Merge branch 'cm-13.0' of https://github.com/CyanogenMod/android_frameworks_av into replicant-6.0replicant-6.0-alpha-0003replicant-6.0-alpha-0002

18 files changed, 230 insertions, 40 deletions

diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 8406ed6..0043bdc 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -148,6 +148,8 @@ public:
         TONE_CDMA_ABBR_ALERT,
         TONE_CDMA_SIGNAL_OFF,
         //CDMA end
+        TONE_HOLD_RECALL,
+
         NUM_TONES,
         NUM_SUP_TONES = LAST_SUP_TONE-FIRST_SUP_TONE+1
     };
diff --git a/media/libavextensions/stagefright/AVUtils.cpp b/media/libavextensions/stagefright/AVUtils.cpp
index bdf5eb6..413026b 100644
--- a/media/libavextensions/stagefright/AVUtils.cpp
+++ b/media/libavextensions/stagefright/AVUtils.cpp
@@ -448,7 +448,7 @@ status_t AVUtils::HEVCMuxer::makeHEVCCodecSpecificData(
     List<HEVCParamSet> picParamSets;
 
     if ((*codecSpecificDataSize = parseHEVCCodecSpecificData(data, size,
-                                   vidParamSets, seqParamSets, picParamSets)) <= 0) {
+                                   vidParamSets, seqParamSets, picParamSets)) == 0) {
         ALOGE("cannot parser codec specific data, bailing out");
         return ERROR_MALFORMED;
     }
@@ -885,12 +885,12 @@ size_t AVUtils::HEVCMuxer::parseHEVCCodecSpecificData(
             }
         } else {
             ALOGE("Only VPS, SPS and PPS Nal units are expected");
-            return ERROR_MALFORMED;
+            return 0;
         }
 
         if (nextStartCode == NULL) {
             ALOGE("Next start code is NULL");
-            return ERROR_MALFORMED;
+            return 0;
         }
 
         // Move on to find the next parameter set
diff --git a/media/libmedia/ICrypto.cpp b/media/libmedia/ICrypto.cpp
index 5d822cf..bc696ca 100644
--- a/media/libmedia/ICrypto.cpp
+++ b/media/libmedia/ICrypto.cpp
@@ -325,7 +325,9 @@ status_t BnCrypto::onTransact(
 
             if (overflow || sumSubsampleSizes != totalSize) {
                 result = -EINVAL;
-            } else if (offset + totalSize > sharedBuffer->size()) {
+            } else if (totalSize > sharedBuffer->size()) {
+                result = -EINVAL;
+            } else if ((size_t)offset > sharedBuffer->size() - totalSize) {
                 result = -EINVAL;
             } else {
                 result = decrypt(
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index d738ea7..af75e0f 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -698,7 +698,11 @@ const ToneGenerator::ToneDescriptor ToneGenerator::sToneDescriptors[] = {
           { .segments = { { .duration = 0, .waveFreq = { 0 }, 0, 0 }},
           .repeatCnt = 0,
           .repeatSegment = 0 },                            // TONE_CDMA_SIGNAL_OFF
-
+        { .segments = { { .duration = 15000, .waveFreq = { 0 }, 0, 0 },
+                      { .duration = 500, .waveFreq = { 450, 0 }, 0, 0 },
+                      { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
+          .repeatCnt = ToneGenerator::TONEGEN_INF,
+          .repeatSegment = 0 },                              // TONE_HOLD_RECALL
         { .segments = { { .duration = ToneGenerator::TONEGEN_INF, .waveFreq = { 350, 440, 0 }, 0, 0 },
                         { .duration = 0 , .waveFreq = { 0 }, 0, 0}},
           .repeatCnt = ToneGenerator::TONEGEN_INF,
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 7c71e4e..4383fce 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -750,12 +750,19 @@ void NuPlayerDriver::notifyListener_l(
                     }
                 }
                 if (mLooping || mAutoLoop) {
-                    mPlayer->seekToAsync(0);
-                    if (mAudioSink != NULL) {
-                        // The renderer has stopped the sink at the end in order to play out
-                        // the last little bit of audio. If we're looping, we need to restart it.
-                        mAudioSink->start();
+                    if (mState == STATE_RUNNING) {
+                        mPlayer->seekToAsync(0);
+                        if (mAudioSink != NULL) {
+                            // The renderer has stopped the sink at the end in order to play out
+                            // the last little bit of audio. If we're looping, we need to restart it.
+                            mAudioSink->start();
+                        }
+                    } else {
+                        mPlayer->pause();
+                        mState = STATE_PAUSED;
+                        mAtEOS = true;
                     }
+
                     // don't send completion event when looping
                     return;
                 }
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index d959e62..2336eb7 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -1590,6 +1590,13 @@ void NuPlayer::Renderer::onResume() {
             ALOGD("received error :%d on resume for offload track posting TEAR_DOWN event",status);
             notifyAudioTearDown();
         }
+        //Update anchor time after resuming playback.
+        if (offloadingAudio()) {
+            int64_t nowUs = ALooper::GetNowUs();
+            int64_t nowMediaUs =
+                mAudioFirstAnchorTimeMediaUs + getPlayedOutAudioDurationUs(nowUs);
+            mMediaClock->updateAnchor(nowMediaUs, nowUs, INT64_MAX);
+        }
     }
 
