diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index 22199fa..85f44f0 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -77,6 +77,32 @@
     return CameraBaseT::connect(cameraId, clientPackageName, clientUid);
 }
 
+status_t Camera::connectLegacy(int cameraId, int halVersion,
+        const String16& clientPackageName,
+        int clientUid,
+        sp<Camera>& camera)
+{
+    ALOGV("%s: connect legacy camera device", __FUNCTION__);
+    sp<Camera> c = new Camera(cameraId);
+    sp<ICameraClient> cl = c;
+    status_t status = NO_ERROR;
+    const sp<ICameraService>& cs = CameraBaseT::getCameraService();
+
+    if (cs != 0) {
+        status = cs.get()->connectLegacy(cl, cameraId, halVersion, clientPackageName,
+                                        clientUid, /*out*/c->mCamera);
+    }
+    if (status == OK && c->mCamera != 0) {
+        c->mCamera->asBinder()->linkToDeath(c);
+        c->mStatus = NO_ERROR;
+        camera = c;
+    } else {
+        ALOGW("An error occurred while connecting to camera: %d", cameraId);
+        c.clear();
+    }
+    return status;
+}
+
 status_t Camera::reconnect()
 {
     ALOGV("reconnect");
diff --git a/camera/ICameraService.cpp b/camera/ICameraService.cpp
index 79c33f9..5485205 100644
--- a/camera/ICameraService.cpp
+++ b/camera/ICameraService.cpp
@@ -186,6 +186,29 @@
         return status;
     }
 
+    // connect to camera service (android.hardware.Camera)
+    virtual status_t connectLegacy(const sp<ICameraClient>& cameraClient, int cameraId,
+                             int halVersion,
+                             const String16 &clientPackageName, int clientUid,
+                             /*out*/sp<ICamera>& device)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(ICameraService::getInterfaceDescriptor());
+        data.writeStrongBinder(cameraClient->asBinder());
+        data.writeInt32(cameraId);
+        data.writeInt32(halVersion);
+        data.writeString16(clientPackageName);
+        data.writeInt32(clientUid);
+        remote()->transact(BnCameraService::CONNECT_LEGACY, data, &reply);
+
+        if (readExceptionCode(reply)) return -EPROTO;
+        status_t status = reply.readInt32();
+        if (reply.readInt32() != 0) {
+            device = interface_cast<ICamera>(reply.readStrongBinder());
+        }
+        return status;
+    }
+
     // connect to camera service (pro client)
     virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb, int cameraId,
                                 const String16 &clientPackageName, int clientUid,
@@ -446,6 +469,27 @@
             reply->writeInt32(supportsCameraApi(cameraId, apiVersion));
             return NO_ERROR;
         } break;
+        case CONNECT_LEGACY: {
+            CHECK_INTERFACE(ICameraService, data, reply);
+            sp<ICameraClient> cameraClient =
+                    interface_cast<ICameraClient>(data.readStrongBinder());
+            int32_t cameraId = data.readInt32();
+            int32_t halVersion = data.readInt32();
+            const String16 clientName = data.readString16();
+            int32_t clientUid = data.readInt32();
+            sp<ICamera> camera;
+            status_t status = connectLegacy(cameraClient, cameraId, halVersion,
+                    clientName, clientUid, /*out*/camera);
+            reply->writeNoException();
+            reply->writeInt32(status);
+            if (camera != NULL) {
+                reply->writeInt32(1);
+                reply->writeStrongBinder(camera->asBinder());
+            } else {
+                reply->writeInt32(0);
+            }
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/cmds/screenrecord/Overlay.cpp b/cmds/screenrecord/Overlay.cpp
index c2a8f1b..af84069 100644
--- a/cmds/screenrecord/Overlay.cpp
+++ b/cmds/screenrecord/Overlay.cpp
@@ -14,6 +14,10 @@
  * limitations under the License.
  */
 
+#include <assert.h>
+#include <inttypes.h>
+#include <stdlib.h>
+
 #define LOG_TAG "ScreenRecord"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
@@ -27,9 +31,6 @@
 #include <GLES2/gl2.h>
 #include <GLES2/gl2ext.h>
 
