diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index cef551c..092f512 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -154,8 +154,12 @@
       mBtNrecIsOff(false),
       mIsLowRamDevice(true),
       mIsDeviceTypeKnown(false),
+#ifdef QCOM_HARDWARE
       mGlobalEffectEnableTime(0),
       mAllChainsLocked(false)
+#else
+      mGlobalEffectEnableTime(0)
+#endif
 {
     getpid_cached = getpid();
     char value[PROPERTY_VALUE_MAX];
@@ -570,6 +574,7 @@
     return trackHandle;
 }
 
+#ifdef QCOM_HARDWARE
 sp<IDirectTrack> AudioFlinger::createDirectTrack(
         pid_t pid,
         uint32_t sampleRate,
@@ -676,16 +681,19 @@
         mLPAEffectChain = NULL;
     }
 }
+#endif
 
 uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
 {
     Mutex::Autolock _l(mLock);
+#ifdef QCOM_HARDWARE
     if (!mDirectAudioTracks.isEmpty()) {
         AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output);
         if(desc != NULL) {
             return desc->stream->common.get_sample_rate(&desc->stream->common);
         }
     }
+#endif
     PlaybackThread *thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
         ALOGW("sampleRate() unknown thread %d", output);
@@ -697,10 +705,12 @@
 int AudioFlinger::channelCount(audio_io_handle_t output) const
 {
     Mutex::Autolock _l(mLock);
+#ifdef QCOM_HARDWARE
     AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output);
     if(desc != NULL) {
         return desc->stream->common.get_channels(&desc->stream->common);
     }
+#endif
     PlaybackThread *thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
         ALOGW("channelCount() unknown thread %d", output);
@@ -723,10 +733,12 @@
 size_t AudioFlinger::frameCount(audio_io_handle_t output) const
 {
     Mutex::Autolock _l(mLock);
+#ifdef QCOM_HARDWARE
     AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output);
     if(desc != NULL) {
         return desc->stream->common.get_buffer_size(&desc->stream->common);
     }
+#endif
     PlaybackThread *thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
         ALOGW("frameCount() unknown thread %d", output);
@@ -740,12 +752,14 @@
 uint32_t AudioFlinger::latency(audio_io_handle_t output) const
 {
     Mutex::Autolock _l(mLock);
+#ifdef QCOM_HARDWARE
     if (!mDirectAudioTracks.isEmpty()) {
         AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output);
         if (desc != NULL) {
             return desc->stream->get_latency(desc->stream);
         }
     }
+#endif
 
     PlaybackThread *thread = checkPlaybackThread_l(output);
     if (thread == NULL) {
@@ -937,6 +951,7 @@
     }
 
     AutoMutex lock(mLock);
+#ifdef QCOM_HARDWARE
     ALOGV("setStreamVolume stream %d, output %d, value %f",stream, output, value);
     AudioSessionDescriptor *desc = NULL;
     if (!mDirectAudioTracks.isEmpty()) {
@@ -953,13 +968,16 @@
             }
         }
     }
+#endif
     PlaybackThread *thread = NULL;
     if (output) {
         thread = checkPlaybackThread_l(output);
         if (thread == NULL) {
+#ifdef QCOM_HARDWARE
             if (desc != NULL) {
                 return NO_ERROR;
             }
+#endif
             return BAD_VALUE;
         }
     }
@@ -1094,6 +1112,7 @@
         return final_result;
     }
 
+#ifdef QCOM_HARDWARE
     AudioSessionDescriptor *desc = NULL;
     if (!mDirectAudioTracks.isEmpty()) {
         desc = mDirectAudioTracks.valueFor(ioHandle);
@@ -1116,6 +1135,7 @@
             }
         }
     }
+#endif
 
     // hold a strong ref on thread in case closeOutput() or closeInput() is called
     // and the thread is exited once the lock is released
@@ -1284,6 +1304,7 @@
     }
 }
 
