate != NULL) {
+            *pSamplingRate = samplingRate;
+        }
         format = (audio_format_t) reply.readInt32();
-        if (pFormat != NULL) *pFormat = format;
+        if (pFormat != NULL) {
+            *pFormat = format;
+        }
         channelMask = (audio_channel_mask_t)reply.readInt32();
-        if (pChannelMask != NULL) *pChannelMask = channelMask;
+        if (pChannelMask != NULL) {
+            *pChannelMask = channelMask;
+        }
         return input;
     }
 
@@ -517,11 +535,11 @@
         status_t status = reply.readInt32();
         if (status == NO_ERROR) {
             uint32_t tmp = reply.readInt32();
-            if (halFrames) {
+            if (halFrames != NULL) {
                 *halFrames = tmp;
             }
             tmp = reply.readInt32();
-            if (dspFrames) {
+            if (dspFrames != NULL) {
                 *dspFrames = tmp;
             }
         }
@@ -639,7 +657,7 @@
 
         if (pDesc == NULL) {
             return effect;
-            if (status) {
+            if (status != NULL) {
                 *status = BAD_VALUE;
             }
         }
@@ -657,7 +675,7 @@
         } else {
             lStatus = reply.readInt32();
             int tmp = reply.readInt32();
-            if (id) {
+            if (id != NULL) {
                 *id = tmp;
             }
             tmp = reply.readInt32();
@@ -667,7 +685,7 @@
             effect = interface_cast<IEffect>(reply.readStrongBinder());
             reply.read(pDesc, sizeof(effect_descriptor_t));
         }
-        if (status) {
+        if (status != NULL) {
             *status = lStatus;
         }
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3d65c44..1bac09e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -537,9 +537,7 @@
     }
 
 Exit:
-    if (status != NULL) {
-        *status = lStatus;
-    }
+    *status = lStatus;
     return trackHandle;
 }
 
@@ -1273,12 +1271,9 @@
 
     // return to handle to client
     recordHandle = new RecordHandle(recordTrack);
-    lStatus = NO_ERROR;
 
 Exit:
-    if (status) {
-        *status = lStatus;
-    }
+    *status = lStatus;
     return recordHandle;
 }
 
@@ -1421,10 +1416,11 @@
 {
     PlaybackThread *thread = NULL;
     struct audio_config config;
+    memset(&config, 0, sizeof(config));
     config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
     config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
     config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
-    if (offloadInfo) {
+    if (offloadInfo != NULL) {
         config.offload_info = *offloadInfo;
     }
 
@@ -1645,13 +1641,14 @@
     status_t status;
     RecordThread *thread = NULL;
     struct audio_config config;
+    memset(&config, 0, sizeof(config));
     config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0;
     config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0;
     config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT;
 
     uint32_t reqSamplingRate = config.sample_rate;
     audio_format_t reqFormat = config.format;
-    audio_channel_mask_t reqChannels = config.channel_mask;
+    audio_channel_mask_t reqChannelMask = config.channel_mask;
     audio_stream_in_t *inStream = NULL;
     AudioHwDevice *inHwDev;
 
@@ -1684,7 +1681,7 @@
     if (status == BAD_VALUE &&
         reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
         (config.sample_rate <= 2 * reqSamplingRate) &&
-        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) {
+        (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannelMask) <= FCC_2)) {
         ALOGV("openInput() reopening with proposed sampling rate and channel mask");
         inStream = NULL;
         status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream);
@@ -1749,7 +1746,7 @@
         thread = new RecordThread(this,
                                   input,
                                   reqSamplingRate,
-                                  reqChannels,
+                                  reqChannelMask,
                                   id,
                                   primaryOutputDevice_l(),
                                   *pDevices
@@ -1766,7 +1763,7 @@
             *pFormat = config.format;
         }
         if (pChannelMask != NULL) {
-            *pChannelMask = reqChannels;
+            *pChannelMask = reqChannelMask;
         }
 
         // notify client processes of the new input creation
@@ -2209,9 +2206,7 @@
     }
 
 Exit:
-    if (status != NULL) {
-        *status = lStatus;
-    }
+    *status = lStatus;
     return handle;
 }
 