-#include <stdlib.h>
-#include <assert.h>
-
 #include "screenrecord.h"
 #include "Overlay.h"
 #include "TextRenderer.h"
@@ -235,7 +236,7 @@
 
     char textBuf[64];
     getTimeString_l(monotonicNsec, textBuf, sizeof(textBuf));
-    String8 timeStr(String8::format("%s f=%lld (%zd)",
+    String8 timeStr(String8::format("%s f=%" PRId64 " (%zd)",
             textBuf, frameNumber, mTotalDroppedFrames));
     mTextRenderer.drawString(mTexProgram, Program::kIdentity, 0, 0, timeStr);
 
diff --git a/cmds/screenrecord/TextRenderer.cpp b/cmds/screenrecord/TextRenderer.cpp
index 784055c..6a9176b 100644
--- a/cmds/screenrecord/TextRenderer.cpp
+++ b/cmds/screenrecord/TextRenderer.cpp
@@ -353,6 +353,6 @@
         }
     }
 
-    ALOGV("goodPos=%d for str='%s'", goodPos, str);
+    ALOGV("goodPos=%zu for str='%s'", goodPos, str);
     return const_cast<char*>(str + goodPos);
 }
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 02ed53a..02df1d2 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -14,6 +14,19 @@
  * limitations under the License.
  */
 
+#include <assert.h>
+#include <ctype.h>
+#include <fcntl.h>
+#include <inttypes.h>
+#include <getopt.h>
+#include <signal.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/wait.h>
+#include <termios.h>
+#include <unistd.h>
+
 #define LOG_TAG "ScreenRecord"
 #define ATRACE_TAG ATRACE_TAG_GRAPHICS
 //#define LOG_NDEBUG 0
@@ -36,18 +49,6 @@
 #include <media/stagefright/MediaMuxer.h>
 #include <media/ICrypto.h>
 
-#include <stdlib.h>
-#include <unistd.h>
-#include <string.h>
-#include <stdio.h>
-#include <ctype.h>
-#include <fcntl.h>
-#include <signal.h>
-#include <getopt.h>
-#include <sys/wait.h>
-#include <termios.h>
-#include <assert.h>
-
 #include "screenrecord.h"
 #include "Overlay.h"
 #include "FrameOutput.h"
@@ -354,7 +355,7 @@
         case NO_ERROR:
             // got a buffer
             if ((flags & MediaCodec::BUFFER_FLAG_CODECCONFIG) != 0) {
-                ALOGV("Got codec config buffer (%u bytes)", size);
+                ALOGV("Got codec config buffer (%zu bytes)", size);
                 if (muxer != NULL) {
                     // ignore this -- we passed the CSD into MediaMuxer when
                     // we got the format change notification
@@ -362,7 +363,7 @@
                 }
             }
             if (size != 0) {
-                ALOGV("Got data in buffer %d, size=%d, pts=%lld",
+                ALOGV("Got data in buffer %zu, size=%zu, pts=%" PRId64,
                         bufIndex, size, ptsUsec);
 
                 { // scope
@@ -473,7 +474,7 @@
 
     ALOGV("Encoder stopping (req=%d)", gStopRequested);
     if (gVerbose) {
-        printf("Encoder stopping; recorded %u frames in %lld seconds\n",
+        printf("Encoder stopping; recorded %u frames in %" PRId64 " seconds\n",
                 debugNumFrames, nanoseconds_to_seconds(
                         systemTime(CLOCK_MONOTONIC) - startWhenNsec));
     }
diff --git a/drm/drmserver/DrmManagerService.cpp b/drm/drmserver/DrmManagerService.cpp
index 2b71904..63341e0 100644
--- a/drm/drmserver/DrmManagerService.cpp
+++ b/drm/drmserver/DrmManagerService.cpp
@@ -34,7 +34,18 @@
 static Vector<uid_t> trustedUids;
 
 static bool isProtectedCallAllowed() {
-    return true;
+    // TODO
+    // Following implementation is just for reference.
+    // Each OEM manufacturer should implement/replace with their own solutions.
+    IPCThreadState* ipcState = IPCThreadState::self();
+    uid_t uid = ipcState->getCallingUid();
+
+    for (unsigned int i = 0; i < trustedUids.size(); ++i) {
+        if (trustedUids[i] == uid) {
+            return true;
+        }
+    }
+    return false;
 }
 
 void DrmManagerService::instantiate() {
diff --git a/include/camera/Camera.h b/include/camera/Camera.h
index 79682b8..2b60842 100644
--- a/include/camera/Camera.h
+++ b/include/camera/Camera.h
@@ -74,6 +74,10 @@
                                 const String16& clientPackageName,
                                 int clientUid);
 