+#ifdef QCOM_HARDWARE
 status_t AudioFlinger::deregisterClient(const sp<IAudioFlingerClient>& client)
 {
     ALOGV("deregisterClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
@@ -1298,6 +1319,7 @@
 
     return false;
 }
+#endif
 
 void AudioFlinger::removeNotificationClient(pid_t pid)
 {
@@ -1329,16 +1351,19 @@
 // audioConfigChanged_l() must be called with AudioFlinger::mLock held
 void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2)
 {
+#ifdef QCOM_HARDWARE
     ALOGV("AudioFlinger::audioConfigChanged_l: event %d", event);
     if (event == AudioSystem::EFFECT_CONFIG_CHANGED) {
         mIsEffectConfigChanged = true;
     }
     if (!mNotificationClients.isEmpty()){
+#endif
         size_t size = mNotificationClients.size();
         for (size_t i = 0; i < size; i++) {
             mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
                                                                                    param2);
         }
+#ifdef QCOM_HARDWARE
     }
     if ((!mDirectAudioTracks.isEmpty())&& (event == AudioSystem::EFFECT_CONFIG_CHANGED)){
         size_t dsize = mDirectAudioTracks.size();
@@ -1352,6 +1377,7 @@
             }
         }
     }
+#endif
 }
 
 // removeClient_l() must be called with AudioFlinger::mLock held
@@ -1472,8 +1498,10 @@
     RecordThread *thread;
     size_t inFrameCount;
     int lSessionId;
+#ifdef QCOM_HARDWARE
     size_t inputBufferSize = 0;
     uint32_t channelCount = popcount(channelMask);
+#endif
 
     // check calling permissions
     if (!recordingAllowed()) {
@@ -1482,6 +1510,7 @@
         goto Exit;
     }
 
+#ifdef QCOM_HARDWARE
     // Check that audio input stream accepts requested audio parameters
     inputBufferSize = getInputBufferSize(sampleRate, format, channelCount);
     if (inputBufferSize == 0) {
@@ -1489,6 +1518,13 @@
         ALOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d",  sampleRate, format, channelCount);
         goto Exit;
     }
+#else
+    if (format != AUDIO_FORMAT_PCM_16_BIT) {
+        ALOGE("openRecord() invalid format %d", format);
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+#endif
 
     // add client to list
     { // scope for mLock
@@ -1520,6 +1556,7 @@
             }
         }
 
+#ifdef QCOM_HARDWARE
         // frameCount must be a multiple of input buffer size
         // Change for Codec type
         uint8_t channelCount = popcount(channelMask);
@@ -1542,6 +1579,7 @@
             }
         }
         frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
+#endif
 
         // create new record track.
         // The record track uses one track in mHardwareMixerThread by convention.
@@ -1779,6 +1817,7 @@
         if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
             thread = new OffloadThread(this, output, id, *pDevices);
             ALOGV("openOutput() created offload output: ID %d thread %p", id, thread);
+#ifdef QCOM_HARDWARE
         }
         if (flags & AUDIO_OUTPUT_FLAG_LPA || flags & AUDIO_OUTPUT_FLAG_TUNNEL ) {
             AudioSessionDescriptor *desc = new AudioSessionDescriptor(hwDevHal, outStream, flags);
@@ -1791,6 +1830,7 @@
             desc->trackRefPtr = NULL;
             mDirectAudioTracks.add(id, desc);
             mDirectDevice = desc->device;
+#endif
         } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) ||
             (config.format != AUDIO_FORMAT_PCM_16_BIT) ||
             (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) {
@@ -1818,6 +1858,7 @@
             // notify client processes of the new output creation
             thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
         }
+#ifdef QCOM_HARDWARE
         else {
             *pLatencyMs = 0;
             if ((flags & AUDIO_OUTPUT_FLAG_LPA) || (flags & AUDIO_OUTPUT_FLAG_TUNNEL)) {
@@ -1825,6 +1866,7 @@
                 *pLatencyMs = desc->stream->get_latency(desc->stream);
             }
         }
+#endif
 
 
         // the first primary output opened designates the primary hw device
@@ -1877,6 +1919,7 @@
 {
     // keep strong reference on the playback thread so that
     // it is not destroyed while exit() is executed
+#ifdef QCOM_HARDWARE
     AudioSessionDescriptor *desc = mDirectAudioTracks.valueFor(output);
     if (desc) {
         ALOGV("Closing DirectTrack output %d", output);
@@ -1890,6 +1933,7 @@
         delete desc;
         return NO_ERROR;
     }
+#endif
 
     sp<PlaybackThread> thread;
     {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6392bed..9844659 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -29,8 +29,10 @@
 