     {
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ad4676f..4807b65 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -925,7 +925,7 @@ status_t ACodec::setupNativeWindowSizeFormatAndUsage(
 #endif
 
     ALOGV("gralloc usage: %#x(OMX) => %#x(ACodec)", omxUsage, usage);
-    return setNativeWindowSizeFormatAndUsage(
+    err = setNativeWindowSizeFormatAndUsage(
             nativeWindow,
             def.format.video.nFrameWidth,
             def.format.video.nFrameHeight,
@@ -936,6 +936,26 @@ status_t ACodec::setupNativeWindowSizeFormatAndUsage(
 #endif
             mRotationDegrees,
             usage);
+#ifdef QCOM_HARDWARE
+    if (err == OK) {
+        OMX_CONFIG_RECTTYPE rect;
+        InitOMXParams(&rect);
+        rect.nPortIndex = kPortIndexOutput;
+        err = mOMX->getConfig(
+                mNode, OMX_IndexConfigCommonOutputCrop, &rect, sizeof(rect));
+        if (err == OK) {
+            ALOGV("rect size = %d, %d, %d, %d", rect.nLeft, rect.nTop, rect.nWidth, rect.nHeight);
+            android_native_rect_t crop;
+            crop.left = rect.nLeft;
+            crop.top = rect.nTop;
+            crop.right = rect.nLeft + rect.nWidth - 1;
+            crop.bottom = rect.nTop + rect.nHeight - 1;
+            ALOGV("crop update (%d, %d), (%d, %d)", crop.left, crop.top, crop.right, crop.bottom);
+            err = native_window_set_crop(nativeWindow, &crop);
+        }
+    }
+#endif
+    return err;
 }
 
 status_t ACodec::configureOutputBuffersFromNativeWindow(
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 4ee878a..e2ad924 100644
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -325,12 +325,6 @@ status_t CameraSource::isCameraColorFormatSupported(
     return OK;
 }
 
-static int32_t getHighSpeedFrameRate(const CameraParameters& params) {
-    const char* hsr = params.get("video-hsr");
-    int32_t rate = (hsr != NULL && strncmp(hsr, "off", 3)) ? atoi(hsr) : 0;
-    return rate > 240 ? 240 : rate;
-}
-
 /*
  * Configure the camera to use the requested video size
  * (width and height) and/or frame rate. If both width and
@@ -383,10 +377,6 @@ status_t CameraSource::configureCamera(
                 params->get(CameraParameters::KEY_SUPPORTED_PREVIEW_FRAME_RATES);
         CHECK(supportedFrameRates != NULL);
         ALOGV("Supported frame rates: %s", supportedFrameRates);
-        if (getHighSpeedFrameRate(*params)) {
-            ALOGI("Use default 30fps for HighSpeed %dfps", frameRate);
-            frameRate = 30;
-        }
         char buf[4];
         snprintf(buf, 4, "%d", frameRate);
         if (strstr(supportedFrameRates, buf) == NULL) {
@@ -488,8 +478,6 @@ status_t CameraSource::checkFrameRate(
         ALOGE("Failed to retrieve preview frame rate (%d)", frameRateActual);
         return UNKNOWN_ERROR;
     }
-    int32_t highSpeedRate = getHighSpeedFrameRate(params);
-    frameRateActual = highSpeedRate ? highSpeedRate : frameRateActual;
 
     // Check the actual video frame rate against the target/requested
     // video frame rate.
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 54c57ee..5e0ee55 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -2575,7 +2575,7 @@ ssize_t MediaCodec::dequeuePortBuffer(int32_t portIndex) {
         info->mOwnedByClient = true;
 