+    static  status_t  connectLegacy(int cameraId, int halVersion,
+                                     const String16& clientPackageName,
+                                     int clientUid, sp<Camera>& camera);
+
             virtual     ~Camera();
 
             status_t    reconnect();
diff --git a/include/camera/ICameraService.h b/include/camera/ICameraService.h
index d7e5dac..f7f06bb 100644
--- a/include/camera/ICameraService.h
+++ b/include/camera/ICameraService.h
@@ -52,6 +52,7 @@
         GET_CAMERA_VENDOR_TAG_DESCRIPTOR,
         GET_LEGACY_PARAMETERS,
         SUPPORTS_CAMERA_API,
+        CONNECT_LEGACY,
     };
 
     enum {
@@ -63,6 +64,10 @@
         API_VERSION_2 = 2,
     };
 
+    enum {
+        CAMERA_HAL_API_VERSION_UNSPECIFIED = -1
+      };
+
 public:
     DECLARE_META_INTERFACE(CameraService);
 
@@ -125,6 +130,18 @@
      */
     virtual status_t supportsCameraApi(
             int cameraId, int apiVersion) = 0;
+
+    /**
+     * Connect the device as a legacy device for a given HAL version.
+     * For halVersion, use CAMERA_API_DEVICE_VERSION_* for a particular
+     * version, or CAMERA_HAL_API_VERSION_UNSPECIFIED for a service-selected version.
+     */
+    virtual status_t connectLegacy(const sp<ICameraClient>& cameraClient,
+            int cameraId, int halVersion,
+            const String16& clientPackageName,
+            int clientUid,
+            /*out*/
+            sp<ICamera>& device) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index a4722cb..c89ceaa 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -99,6 +99,8 @@
     // to be non-zero if status == NO_ERROR
     static status_t getOutputSamplingRate(uint32_t* samplingRate,
             audio_stream_type_t stream);
+    static status_t getOutputSamplingRateForAttr(uint32_t* samplingRate,
+                const audio_attributes_t *attr);
     static status_t getOutputFrameCount(size_t* frameCount,
             audio_stream_type_t stream);
     static status_t getOutputLatency(uint32_t* latency,
@@ -212,7 +214,12 @@
                                         audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
                                         audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                         const audio_offload_info_t *offloadInfo = NULL);
-
+    static audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+                                        uint32_t samplingRate = 0,
+                                        audio_format_t format = AUDIO_FORMAT_DEFAULT,
+                                        audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO,
+                                        audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                        const audio_offload_info_t *offloadInfo = NULL);
     static status_t startOutput(audio_io_handle_t output,
                                 audio_stream_type_t stream,
                                 int session);
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 79db323..3492520 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -253,7 +253,8 @@
                             transfer_type transferType = TRANSFER_DEFAULT,
                             const audio_offload_info_t *offloadInfo = NULL,
                             int uid = -1,
-                            pid_t pid = -1);
+                            pid_t pid = -1,
+                            audio_attributes_t* pAttributes = NULL);
 
     /* Result of constructing the AudioTrack. This must be checked for successful initialization
      * before using any AudioTrack API (except for set()), because using
@@ -586,6 +587,11 @@
                         AudioTrack(const AudioTrack& other);
             AudioTrack& operator = (const AudioTrack& other);
 
+            void        setAttributesFromStreamType(audio_stream_type_t streamType);
+            void        setStreamTypeFromAttributes(audio_attributes_t& aa);
+    /* paa is guaranteed non-NULL */
+            bool        isValidAttributes(const audio_attributes_t *paa);
+
     /* a small internal class to handle the callback */
     class AudioTrackThread : public Thread
     {
@@ -626,6 +632,8 @@
             nsecs_t processAudioBuffer();
 
             bool     isOffloaded() const;
+            bool     isDirect() const;
+            bool     isOffloadedOrDirect() const;
 
             // caller must hold lock on mLock for all _l methods
 
@@ -642,6 +650,13 @@
             bool     isOffloaded_l() const
                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
 
+            bool     isOffloadedOrDirect_l() const
+                { return (mFlags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|
+                                                AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }
+
+            bool     isDirect_l() const
+                { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
+
     // Next 4 fields may be changed if IAudioTrack is re-created, but always != 0
     sp<IAudioTrack>         mAudioTrack;
     sp<IMemory>             mCblkMemory;
@@ -667,6 +682,7 @@
     transfer_type           mTransfer;
     audio_offload_info_t    mOffloadInfoCopy;
     const audio_offload_info_t* mOffloadInfo;
+    audio_attributes_t      mAttributes;
 