 #include <media/IAudioFlinger.h>
 #include <media/IAudioFlingerClient.h>
+#ifdef QCOM_HARDWARE
 #include <media/IDirectTrack.h>
 #include <media/IDirectTrackClient.h>
+#endif
 #include <media/IAudioTrack.h>
 #include <media/IAudioRecord.h>
 #include <media/AudioSystem.h>
@@ -91,7 +93,11 @@
 
 static uint32_t getInputChannelCount(uint32_t channels) {
     // only mono, stereo, and 5.1 are supported for input sources
+#ifdef QCOM_HARDWARE
     return popcount((channels)&(AUDIO_CHANNEL_IN_STEREO|AUDIO_CHANNEL_IN_MONO|AUDIO_CHANNEL_IN_5POINT1));
+#else
+    return popcount(channels);
+#endif
 }
 class AudioFlinger :
     public BinderService<AudioFlinger>,
@@ -117,6 +123,7 @@
                                 int *sessionId,
                                 String8& name,
                                 status_t *status);
+#ifdef QCOM_HARDWARE
     virtual sp<IDirectTrack> createDirectTrack(
                                 pid_t pid,
                                 uint32_t sampleRate,
@@ -127,6 +134,7 @@
                                 audio_stream_type_t streamType,
                                 status_t *status);
     virtual void deleteEffectSession();
+#endif
     virtual sp<IAudioRecord> openRecord(
                                 audio_io_handle_t input,
                                 uint32_t sampleRate,
@@ -167,7 +175,9 @@
     virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
 
     virtual     void        registerClient(const sp<IAudioFlingerClient>& client);
-    virtual    status_t     deregisterClient(const sp<IAudioFlingerClient>& client);
+#ifdef QCOM_HARDWARE
+    virtual     status_t    deregisterClient(const sp<IAudioFlingerClient>& client);
+#endif
     virtual     size_t      getInputBufferSize(uint32_t sampleRate, audio_format_t format,
                                                audio_channel_mask_t channelMask) const;
 
@@ -245,11 +255,13 @@
                                 Parcel* reply,
                                 uint32_t flags);
 
+#ifdef QCOM_HARDWARE
     bool applyEffectsOn(void *token,
                         int16_t *buffer1,
                         int16_t *buffer2,
                         int size,
                         bool force);
+#endif
 
     // end of IAudioFlinger interface
 
@@ -396,7 +408,9 @@
     class EffectModule;
     class EffectHandle;
     class EffectChain;
+#ifdef QCOM_HARDWARE
     struct AudioSessionDescriptor;
+#endif
     struct AudioStreamOut;
     struct AudioStreamIn;
 
@@ -486,6 +500,7 @@
 
               sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId);
 
+#ifdef QCOM_HARDWARE
     // server side of the client's IAudioTrack
     class DirectAudioTrack : public android::BnDirectTrack,
                              public AudioEventObserver
@@ -596,6 +611,7 @@
         sp<IBinder>             mWakeLockToken;
         sp<PMDeathRecipient>    mDeathRecipient;
     };
+#endif
                 void        removeClient_l(pid_t pid);
                 void        removeNotificationClient(pid_t pid);
 
@@ -657,6 +673,7 @@
         AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) :
             audioHwDev(dev), stream(in) {}
     };
+#ifdef QCOM_HARDWARE
     struct AudioSessionDescriptor {
         bool    mActive;
         int     mStreamType;
@@ -671,6 +688,7 @@
         AudioSessionDescriptor(audio_hw_device_t *dev, audio_stream_out_t *out, audio_output_flags_t outflag) :
             hwDev(dev), stream(out), flag(outflag)  {}
     };
+#endif
 