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index e5e4113..50c20b8 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -109,7 +109,7 @@
                                 pid_t tid,
                                 int *sessionId,
                                 String8& name,
-                                status_t *status);
+                                status_t *status /*non-NULL*/);
 
     virtual sp<IAudioRecord> openRecord(
                                 audio_io_handle_t input,
@@ -120,7 +120,7 @@
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
                                 int *sessionId,
-                                status_t *status);
+                                status_t *status /*non-NULL*/);
 
     virtual     uint32_t    sampleRate(audio_io_handle_t output) const;
     virtual     int         channelCount(audio_io_handle_t output) const;
@@ -209,7 +209,7 @@
                         int32_t priority,
                         audio_io_handle_t io,
                         int sessionId,
-                        status_t *status,
+                        status_t *status /*non-NULL*/,
                         int *id,
                         int *enabled);
 
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 5600411c..43fb946 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -33,6 +33,7 @@
                                 int sessionId,
                                 IAudioFlinger::track_flags_t flags);
     virtual             ~Track();
+    virtual status_t    initCheck() const;
 
     static  void        appendDumpHeader(String8& result);
             void        dump(char* buffer, size_t size);
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index ffe3e9f..6c0d1d3 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -57,5 +57,4 @@
     // releaseBuffer() not overridden
 
     bool                mOverflow;  // overflow on most recent attempt to fill client buffer
-    AudioRecordServerProxy* mAudioRecordServerProxy;
 };
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 2c2931f..db3b82b 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -266,8 +266,8 @@
     :   Thread(false /*canCallJava*/),
         mType(type),
         mAudioFlinger(audioFlinger),
-        // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are
-        // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
+        // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
+        // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
         mParamStatus(NO_ERROR),
         mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
         mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
@@ -293,6 +293,17 @@
     }
 }
 
+status_t AudioFlinger::ThreadBase::readyToRun()
+{
+    status_t status = initCheck();
+    if (status == NO_ERROR) {
+        ALOGI("AudioFlinger's thread %p ready to run", this);
+    } else {
+        ALOGE("No working audio driver found.");
+    }
+    return status;
+}
+
 void AudioFlinger::ThreadBase::exit()
 {
     ALOGV("ThreadBase::exit");
@@ -423,6 +434,8 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount);
     result.append(buffer);
+    snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize);
+    result.append(buffer);
     snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount);
     result.append(buffer);
     snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask);
@@ -682,8 +695,7 @@
         int sessionId,
         effect_descriptor_t *desc,
         int *enabled,
-        status_t *status
-        )
+        status_t *status)
 {
     sp<EffectModule> effect;
     sp<EffectHandle> handle;
@@ -783,9 +795,7 @@
         handle.clear();
     }
 
-    if (status != NULL) {
-        *status = lStatus;
-    }
+    *status = lStatus;
     return handle;
 }
 
@@ -929,7 +939,7 @@
                                              type_t type)
     :   ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
         mNormalFrameCount(0), mMixBuffer(NULL),
-        mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+        mSuspended(0), mBytesWritten(0),
         // mStreamTypes[] initialized in constructor body
         mOutput(output),
         mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
@@ -983,7 +993,7 @@
 AudioFlinger::PlaybackThread::~PlaybackThread()
 {
     mAudioFlinger->unregisterWriter(mNBLogWriter);
-    delete [] mAllocMixBuffer;
+    delete[] mMixBuffer;
 }
 
 void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
@@ -1073,16 +1083,6 @@
 }
 
 // Thread virtuals
-status_t AudioFlinger::PlaybackThread::readyToRun()
-{
-    status_t status = initCheck();
-    if (status == NO_ERROR) {
-        ALOGI("AudioFlinger's thread %p ready to run", this);
-    } else {
-        ALOGE("No working audio driver found.");
-    }
-    return status;
-}
 
 void AudioFlinger::PlaybackThread::onFirstRef()
 {
@@ -1248,8 +1248,12 @@
             track = TimedTrack::create(this, client, streamType, sampleRate, format,
                     channelMask, frameCount, sharedBuffer, sessionId);
         }
-        if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
-            lStatus = NO_MEMORY;
+
+        // new Track always returns non-NULL,
+        // but TimedTrack::create() is a factory that could fail by returning NULL
+        lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
+        if (lStatus != NO_ERROR) {
+            track.clear();
             goto Exit;
         }
 