     // mFrameSize is equal to mFrameSizeAF for non-PCM or 16-bit PCM data.  For 8-bit PCM data, it's
     // twice as large as mFrameSize because data is expanded to 16-bit before it's stored in buffer.
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index c742810..fc8be20 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -50,6 +50,7 @@
         TRACK_TIMED   = 1,  // client requests a TimedAudioTrack
         TRACK_FAST    = 2,  // client requests a fast AudioTrack or AudioRecord
         TRACK_OFFLOAD = 4,  // client requests offload to hw codec
+        TRACK_DIRECT = 8,   // client requests a direct output
     };
     typedef uint32_t track_flags_t;
 
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index d422aa3..959e4c3 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -56,6 +56,12 @@
                                         audio_channel_mask_t channelMask = 0,
                                         audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
                                         const audio_offload_info_t *offloadInfo = NULL) = 0;
+    virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+                                            uint32_t samplingRate = 0,
+                                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
+                                            audio_channel_mask_t channelMask = 0,
+                                            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                            const audio_offload_info_t *offloadInfo = NULL) = 0;
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
                                  int session = 0) = 0;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 50b444a..f865d38 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -18,7 +18,9 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "AudioRecord"
 
+#include <inttypes.h>
 #include <sys/resource.h>
+
 #include <binder/IPCThreadState.h>
 #include <media/AudioRecord.h>
 #include <utils/Log.h>
@@ -468,7 +470,7 @@
         if (frameCount == 0) {
             frameCount = minFrameCount;
         } else if (frameCount < minFrameCount) {
-            ALOGE("frameCount %u < minFrameCount %u", frameCount, minFrameCount);
+            ALOGE("frameCount %zu < minFrameCount %zu", frameCount, minFrameCount);
             return BAD_VALUE;
         }
 
@@ -555,17 +557,17 @@
     mCblk = cblk;
     // note that temp is the (possibly revised) value of frameCount
     if (temp < frameCount || (frameCount == 0 && temp == 0)) {
-        ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+        ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
     }
     frameCount = temp;
 
     mAwaitBoost = false;
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
         if (trackFlags & IAudioFlinger::TRACK_FAST) {
-            ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %u", frameCount);
+            ALOGV("AUDIO_INPUT_FLAG_FAST successful; frameCount %zu", frameCount);
             mAwaitBoost = true;
         } else {
-            ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
+            ALOGV("AUDIO_INPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
             // once denied, do not request again if IAudioRecord is re-created
             mFlags = (audio_input_flags_t) (mFlags & ~AUDIO_INPUT_FLAG_FAST);
         }
@@ -740,7 +742,7 @@
     if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
         // sanity-check. user is most-likely passing an error code, and it would
         // make the return value ambiguous (actualSize vs error).
-        ALOGE("AudioRecord::read(buffer=%p, size=%u (%d)", buffer, userSize, userSize);
+        ALOGE("AudioRecord::read(buffer=%p, size=%zu (%zu)", buffer, userSize, userSize);
         return BAD_VALUE;
     }
 
@@ -921,10 +923,10 @@
         size_t nonContig;
         status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
         LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
-                "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+                "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
         requested = &ClientProxy::kNonBlocking;
         size_t avail = audioBuffer.frameCount + nonContig;
-        ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+        ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
                 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
         if (err != NO_ERROR) {
             if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR) {
@@ -952,8 +954,8 @@
 
         // Sanity check on returned size
         if (ssize_t(readSize) < 0 || readSize > reqSize) {
-            ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
-                    reqSize, (int) readSize);
+            ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+                    reqSize, ssize_t(readSize));
             return NS_NEVER;
         }
 
@@ -1092,7 +1094,7 @@
         ns = 1000000000LL;
         // fall through
     default:
-        LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+        LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
         pauseInternal(ns);
         return true;
     }
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 15b32ff..a47d45c 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -245,6 +245,19 @@
     return getSamplingRate(output, samplingRate);
 }
 
+status_t AudioSystem::getOutputSamplingRateForAttr(uint32_t* samplingRate,
+        const audio_attributes_t *attr)
+{
+    if (attr == NULL) {
+        return BAD_VALUE;
+    }
+    audio_io_handle_t output = getOutputForAttr(attr);
+    if (output == 0) {
+        return PERMISSION_DENIED;
+    }
+    return getSamplingRate(output, samplingRate);
+}
+
 status_t AudioSystem::getSamplingRate(audio_io_handle_t output,
                                       uint32_t* samplingRate)
 {
@@ -310,7 +323,7 @@
         return BAD_VALUE;
     }
 