     // for mAudioSessionRefs only
     struct AudioSessionRef {
@@ -735,16 +753,20 @@
                 volatile int32_t                    mNextUniqueId;  // updated by android_atomic_inc
                 audio_mode_t                        mMode;
                 bool                                mBtNrecIsOff;
+#ifdef QCOM_HARDWARE
                 DefaultKeyedVector<audio_io_handle_t, AudioSessionDescriptor *> mDirectAudioTracks;
                 // protected by mLock
                 volatile bool                       mIsEffectConfigChanged;
+#endif
                 Vector<AudioSessionRef*> mAudioSessionRefs;
+#ifdef QCOM_HARDWARE
                 sp<EffectChain> mLPAEffectChain;
                 int         mLPASessionId;
                 audio_devices_t mDirectDevice;//device for directTrack,used for effects
                 int                                 mLPASampleRate;
                 int                                 mLPANumChannels;
                 volatile bool                       mAllChainsLocked;
+#endif
                 float       masterVolume_l() const;
                 bool        masterMute_l() const;
                 audio_module_handle_t loadHwModule_l(const char *name);
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index f45476c..95f7381 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -561,7 +561,11 @@
                       (value == 48000 && devSampleRate == 44100))) {
                     quality = AudioResampler::LOW_QUALITY;
                 } else {
+#ifdef QCOM_HARDWARE
                     quality = AudioResampler::VERY_HIGH_QUALITY;
+#else
+                    quality = AudioResampler::DEFAULT_QUALITY;
+#endif
                 }
                 resampler = AudioResampler::create(
                         format,
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index f916dfc..a8f0511 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -72,7 +72,11 @@
       mStatus(NO_INIT), mState(IDLE),
       // mMaxDisableWaitCnt is set by configure() and not used before then
       // mDisableWaitCnt is set by process() and updateState() and not used before then
+#ifdef QCOM_HARDWARE
       mSuspended(false), mIsForLPA(false)
+#else
+      mSuspended(false)
+#endif
 {
     ALOGV("Constructor %p", this);
     int lStatus;
@@ -307,14 +311,20 @@
     (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL);
 }
 
+#ifdef QCOM_HARDWARE
 status_t AudioFlinger::EffectModule::configure(bool isForLPA, int sampleRate, int channelCount, int frameCount)
+#else
+status_t AudioFlinger::EffectModule::configure()
+#endif
 {
     status_t status;
     status_t cmdStatus = 0;
     sp<ThreadBase> thread;
     uint32_t size;
     audio_channel_mask_t channelMask;
+#ifdef QCOM_HARDWARE
     uint32_t channels;
+#endif
 
     if (mEffectInterface == NULL) {
         status = NO_INIT;
@@ -329,6 +339,7 @@
 
     // TODO: handle configuration of effects replacing track process
     channelMask = thread->channelMask();
+#ifdef QCOM_HARDWARE
     if(popcount(channelMask) > 2) {
         ALOGE("Error: Trying to apply effect on  %d channel content",popcount(channelMask));
         return INVALID_OPERATION;
@@ -347,6 +358,7 @@
             channels = AUDIO_CHANNEL_OUT_STEREO;
         }
     }
+#endif
 
     if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
         mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO;
@@ -356,11 +368,15 @@
     mConfig.outputCfg.channels = channelMask;
     mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
     mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+#ifdef QCOM_HARDWARE
     if(isForLPA){
         mConfig.inputCfg.samplingRate = sampleRate;
     } else {
+#endif
         mConfig.inputCfg.samplingRate = thread->sampleRate();
+#ifdef QCOM_HARDWARE
     }
+#endif
     mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
     mConfig.inputCfg.bufferProvider.cookie = NULL;
     mConfig.inputCfg.bufferProvider.getBuffer = NULL;
@@ -385,11 +401,15 @@
     }
     mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
     mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
+#ifdef QCOM_HARDWARE
     if(isForLPA) {
         mConfig.inputCfg.buffer.frameCount = frameCount;
     } else {
+#endif
         mConfig.inputCfg.buffer.frameCount = thread->frameCount();
+#ifdef QCOM_HARDWARE
     }
+#endif
     mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
 
     ALOGV("configure() %p thread %p buffer %p framecount %d",
@@ -586,11 +606,15 @@
 // must be called with EffectModule::mLock held
 status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled)
 {
+#ifdef QCOM_HARDWARE
     bool effectStateChanged = false;
+#endif
     ALOGV("setEnabled %p enabled %d", this, enabled);
 
     if (enabled != isEnabled()) {
+#ifdef QCOM_HARDWARE
         effectStateChanged = true;
+#endif
         status_t status = AudioSystem::setEffectEnabled(mId, enabled);
         if (enabled && status != NO_ERROR) {
             return status;
@@ -628,6 +652,7 @@
             }
         }
     }
+#ifdef QCOM_HARDWARE
     /*
        Send notification event to LPA Player when an effect for
        LPA output is enabled or disabled.
@@ -639,6 +664,7 @@
         thread->effectConfigChanged();
         lock();
     }
+#endif
     return NO_ERROR;
 }
 