@@ -1274,9 +1278,7 @@
     lStatus = NO_ERROR;
 
 Exit:
-    if (status) {
-        *status = lStatus;
-    }
+    *status = lStatus;
     return track;
 }
 
@@ -1559,7 +1561,8 @@
                 mFormat);
     }
     mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
-    mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
+    mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common);
+    mFrameCount = mBufferSize / mFrameSize;
     if (mFrameCount & 15) {
         ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames",
                 mFrameCount);
@@ -1615,11 +1618,11 @@
     ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
             mNormalFrameCount);
 
-    delete[] mAllocMixBuffer;
-    size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize;
-    mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1];
-    mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align);
-    memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize);
+    delete[] mMixBuffer;
+    size_t normalBufferSize = mNormalFrameCount * mFrameSize;
+    // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1)
+    mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1];
+    memset(mMixBuffer, 0, normalBufferSize);
 
     // force reconfiguration of effect chains and engines to take new buffer size and audio
     // parameters into account
@@ -4159,7 +4162,7 @@
                                          ) :
     ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD),
     mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
-    // mRsmpInIndex and mBufferSize set by readInputParameters()
+    // mRsmpInIndex set by readInputParameters()
     mReqChannelCount(popcount(channelMask)),
     mReqSampleRate(sampleRate)
     // mBytesRead is only meaningful while active, and so is cleared in start()
@@ -4187,13 +4190,6 @@
     run(mName, PRIORITY_URGENT_AUDIO);
 }
 
-status_t AudioFlinger::RecordThread::readyToRun()
-{
-    status_t status = initCheck();
-    ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this);
-    return status;
-}
-
 bool AudioFlinger::RecordThread::threadLoop()
 {
     AudioBufferProvider::Buffer buffer;
@@ -4505,8 +4501,9 @@
         track = new RecordTrack(this, client, sampleRate,
                       format, channelMask, frameCount, sessionId);
 
-        if (track->getCblk() == 0) {
-            lStatus = NO_MEMORY;
+        lStatus = track->initCheck();
+        if (lStatus != NO_ERROR) {
+            track.clear();
             goto Exit;
         }
         mTracks.add(track);
@@ -4527,9 +4524,7 @@
     lStatus = NO_ERROR;
 
 Exit:
-    if (status) {
-        *status = lStatus;
-    }
+    *status = lStatus;
     return track;
 }
 
@@ -4827,7 +4822,7 @@
         int value;
         audio_format_t reqFormat = mFormat;
         uint32_t reqSamplingRate = mReqSampleRate;
-        uint32_t reqChannelCount = mReqChannelCount;
+        audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount);
 
         if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
             reqSamplingRate = value;
@@ -4842,8 +4837,13 @@
             }
         }
         if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
-            reqChannelCount = popcount(value);
-            reconfig = true;
+            audio_channel_mask_t mask = (audio_channel_mask_t) value;
+            if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
+                status = BAD_VALUE;
+            } else {
+                reqChannelMask = mask;
+                reconfig = true;
+            }
         }
         if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
             // do not accept frame count changes if tracks are open as the track buffer
@@ -4908,7 +4908,8 @@
                             <= (2 * reqSamplingRate)) &&
                     popcount(mInput->stream->common.get_channels(&mInput->stream->common))
                             <= FCC_2 &&
-                    (reqChannelCount <= FCC_2)) {
+                    (reqChannelMask == AUDIO_CHANNEL_IN_MONO ||
+                            reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) {
                     status = NO_ERROR;
                 }
                 if (status == NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 31d5323..fce3245 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -36,6 +36,8 @@
                 audio_devices_t outDevice, audio_devices_t inDevice, type_t type);
     virtual             ~ThreadBase();
 
+    virtual status_t    readyToRun();
+
     void dumpBase(int fd, const Vector<String16>& args);
     void dumpEffectChains(int fd, const Vector<String16>& args);
 
@@ -156,7 +158,7 @@
                                     int sessionId,
                                     effect_descriptor_t *desc,
                                     int *enabled,
-                                    status_t *status);
+                                    status_t *status /*non-NULL*/);
                 void disconnectEffect(const sp< EffectModule>& effect,
                                       EffectHandle *handle,
                                       bool unpinIfLast);
@@ -270,6 +272,7 @@
                 uint32_t                mChannelCount;
                 size_t                  mFrameSize;
                 audio_format_t          mFormat;
+                size_t                  mBufferSize;       // HAL buffer size for read() or write()
 