-    ALOGV("getFrameCount() output %d, frameCount %d", output, *frameCount);
+    ALOGV("getFrameCount() output %d, frameCount %zu", output, *frameCount);
 
     return NO_ERROR;
 }
@@ -476,7 +489,7 @@
 
         OutputDescriptor *outputDesc =  new OutputDescriptor(*desc);
         gOutputs.add(ioHandle, outputDesc);
-        ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %u "
+        ALOGV("ioConfigChanged() new output samplingRate %u, format %#x channel mask %#x frameCount %zu "
                 "latency %d",
                 outputDesc->samplingRate, outputDesc->format, outputDesc->channelMask,
                 outputDesc->frameCount, outputDesc->latency);
@@ -501,7 +514,7 @@
         desc = (const OutputDescriptor *)param2;
 
         ALOGV("ioConfigChanged() new config for output %d samplingRate %u, format %#x channel mask %#x "
-                "frameCount %d latency %d",
+                "frameCount %zu latency %d",
                 ioHandle, desc->samplingRate, desc->format,
                 desc->channelMask, desc->frameCount, desc->latency);
         OutputDescriptor *outputDesc = gOutputs.valueAt(index);
@@ -633,6 +646,19 @@
     return aps->getOutput(stream, samplingRate, format, channelMask, flags, offloadInfo);
 }
 
+audio_io_handle_t AudioSystem::getOutputForAttr(const audio_attributes_t *attr,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_output_flags_t flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    if (attr == NULL) return 0;
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return 0;
+    return aps->getOutputForAttr(attr, samplingRate, format, channelMask, flags, offloadInfo);
+}
+
 status_t AudioSystem::startOutput(audio_io_handle_t output,
                                   audio_stream_type_t stream,
                                   int session)
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index e6827ee..898d58d 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -15,12 +15,13 @@
 ** limitations under the License.
 */
 
-
 //#define LOG_NDEBUG 0
 #define LOG_TAG "AudioTrack"
 
+#include <inttypes.h>
 #include <math.h>
 #include <sys/resource.h>
+
 #include <audio_utils/primitives.h>
 #include <binder/IPCThreadState.h>
 #include <media/AudioTrack.h>
@@ -89,7 +90,7 @@
                 streamType, sampleRate);
         return BAD_VALUE;
     }
-    ALOGV("getMinFrameCount=%d: afFrameCount=%d, minBufCount=%d, afSampleRate=%d, afLatency=%d",
+    ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d",
             *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency);
     return NO_ERROR;
 }
@@ -103,6 +104,10 @@
       mPreviousSchedulingGroup(SP_DEFAULT),
       mPausedPosition(0)
 {
+    mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
+    mAttributes.usage = AUDIO_USAGE_UNKNOWN;
+    mAttributes.flags = 0x0;
+    strcpy(mAttributes.tags, "");
 }
 
 AudioTrack::AudioTrack(
@@ -129,7 +134,7 @@
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
-            offloadInfo, uid, pid);
+            offloadInfo, uid, pid, NULL /*no audio attributes*/);
 }
 
 AudioTrack::AudioTrack(
@@ -156,7 +161,7 @@
     mStatus = set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
             sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
-            uid, pid);
+            uid, pid, NULL /*no audio attributes*/);
 }
 
 AudioTrack::~AudioTrack()
@@ -199,7 +204,8 @@
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
         int uid,
-        pid_t pid)
+        pid_t pid,
+        audio_attributes_t* pAttributes)
 {
     ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
           "flags #%x, notificationFrames %u, sessionId %d, transferType %d",
@@ -245,7 +251,7 @@
     ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(),
             sharedBuffer->size());
 
-    ALOGV("set() streamType %d frameCount %u flags %04x", streamType, frameCount, flags);
+    ALOGV("set() streamType %d frameCount %zu flags %04x", streamType, frameCount, flags);
 
     AutoMutex lock(mLock);
 