         // set image-data
-        if (info->mFormat != NULL) {
+        if (info->mFormat != NULL && mIsVideo) {
             sp<ABuffer> imageData;
             if (info->mFormat->findBuffer("image-data", &imageData)) {
                 info->mData->meta()->setBuffer("image-data", imageData);
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 5b96040..d39f34b 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -273,6 +273,9 @@ static VideoFrame *extractVideoFrame(
             if (err != OK) {
                 ALOGW("Input Error or EOS");
                 haveMoreInputs = false;
+                //correct the status to continue to get output from decoder
+                err = OK;
+                inputIndex = -1;
                 break;
             }
 
diff --git a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
index 8240f83..f2a4e65 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/SoftMPEG4Encoder.cpp
@@ -37,6 +37,10 @@
 
 #include <inttypes.h>
 
+#ifndef INT32_MAX
+#define INT32_MAX   2147483647
+#endif
+
 namespace android {
 
 template<class T>
@@ -137,6 +141,11 @@ OMX_ERRORTYPE SoftMPEG4Encoder::initEncParams() {
     if (mColorFormat != OMX_COLOR_FormatYUV420Planar || mInputDataIsMeta) {
         // Color conversion is needed.
         free(mInputFrameData);
+        mInputFrameData = NULL;
+        if (((uint64_t)mWidth * mHeight) > ((uint64_t)INT32_MAX / 3)) {
+            ALOGE("b/25812794, Buffer size is too big.");
+            return OMX_ErrorBadParameter;
+        }
         mInputFrameData =
             (uint8_t *) malloc((mWidth * mHeight * 3 ) >> 1);
         CHECK(mInputFrameData != NULL);
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
index e654843..410f9d0 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.cpp
@@ -26,6 +26,10 @@
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/MediaDefs.h>
 
+#ifndef INT32_MAX
+#define INT32_MAX   2147483647
+#endif
+
 namespace android {
 
 template<class T>
@@ -315,6 +319,11 @@ status_t SoftVPXEncoder::initEncoder() {
 
     if (mColorFormat != OMX_COLOR_FormatYUV420Planar || mInputDataIsMeta) {
         free(mConversionBuffer);
+        mConversionBuffer = NULL;
+        if (((uint64_t)mWidth * mHeight) > ((uint64_t)INT32_MAX / 3)) {
+            ALOGE("b/25812794, Buffer size is too big.");
+            return UNKNOWN_ERROR;
+        }
         mConversionBuffer = (uint8_t *)malloc(mWidth * mHeight * 3 / 2);
         if (mConversionBuffer == NULL) {
             ALOGE("Allocating conversion buffer failed.");
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 474fb46..0dd2af6 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -1,3 +1,23 @@
+#
+# This file was modified by DTS, Inc. The portions of the
+# code that are surrounded by "DTS..." are copyrighted and
+# licensed separately, as follows:
+#
+#  (C) 2015 DTS, Inc.
+#
+# 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.
+#
+
 LOCAL_PATH:= $(call my-dir)
 
 include $(CLEAR_VARS)
@@ -86,6 +106,11 @@ LOCAL_SRC_FILES += \
 LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
 
 LOCAL_CFLAGS += -fvisibility=hidden
+ifeq ($(strip $(BOARD_USES_SRS_TRUEMEDIA)),true)
+LOCAL_SHARED_LIBRARIES += libsrsprocessing
+LOCAL_CFLAGS += -DSRS_PROCESSING
+LOCAL_C_INCLUDES += $(TARGET_OUT_HEADERS)/mm-audio/audio-effects
+endif
 
 include $(BUILD_SHARED_LIBRARY)
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 1acfaad..23215dd 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -13,6 +13,25 @@
 ** 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.
+**
+** This file was modified by DTS, Inc. The portions of the
+** code that are surrounded by "DTS..." are copyrighted and
+** licensed separately, as follows:
+**
+**  (C) 2015 DTS, Inc.
+**
+** 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.
+**
 */
 
 
@@ -64,6 +83,9 @@
 #include <media/nbaio/PipeReader.h>
 #include <media/AudioParameter.h>
 #include <private/android_filesystem_config.h>
+#ifdef SRS_PROCESSING
+#include "postpro_patch.h"
+#endif
 