                 // Parameter sequence by client: binder thread calling setParameters():
                 //  1. Lock mLock
@@ -353,7 +356,6 @@
                 void        dump(int fd, const Vector<String16>& args);
 
     // Thread virtuals
-    virtual     status_t    readyToRun();
     virtual     bool        threadLoop();
 
     // RefBase
@@ -419,7 +421,7 @@
                                 int sessionId,
                                 IAudioFlinger::track_flags_t *flags,
                                 pid_t tid,
-                                status_t *status);
+                                status_t *status /*non-NULL*/);
 
                 AudioStreamOut* getOutput() const;
                 AudioStreamOut* clearOutput();
@@ -471,7 +473,6 @@
     size_t                          mNormalFrameCount;  // normal mixer and effects
 
     int16_t*                        mMixBuffer;         // frame size aligned mix buffer
-    int8_t*                         mAllocMixBuffer;    // mixer buffer allocation address
 
     // suspend count, > 0 means suspended.  While suspended, the thread continues to pull from
     // tracks and mix, but doesn't write to HAL.  A2DP and SCO HAL implementations can't handle
@@ -824,7 +825,6 @@
 
     // Thread virtuals
     virtual bool        threadLoop();
-    virtual status_t    readyToRun();
 
     // RefBase
     virtual void        onFirstRef();
@@ -839,7 +839,7 @@
                     int sessionId,
                     IAudioFlinger::track_flags_t *flags,
                     pid_t tid,
-                    status_t *status);
+                    status_t *status /*non-NULL*/);
 
             status_t    start(RecordTrack* recordTrack,
                               AudioSystem::sync_event_t event,
@@ -903,7 +903,6 @@
             int32_t                             *mRsmpOutBuffer;
             int16_t                             *mRsmpInBuffer; // [mFrameCount * mChannelCount]
             size_t                              mRsmpInIndex;
-            size_t                              mBufferSize;    // stream buffer size for read()
             const uint32_t                      mReqChannelCount;
             const uint32_t                      mReqSampleRate;
             ssize_t                             mBytesRead;
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 523e4b2..00a91b7 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -47,6 +47,7 @@
                                 int sessionId,
                                 bool isOut);
     virtual             ~TrackBase();
+    virtual status_t    initCheck() const { return getCblk() != 0 ? NO_ERROR : NO_MEMORY; }
 
     virtual status_t    start(AudioSystem::sync_event_t event,
                              int triggerSession) = 0;
@@ -76,15 +77,6 @@
 
     virtual uint32_t sampleRate() const { return mSampleRate; }
 
-    // Return a pointer to the start of a contiguous slice of the track buffer.
-    // Parameter 'offset' is the requested start position, expressed in
-    // monotonically increasing frame units relative to the track epoch.
-    // Parameter 'frames' is the requested length, also in frame units.
-    // Always returns non-NULL.  It is the caller's responsibility to
-    // verify that this will be successful; the result of calling this
-    // function with invalid 'offset' or 'frames' is undefined.
-    void* getBuffer(uint32_t offset, uint32_t frames) const;
-
     bool isStopped() const {
         return (mState == STOPPED || mState == FLUSHED);
     }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index e676365..6039078 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -361,6 +361,15 @@
     ALOGV("PlaybackThread::Track destructor");
 }
 
+status_t AudioFlinger::PlaybackThread::Track::initCheck() const
+{
+    status_t status = TrackBase::initCheck();
+    if (status == NO_ERROR && mName < 0) {
+        status = NO_MEMORY;
+    }
+    return status;
+}
+
 void AudioFlinger::PlaybackThread::Track::destroy()
 {
     // NOTE: destroyTrack_l() can remove a strong reference to this Track
@@ -1651,9 +1660,7 @@
 {
     ALOGV("RecordTrack constructor");
     if (mCblk != NULL) {
-        mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
-                mFrameSize);
-        mServerProxy = mAudioRecordServerProxy;
+        mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount, mFrameSize);
     }
 }