@@ -259,18 +265,33 @@
     if (streamType == AUDIO_STREAM_DEFAULT) {
         streamType = AUDIO_STREAM_MUSIC;
     }
-    if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
-        ALOGE("Invalid stream type %d", streamType);
-        return BAD_VALUE;
+
+    if (pAttributes == NULL) {
+        if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+            ALOGE("Invalid stream type %d", streamType);
+            return BAD_VALUE;
+        }
+        setAttributesFromStreamType(streamType);
+        mStreamType = streamType;
+    } else {
+        if (!isValidAttributes(pAttributes)) {
+            ALOGE("Invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+                pAttributes->usage, pAttributes->content_type, pAttributes->flags,
+                pAttributes->tags);
+        }
+        // stream type shouldn't be looked at, this track has audio attributes
+        memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
+        setStreamTypeFromAttributes(mAttributes);
+        ALOGV("Building AudioTrack with attributes: usage=%d content=%d flags=0x%x tags=[%s]",
+                mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
     }
-    mStreamType = streamType;
 
     status_t status;
     if (sampleRate == 0) {
-        status = AudioSystem::getOutputSamplingRate(&sampleRate, streamType);
+        status = AudioSystem::getOutputSamplingRateForAttr(&sampleRate, &mAttributes);
         if (status != NO_ERROR) {
             ALOGE("Could not get output sample rate for stream type %d; status %d",
-                    streamType, status);
+                    mStreamType, status);
             return status;
         }
     }
@@ -314,7 +335,7 @@
                 ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
     }
     // only allow deep buffering for music stream type
-    if (streamType != AUDIO_STREAM_MUSIC) {
+    if (mStreamType != AUDIO_STREAM_MUSIC) {
         flags = (audio_output_flags_t)(flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
     }
 
@@ -615,12 +636,12 @@
 
 status_t AudioTrack::setSampleRate(uint32_t rate)
 {
-    if (mIsTimed || isOffloaded()) {
+    if (mIsTimed || isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
 
     uint32_t afSamplingRate;
-    if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
+    if (AudioSystem::getOutputSamplingRateForAttr(&afSamplingRate, &mAttributes) != NO_ERROR) {
         return NO_INIT;
     }
     // Resampler implementation limits input sampling rate to 2 x output sampling rate.
@@ -646,7 +667,7 @@
     // sample rate can be updated during playback by the offloaded decoder so we need to
     // query the HAL and update if needed.
 // FIXME use Proxy return channel to update the rate from server and avoid polling here
-    if (isOffloaded_l()) {
+    if (isOffloadedOrDirect_l()) {
         if (mOutput != AUDIO_IO_HANDLE_NONE) {
             uint32_t sampleRate = 0;
             status_t status = AudioSystem::getSamplingRate(mOutput, &sampleRate);
@@ -660,7 +681,7 @@
 
 status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount)
 {
-    if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+    if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
 
@@ -694,7 +715,7 @@
 status_t AudioTrack::setMarkerPosition(uint32_t marker)
 {
     // The only purpose of setting marker position is to get a callback
-    if (mCbf == NULL || isOffloaded()) {
+    if (mCbf == NULL || isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
 
@@ -707,7 +728,7 @@
 
 status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
 {
-    if (isOffloaded()) {
+    if (isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
     if (marker == NULL) {
@@ -723,7 +744,7 @@
 status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
 {
     // The only purpose of setting position update period is to get a callback
-    if (mCbf == NULL || isOffloaded()) {
+    if (mCbf == NULL || isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
 
@@ -736,7 +757,7 @@
 
 status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
 {
-    if (isOffloaded()) {
+    if (isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
     if (updatePeriod == NULL) {
@@ -751,7 +772,7 @@
 
 status_t AudioTrack::setPosition(uint32_t position)
 {
-    if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+    if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
     if (position > mFrameCount) {
@@ -784,10 +805,10 @@
     }
 
     AutoMutex lock(mLock);
-    if (isOffloaded_l()) {
+    if (isOffloadedOrDirect_l()) {
         uint32_t dspFrames = 0;
 
-        if ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING)) {
+        if (isOffloaded_l() && ((mState == STATE_PAUSED) || (mState == STATE_PAUSED_STOPPING))) {
             ALOGV("getPosition called in paused state, return cached position %u", mPausedPosition);
             *position = mPausedPosition;
             return NO_ERROR;
@@ -822,7 +843,7 @@
 
 status_t AudioTrack::reload()
 {
-    if (mSharedBuffer == 0 || mIsTimed || isOffloaded()) {
+    if (mSharedBuffer == 0 || mIsTimed || isOffloadedOrDirect()) {
         return INVALID_OPERATION;
     }
 