@@ -1182,6 +1208,7 @@
         return disable();
     }
 
+#ifdef QCOM_HARDWARE
     if(mEffect->isOnLPA() &&
        ((cmdCode == EFFECT_CMD_SET_PARAM) || (cmdCode == EFFECT_CMD_SET_PARAM_DEFERRED) ||
         (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) || (cmdCode == EFFECT_CMD_SET_DEVICE) ||
@@ -1191,6 +1218,7 @@
         ALOGV("Notifying Direct Track for the change in effect config %d", cmdCode);
         mClient->audioFlinger()->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
     }
+#endif
     return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
 }
 
@@ -1258,7 +1286,11 @@
                                         int sessionId)
     : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
       mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
+#ifdef QCOM_HARDWARE
       mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX), mIsForLPATrack(false)
+#else
+      mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
+#endif
 {
     mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
     if (thread == NULL) {
@@ -1304,6 +1336,7 @@
     return 0;
 }
 
+#ifdef QCOM_HARDWARE
 sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromIndex_l(int idx)
 {
     sp<EffectModule> effect = NULL;
@@ -1315,6 +1348,7 @@
     }
     return effect;
 }
+#endif
 
 // getEffectFromType_l() must be called with ThreadBase::mLock held
 sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
@@ -1381,7 +1415,11 @@
     }
 
     size_t size = mEffects.size();
+#ifdef QCOM_HARDWARE
     if (doProcess || isForLPATrack()) {
+#else
+    if (doProcess) {
+#endif
         for (size_t i = 0; i < size; i++) {
             mEffects[i]->process();
         }
@@ -1859,6 +1897,7 @@
     return false;
 }
 
+#ifdef QCOM_HARDWARE
 #define DEAFULT_FRAME_COUNT 1200
 bool AudioFlinger::applyEffectsOn(void *token, int16_t *inBuffer,
                             int16_t *outBuffer, int size, bool force)
@@ -2033,5 +2072,6 @@
     ALOGV("Creating Effects Thread");
     pthread_create(&mEffectsThread, &attr, EffectsThreadWrapper, this);
 }
+#endif
 
 }; // namespace android
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 10c2227..5ad312b 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -69,10 +69,14 @@
                      void *pReplyData);
 
     void reset_l();
+#ifdef QCOM_HARDWARE
     status_t configure(bool isForLPA = false,
                        int sampleRate = 0,
                        int channelCount = 0,
                        int frameCount = 0);
+#else
+    status_t configure();
+#endif
     status_t init();
     effect_state state() const {
         return mState;
@@ -123,8 +127,10 @@
                         { return (mDescriptor.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) != 0; }
     status_t         setOffloaded(bool offloaded, audio_io_handle_t io);
     bool             isOffloaded() const;
+#ifdef QCOM_HARDWARE
     bool             isOnLPA() { return mIsForLPA;}
     void             setLPAFlag(bool isForLPA) {mIsForLPA = isForLPA; }
+#endif
 
     void             dump(int fd, const Vector<String16>& args);
 
@@ -159,7 +165,9 @@
     uint32_t mDisableWaitCnt;       // current process() calls count during disable period.
     bool     mSuspended;            // effect is suspended: temporarily disabled by framework
     bool     mOffloaded;            // effect is currently offloaded to the audio DSP
+#ifdef QCOM_HARDWARE
     bool     mIsForLPA;
+#endif
 };
 
 // The EffectHandle class implements the IEffect interface. It provides resources
@@ -269,14 +277,18 @@
 
     status_t addEffect_l(const sp<EffectModule>& handle);
     size_t removeEffect_l(const sp<EffectModule>& handle);
+#ifdef QCOM_HARDWARE
     size_t getNumEffects() { return mEffects.size(); }
+#endif
 
     int sessionId() const { return mSessionId; }
     void setSessionId(int sessionId) { mSessionId = sessionId; }
 
     sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
     sp<EffectModule> getEffectFromId_l(int id);
+#ifdef QCOM_HARDWARE
     sp<EffectModule> getEffectFromIndex_l(int idx);
+#endif
     sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
     bool setVolume_l(uint32_t *left, uint32_t *right);
     void setDevice_l(audio_devices_t device);
@@ -326,8 +338,10 @@
 