 // ----------------------------------------------------------------------------
 
@@ -1051,6 +1073,13 @@ status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8&
     if (ioHandle == AUDIO_IO_HANDLE_NONE) {
         Mutex::Autolock _l(mLock);
         status_t final_result = NO_ERROR;
+#ifdef SRS_PROCESSING
+        POSTPRO_PATCH_PARAMS_SET(keyValuePairs);
+        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+            PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+            thread->setPostPro();
+        }
+#endif
         {
             AutoMutex lock(mHardwareLock);
             mHardwareStatus = AUDIO_HW_SET_PARAMETER;
@@ -1145,6 +1174,9 @@ String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& k
 
     if (ioHandle == AUDIO_IO_HANDLE_NONE) {
         String8 out_s8;
+#ifdef SRS_PROCESSING
+        POSTPRO_PATCH_PARAMS_GET(keys, out_s8);
+#endif
 
         for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
             char *s;
@@ -1370,6 +1402,12 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId,
 
 
 
+void AudioFlinger::PlaybackThread::setPostPro()
+{
+    Mutex::Autolock _l(mLock);
+    if (mType == OFFLOAD)
+        broadcast_l();
+}
 // ----------------------------------------------------------------------------
 
 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
@@ -2991,7 +3029,7 @@ void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_hand
             bool firstRead = true;
 #define TEE_SINK_READ 1024                      // frames per I/O operation
             void *buffer = malloc(TEE_SINK_READ * frameSize);
-            CHECK (buffer != NULL);
+            ALOG_ASSERT(buffer != NULL);
             for (;;) {
                 size_t count = TEE_SINK_READ;
                 ssize_t actual = teeSource->read(buffer, count,
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 27a2f65..806eaf1 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -308,6 +308,11 @@ bool AudioMixer::setChannelMasks(int name,
 void AudioMixer::track_t::unprepareForDownmix() {
     ALOGV("AudioMixer::unprepareForDownmix(%p)", this);
 
+    if (mPostDownmixReformatBufferProvider != NULL) {
+        delete mPostDownmixReformatBufferProvider;
+        mPostDownmixReformatBufferProvider = NULL;
+        reconfigureBufferProviders();
+    }
     mDownmixRequiresFormat = AUDIO_FORMAT_INVALID;
     if (downmixerBufferProvider != NULL) {
         // this track had previously been configured with a downmixer, delete it
@@ -363,18 +368,9 @@ status_t AudioMixer::track_t::prepareForDownmix()
 
 void AudioMixer::track_t::unprepareForReformat() {
     ALOGV("AudioMixer::unprepareForReformat(%p)", this);
-    bool requiresReconfigure = false;
     if (mReformatBufferProvider != NULL) {
         delete mReformatBufferProvider;
         mReformatBufferProvider = NULL;
-        requiresReconfigure = true;
-    }
-    if (mPostDownmixReformatBufferProvider != NULL) {
-        delete mPostDownmixReformatBufferProvider;
-        mPostDownmixReformatBufferProvider = NULL;
-        requiresReconfigure = true;
-    }
-    if (requiresReconfigure) {
         reconfigureBufferProviders();
     }
 }
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index fb1e80c..879b6c9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -1466,14 +1466,14 @@ void AudioFlinger::EffectChain::process_l()
     }
 
     size_t size = mEffects.size();
-    for (size_t i = 0; i < size; i++) {
-        mEffects[i]->updateState();
-    }
     if (doProcess) {
         for (size_t i = 0; i < size; i++) {
             mEffects[i]->process();
         }
     }
+    for (size_t i = 0; i < size; i++) {
+        mEffects[i]->updateState();
+    }
 }
 
 // addEffect_l() must be called with PlaybackThread::mLock held
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 8c59282..1f60924 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -13,6 +13,25 @@
 ** 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.
+**
+** This file was modified by DTS, Inc. The portions of the
+** code that are surrounded by "DTS..." are copyrighted and
+** licensed separately, as follows:
+**
+**  (C) 2015 DTS, Inc.
+**
+** 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.
+**
 */
 
 
@@ -72,6 +91,9 @@
 #include <cpustats/ThreadCpuUsage.h>
 #endif
 
+#ifdef SRS_PROCESSING
+#include "postpro_patch.h"
+#endif
 // ----------------------------------------------------------------------------
 