@@ -867,12 +888,12 @@
         return NO_INIT;
     }
 
-    audio_io_handle_t output = AudioSystem::getOutput(mStreamType, mSampleRate, mFormat,
+    audio_io_handle_t output = AudioSystem::getOutputForAttr(&mAttributes, mSampleRate, mFormat,
             mChannelMask, mFlags, mOffloadInfo);
     if (output == AUDIO_IO_HANDLE_NONE) {
-        ALOGE("Could not get audio output for stream type %d, sample rate %u, format %#x, "
-              "channel mask %#x, flags %#x",
-              mStreamType, mSampleRate, mFormat, mChannelMask, mFlags);
+        ALOGE("Could not get audio output for stream type %d, usage %d, sample rate %u, format %#x,"
+              " channel mask %#x, flags %#x",
+              mStreamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
         return BAD_VALUE;
     }
     {
@@ -973,14 +994,14 @@
 
         // Ensure that buffer depth covers at least audio hardware latency
         uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
-        ALOGV("afFrameCount=%d, minBufCount=%d, afSampleRate=%u, afLatency=%d",
+        ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d",
                 afFrameCount, minBufCount, afSampleRate, afLatency);
         if (minBufCount <= nBuffering) {
             minBufCount = nBuffering;
         }
 
         size_t minFrameCount = (afFrameCount*mSampleRate*minBufCount)/afSampleRate;
-        ALOGV("minFrameCount: %u, afFrameCount=%d, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
+        ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u"
                 ", afLatency=%d",
                 minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency);
 
@@ -988,7 +1009,7 @@
             frameCount = minFrameCount;
         } else if (frameCount < minFrameCount) {
             // not ALOGW because it happens all the time when playing key clicks over A2DP
-            ALOGV("Minimum buffer size corrected from %d to %d",
+            ALOGV("Minimum buffer size corrected from %zu to %zu",
                      frameCount, minFrameCount);
             frameCount = minFrameCount;
         }
@@ -1018,6 +1039,10 @@
         trackFlags |= IAudioFlinger::TRACK_OFFLOAD;
     }
 
+    if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        trackFlags |= IAudioFlinger::TRACK_DIRECT;
+    }
+
     size_t temp = frameCount;   // temp may be replaced by a revised value of frameCount,
                                 // but we will still need the original value also
     sp<IAudioTrack> track = audioFlinger->createTrack(mStreamType,
@@ -1071,14 +1096,14 @@
         // In current design, AudioTrack client checks and ensures frame count validity before
         // passing it to AudioFlinger so AudioFlinger should not return a different value except
         // for fast track as it uses a special method of assigning frame count.
-        ALOGW("Requested frameCount %u but received frameCount %u", frameCount, temp);
+        ALOGW("Requested frameCount %zu but received frameCount %zu", frameCount, temp);
     }
     frameCount = temp;
 
     mAwaitBoost = false;
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         if (trackFlags & IAudioFlinger::TRACK_FAST) {
-            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %u", frameCount);
+            ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount);
             mAwaitBoost = true;
             if (mSharedBuffer == 0) {
                 // Theoretically double-buffering is not required for fast tracks,
@@ -1089,7 +1114,7 @@
                 }
             }
         } else {
-            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %u", frameCount);
+            ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount);
             // once denied, do not request again if IAudioTrack is re-created
             mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
             if (mSharedBuffer == 0) {
@@ -1109,6 +1134,16 @@
             //return NO_INIT;
         }
     }
+    if (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        if (trackFlags & IAudioFlinger::TRACK_DIRECT) {
+            ALOGV("AUDIO_OUTPUT_FLAG_DIRECT successful");
+        } else {
+            ALOGW("AUDIO_OUTPUT_FLAG_DIRECT denied by server");
+            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_DIRECT);
+            // FIXME This is a warning, not an error, so don't return error status
+            //return NO_INIT;
+        }
+    }
 
     // We retain a copy of the I/O handle, but don't own the reference
     mOutput = output;
@@ -1304,6 +1339,16 @@
         return INVALID_OPERATION;
     }
 