 
     void dump(int fd, const Vector<String16>& args);
+#ifdef QCOM_HARDWARE
     bool isForLPATrack() {return mIsForLPATrack; }
     void setLPAFlag(bool flag) {mIsForLPATrack = flag;}
+#endif
 
 protected:
     friend class AudioFlinger;  // for mThread, mEffects
@@ -376,7 +390,9 @@
     uint32_t mNewLeftVolume;       // new volume on left channel
     uint32_t mNewRightVolume;      // new volume on right channel
     uint32_t mStrategy; // strategy for this effect chain
+#ifdef QCOM_HARDWARE
     bool     mIsForLPATrack;
+#endif
     // mSuspendedEffects lists all effects currently suspended in the chain.
     // Use effect type UUID timelow field as key. There is no real risk of identical
     // timeLow fields among effect type UUIDs.
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index cb54348..4b0c4d8 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -28,7 +28,9 @@
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
+#ifdef QCOM_HARDWARE
                                 uint32_t flags,
+#endif
                                 int sessionId);
     virtual             ~RecordTrack();
 
@@ -58,7 +60,9 @@
                                    int64_t pts = kInvalidPTS);
     // releaseBuffer() not overridden
 
+#ifdef QCOM_HARDWARE
     track_flags_t       mFlags;
+#endif
     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 568329f..aa9097b 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -85,9 +85,11 @@
 #define ALOGVV(a...) do { } while(0)
 #endif
 
+#ifdef QCOM_HARDWARE
 #define DIRECT_TRACK_EOS 1
 #define DIRECT_TRACK_HW_FAIL 6
 static const char lockName[] = "DirectTrack";
+#endif
 namespace android {
 
 // retry counts for buffer fill timeout
@@ -349,10 +351,12 @@
     return status;
 }
 
+#ifdef QCOM_HARDWARE
 void AudioFlinger::ThreadBase::effectConfigChanged() {
     ALOGV("New effect is being added to LPA chain, Notifying LPA Direct Track");
     mAudioFlinger->audioConfigChanged_l(AudioSystem::EFFECT_CONFIG_CHANGED, 0, NULL);
 }
+#endif
 
 void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param)
 {
@@ -782,6 +786,7 @@
             addEffectChain_l(chain);
             chain->setStrategy(getStrategyForSession_l(sessionId));
             chainCreated = true;
+#ifdef QCOM_HARDWARE
             if(sessionId == mAudioFlinger->mLPASessionId) {
                 // Clear reference to previous effect chain if any
                 if(mAudioFlinger->mLPAEffectChain.get()) {
@@ -795,6 +800,7 @@
                 uint32_t volume = 0x1000000; // Equals to 1.0 in 8.24 format
                 chain->setVolume_l(&volume,&volume);
             }
+#endif
         } else {
             effect = chain->getEffectFromDesc_l(desc);
         }
@@ -823,7 +829,11 @@
             }
             effectCreated = true;
 
+#ifdef QCOM_HARDWARE
             effect->setDevice(mAudioFlinger->mLPASessionId == sessionId ? mAudioFlinger->mDirectDevice:mOutDevice);
+#else
+            effect->setDevice(mOutDevice);
+#endif
             effect->setDevice(mInDevice);
             effect->setMode(mAudioFlinger->getMode());
             effect->setAudioSource(mAudioSource);
@@ -938,13 +948,20 @@
         Vector< sp<AudioFlinger::EffectChain> >& effectChains)
 {
     effectChains = mEffectChains;
+#ifdef QCOM_HARDWARE
     mAudioFlinger->mAllChainsLocked = true;
+#endif
     for (size_t i = 0; i < mEffectChains.size(); i++) {
+#ifdef QCOM_HARDWARE
         if (mEffectChains[i] != mAudioFlinger->mLPAEffectChain) {
+#else
             mEffectChains[i]->lock();
+#endif
+#ifdef QCOM_HARDWARE
         } else {
             mAudioFlinger-> mAllChainsLocked = false;
         }
+#endif
     }
 }
 