 // Note: the following macro is used for extremely verbose logging message.  In
@@ -2727,6 +2749,19 @@ bool AudioFlinger::PlaybackThread::threadLoop()
     const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid()));
 
     acquireWakeLock();
+#ifdef SRS_PROCESSING
+    String8 bt_param = String8("bluetooth_enabled=0");
+    POSTPRO_PATCH_PARAMS_SET(bt_param);
+    if (mType == MIXER) {
+        POSTPRO_PATCH_OUTPROC_PLAY_INIT(this, myName);
+    } else if (mType == OFFLOAD) {
+        POSTPRO_PATCH_OUTPROC_DIRECT_INIT(this, myName);
+        POSTPRO_PATCH_OUTPROC_PLAY_ROUTE_BY_VALUE(this, mOutDevice);
+    } else if (mType == DIRECT) {
+        POSTPRO_PATCH_OUTPROC_DIRECT_INIT(this, myName);
+        POSTPRO_PATCH_OUTPROC_PLAY_ROUTE_BY_VALUE(this, mOutDevice);
+    }
+#endif
 
     // mNBLogWriter->log can only be called while thread mutex mLock is held.
     // So if you need to log when mutex is unlocked, set logString to a non-NULL string,
@@ -2918,7 +2953,13 @@ bool AudioFlinger::PlaybackThread::threadLoop()
                 effectChains[i]->process_l();
             }
         }
-
+#ifdef SRS_PROCESSING
+            // Offload thread
+            if (mType == OFFLOAD) {
+                char buffer[2];
+                POSTPRO_PATCH_OUTPROC_DIRECT_SAMPLES(this, AUDIO_FORMAT_PCM_16_BIT, (int16_t *) buffer, 2, 48000, 2);
+            }
+#endif
         // Only if the Effects buffer is enabled and there is data in the
         // Effects buffer (buffer valid), we need to
         // copy into the sink buffer.
@@ -2936,6 +2977,11 @@ bool AudioFlinger::PlaybackThread::threadLoop()
             // mSleepTimeUs == 0 means we must write to audio hardware
             if (mSleepTimeUs == 0) {
                 ssize_t ret = 0;
+#ifdef SRS_PROCESSING
+                if (mType == MIXER && mMixerStatus == MIXER_TRACKS_READY) {
+                    POSTPRO_PATCH_OUTPROC_PLAY_SAMPLES(this, mFormat, mSinkBuffer, mSinkBufferSize, mSampleRate, mChannelCount);
+                }
+#endif
                 if (mBytesRemaining) {
                     ret = threadLoop_write();
                     if (ret < 0) {
@@ -3032,7 +3078,15 @@ bool AudioFlinger::PlaybackThread::threadLoop()
         threadLoop_standby();
         mStandby = true;
     }
-
+#ifdef SRS_PROCESSING
+    if (mType == MIXER) {
+        POSTPRO_PATCH_OUTPROC_PLAY_EXIT(this, myName);
+    } else if (mType == OFFLOAD) {
+        POSTPRO_PATCH_OUTPROC_DIRECT_EXIT(this, myName);
+    } else if (mType == DIRECT) {
+        POSTPRO_PATCH_OUTPROC_DIRECT_EXIT(this, myName);
+    }
+#endif
     releaseWakeLock();
     mWakeLockUids.clear();
     mActiveTracksGeneration++;
@@ -3126,6 +3180,10 @@ status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_pat
         type |= patch->sinks[i].ext.device.type;
     }
 
+#ifdef SRS_PROCESSING
+    POSTPRO_PATCH_OUTPROC_PLAY_ROUTE_BY_VALUE(this, type);
+#endif
+
 #ifdef ADD_BATTERY_DATA
     // when changing the audio output device, call addBatteryData to notify
     // the change
@@ -4281,6 +4339,9 @@ bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePa
 
     AudioParameter param = AudioParameter(keyValuePair);
     int value;
+#ifdef SRS_PROCESSING
+        POSTPRO_PATCH_OUTPROC_PLAY_ROUTE(this, param, value);
+#endif
     if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
         reconfig = true;
     }
@@ -4821,6 +4882,7 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& key
 
     AudioParameter param = AudioParameter(keyValuePair);
     int value;
+
     if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
         // forward device change to effects that have requested to be
         // aware of attached audio device.
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 6182364..48ff77d 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -13,6 +13,24 @@
 ** 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.
+**
+** This file was modified by DTS, Inc. The portions of the
+** code that are surrounded by "DTS..." are copyrighted and
+** licensed separately, as follows:
+**
+**  (C) 2015 DTS, Inc.
+**
+** 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 INCLUDING_FROM_AUDIOFLINGER_H
@@ -537,7 +555,7 @@ public:
 
                 void        setMasterVolume(float value);
                 void        setMasterMute(bool muted);
-
+                void        setPostPro();
                 void        setStreamVolume(audio_stream_type_t stream, float value);
                 void        setStreamMute(audio_stream_type_t stream, bool muted);