@@ -952,7 +969,9 @@
         const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
 {
     for (size_t i = 0; i < effectChains.size(); i++) {
+#ifdef QCOM_HARDWARE
         if (mAudioFlinger->mAllChainsLocked || effectChains[i] != mAudioFlinger->mLPAEffectChain)
+#endif
             effectChains[i]->unlock();
     }
 }
@@ -1268,6 +1287,7 @@
     }
 
     if (mType == DIRECT) {
+#ifdef QCOM_HARDWARE
         if (((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM)
                ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AMR_NB)
                ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AMR_WB)
@@ -1275,6 +1295,9 @@
                ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_EVRCB)
                ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_EVRCWB)
                ||((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_EVRCNW)) {
+#else
+        if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
+#endif
             if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
                 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x "
                         "for output %p with format %d",
@@ -2317,9 +2340,13 @@
             // only process effects if we're going to write
             if (sleepTime == 0 && mType != OFFLOAD) {
                 for (size_t i = 0; i < effectChains.size(); i ++) {
+#ifdef QCOM_HARDWARE
                     if (effectChains[i] != mAudioFlinger->mLPAEffectChain) {
+#endif
                         effectChains[i]->process_l();
+#ifdef QCOM_HARDWARE
                   }
+#endif
                }
             }
         }
@@ -4530,6 +4557,7 @@
                         }
                         if (framesOut && (mFrameCount == mRsmpInIndex)) {
                             void *readInto;
+#ifdef QCOM_HARDWARE
                             int InputBytes;
                             /*Fix me: How does framesOut become 0 in this condition */
                             if (( framesOut != mFrameCount) &&
@@ -4549,8 +4577,21 @@
                                      mRsmpInIndex = 0;
                                      InputBytes = mBufferSize;
                              }
+#else
+                            if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) {
+                                readInto = buffer.raw;
+                                framesOut = 0;
+                            } else {
+                                readInto = mRsmpInBuffer;
+                                mRsmpInIndex = 0;
+                            }
+#endif
                             mBytesRead = mInput->stream->read(mInput->stream, readInto,
+#ifdef QCOM_HARDWARE
                                     InputBytes);
+#else
+                                    mBufferSize);
+#endif
                             if (mBytesRead <= 0) {
                                 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE))
                                 {
@@ -4750,7 +4791,11 @@
         Mutex::Autolock _l(mLock);
 
         track = new RecordTrack(this, client, sampleRate,
+#ifdef QCOM_HARDWARE
                       format, channelMask, frameCount, *flags, sessionId);
+#else
+                      format, channelMask, frameCount, sessionId);
+#endif
 
         if (track->getCblk() == 0) {
             ALOGE("createRecordTrack_l() no control block");
@@ -5226,6 +5271,11 @@
     mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
     mChannelCount = (uint16_t)getInputChannelCount(mChannelMask);
     mFormat = mInput->stream->common.get_format(&mInput->stream->common);
+#ifndef QCOM_HARDWARE
+    if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
+        ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat);
+    }
+#endif
     mFrameSize = audio_stream_frame_size(&mInput->stream->common);
     mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common);
     mFrameCount = mBufferSize / mFrameSize;
@@ -5345,6 +5395,7 @@
     return 0;
 }
 
+#ifdef QCOM_HARDWARE
 // ----------------------------------------------------------------------------
 
 AudioFlinger::DirectAudioTrack::DirectAudioTrack(const sp<AudioFlinger>& audioFlinger,
@@ -5659,5 +5710,6 @@
 }
 
 // ----------------------------------------------------------------------------
+#endif
 
 }; // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index eff4bb7..0d01d05 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -138,7 +138,9 @@
     virtual     status_t    setParameters(const String8& keyValuePairs);
     virtual     String8     getParameters(const String8& keys) = 0;
     virtual     void        audioConfigChanged_l(int event, int param = 0) = 0;
+#ifdef QCOM_HARDWARE
                 void        effectConfigChanged();
+#endif
                 void        sendIoConfigEvent(int event, int param = 0);
                 void        sendIoConfigEvent_l(int event, int param = 0);
                 void        sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio);
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 065e020..358d74a 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -43,7 +43,9 @@
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
+#ifdef QCOM_HARDWARE
                                 uint32_t flags,
+#endif
                                 const sp<IMemory>& sharedBuffer,
                                 int sessionId,
                                 bool isOut);
@@ -131,7 +133,9 @@
                                     // 8-bit PCM samples are stored as 16-bit
     const size_t        mFrameCount;// size of track buffer given at createTrack() or
                                     // openRecord(), and then adjusted as needed
+#ifdef QCOM_HARDWARE
     uint32_t            mFlags;
+#endif
     const int           mSessionId;
     Vector < sp<SyncEvent> >mSyncEvents;
     const bool          mIsOut;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 436d4de..dfae5d9 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -66,7 +66,9 @@
             audio_format_t format,
             audio_channel_mask_t channelMask,
             size_t frameCount,
+#ifdef QCOM_HARDWARE
             uint32_t flags,
+#endif
             const sp<IMemory>& sharedBuffer,
             int sessionId,
             bool isOut)
@@ -80,10 +82,17 @@
         mFormat(format),
         mChannelMask(channelMask),
         mChannelCount(popcount(channelMask)),
+#ifdef QCOM_HARDWARE
         mFrameSize((audio_is_linear_pcm(format) || audio_is_supported_compressed(format)) ?
         ((flags & IAudioFlinger::TRACK_VOICE_COMMUNICATION)? mChannelCount * sizeof(int16_t) : mChannelCount * audio_bytes_per_sample(format)) : sizeof(int8_t)),
+#else
+        mFrameSize(audio_is_linear_pcm(format) ?
+                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
+#endif
         mFrameCount(frameCount),
+#ifdef QCOM_HARDWARE
         mFlags(0),
+#endif
         mSessionId(sessionId),
         mIsOut(isOut),
         mServerProxy(NULL),
@@ -98,6 +107,7 @@
 
     // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
     size_t size = sizeof(audio_track_cblk_t);
+#ifdef QCOM_HARDWARE
     uint8_t channelCount = popcount(channelMask);
     size_t bufferSize = 0;
     if (flags & IAudioFlinger::TRACK_VOICE_COMMUNICATION) {
@@ -118,6 +128,9 @@
           bufferSize = frameCount * channelCount * AMR_WB_FRAMESIZE; // full rate frame size
        }
     }
+#else
+    size_t bufferSize = (sharedBuffer == 0 ? roundup(frameCount) : frameCount) * mFrameSize;
+#endif
 
     if (sharedBuffer == 0) {
         size += bufferSize;
@@ -146,6 +159,7 @@
         mCblk->frameCount_ = frameCount;
         if (sharedBuffer == 0) {
             mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+#ifdef QCOM_HARDWARE
             if (flags & IAudioFlinger::TRACK_VOICE_COMMUNICATION) {
                 memset(mBuffer, 0, bufferSize);
             } else {
@@ -169,6 +183,9 @@
                     memset(mBuffer, 0, frameCount * channelCount * AMR_WB_FRAMESIZE);
                 }
             }
+#else
+            memset(mBuffer, 0, bufferSize);
+#endif
         } else {
             mBuffer = sharedBuffer->pointer();
 #if 0
@@ -358,9 +375,14 @@
             const sp<IMemory>& sharedBuffer,
             int sessionId,
             IAudioFlinger::track_flags_t flags)
+#ifdef QCOM_HARDWARE
     :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
      ((audio_stream_type_t)streamType == AUDIO_STREAM_VOICE_CALL)? IAudioFlinger::TRACK_VOICE_COMMUNICATION:0x0,
      sharedBuffer, sessionId, true /*isOut*/),
+#else
+    :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
+            sessionId, true /*isOut*/),
+#endif
     mFillingUpStatus(FS_INVALID),
     // mRetryCount initialized later when needed
     mSharedBuffer(sharedBuffer),
@@ -1783,14 +1805,23 @@
             audio_format_t format,
             audio_channel_mask_t channelMask,
             size_t frameCount,
+#ifdef QCOM_HARDWARE
             uint32_t flags,
+#endif
             int sessionId)
+#ifdef QCOM_HARDWARE
     :   TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
         flags, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
+#else
+    :   TrackBase(thread, client, sampleRate, format,
+                  channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
+#endif
         mOverflow(false)
 {
     ALOGV("RecordTrack constructor");
+#ifdef QCOM_HARDWARE
     mFlags = flags;
+#endif
     if (mCblk != NULL) {
         mAudioRecordServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
                 mFrameSize);