+    if (isDirect()) {
+        AutoMutex lock(mLock);
+        int32_t flags = android_atomic_and(
+                            ~(CBLK_UNDERRUN | CBLK_LOOP_CYCLE | CBLK_LOOP_FINAL | CBLK_BUFFER_END),
+                            &mCblk->mFlags);
+        if (flags & CBLK_INVALID) {
+            return DEAD_OBJECT;
+        }
+    }
+
     if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
         // Sanity-check: user is most-likely passing an error code, and it would
         // make the return value ambiguous (actualSize vs error).
@@ -1452,7 +1497,7 @@
         // for offloaded tracks restoreTrack_l() will just update the sequence and clear
         // AudioSystem cache. We should not exit here but after calling the callback so
         // that the upper layers can recreate the track
-        if (!isOffloaded_l() || (mSequence == mObservedSequence)) {
+        if (!isOffloadedOrDirect_l() || (mSequence == mObservedSequence)) {
             status_t status = restoreTrack_l("processAudioBuffer");
             mLock.unlock();
             // Run again immediately, but with a new IAudioTrack
@@ -1578,7 +1623,7 @@
         mObservedSequence = sequence;
         mCbf(EVENT_NEW_IAUDIOTRACK, mUserData, NULL);
         // for offloaded tracks, just wait for the upper layers to recreate the track
-        if (isOffloaded()) {
+        if (isOffloadedOrDirect()) {
             return NS_INACTIVE;
         }
     }
@@ -1636,10 +1681,10 @@
         size_t nonContig;
         status_t err = obtainBuffer(&audioBuffer, requested, NULL, &nonContig);
         LOG_ALWAYS_FATAL_IF((err != NO_ERROR) != (audioBuffer.frameCount == 0),
-                "obtainBuffer() err=%d frameCount=%u", err, audioBuffer.frameCount);
+                "obtainBuffer() err=%d frameCount=%zu", err, audioBuffer.frameCount);
         requested = &ClientProxy::kNonBlocking;
         size_t avail = audioBuffer.frameCount + nonContig;
-        ALOGV("obtainBuffer(%u) returned %u = %u + %u err %d",
+        ALOGV("obtainBuffer(%u) returned %zu = %zu + %zu err %d",
                 mRemainingFrames, avail, audioBuffer.frameCount, nonContig, err);
         if (err != NO_ERROR) {
             if (err == TIMED_OUT || err == WOULD_BLOCK || err == -EINTR ||
@@ -1674,8 +1719,8 @@
 
         // Sanity check on returned size
         if (ssize_t(writtenSize) < 0 || writtenSize > reqSize) {
-            ALOGE("EVENT_MORE_DATA requested %u bytes but callback returned %d bytes",
-                    reqSize, (int) writtenSize);
+            ALOGE("EVENT_MORE_DATA requested %zu bytes but callback returned %zd bytes",
+                    reqSize, ssize_t(writtenSize));
             return NS_NEVER;
         }
 
@@ -1736,7 +1781,7 @@
 status_t AudioTrack::restoreTrack_l(const char *from)
 {
     ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
-          isOffloaded_l() ? "Offloaded" : "PCM", from);
+          isOffloadedOrDirect_l() ? "Offloaded or Direct" : "PCM", from);
     ++mSequence;
     status_t result;
 
@@ -1744,7 +1789,7 @@
     // output parameters in createTrack_l()
     AudioSystem::clearAudioConfigCache();
 
-    if (isOffloaded_l()) {
+    if (isOffloadedOrDirect_l()) {
         // FIXME re-creation of offloaded tracks is not yet implemented
         return DEAD_OBJECT;
     }
@@ -1830,6 +1875,19 @@
     return isOffloaded_l();
 }
 
+bool AudioTrack::isDirect() const
+{
+    AutoMutex lock(mLock);
+    return isDirect_l();
+}
+
+bool AudioTrack::isOffloadedOrDirect() const
+{
+    AutoMutex lock(mLock);
+    return isOffloadedOrDirect_l();
+}
+
+
 status_t AudioTrack::dump(int fd, const Vector<String16>& args __unused) const
 {
 
@@ -1858,6 +1916,136 @@
     return mProxy->getUnderrunFrames();
 }
 
+void AudioTrack::setAttributesFromStreamType(audio_stream_type_t streamType) {
+    mAttributes.flags = 0x0;
+
+    switch(streamType) {
+    case AUDIO_STREAM_DEFAULT:
+    case AUDIO_STREAM_MUSIC:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+        mAttributes.usage = AUDIO_USAGE_MEDIA;
+        break;
+    case AUDIO_STREAM_VOICE_CALL:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+        mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+        break;
+    case AUDIO_STREAM_ENFORCED_AUDIBLE:
+        mAttributes.flags  |= AUDIO_FLAG_AUDIBILITY_ENFORCED;
+        // intended fall through, attributes in common with STREAM_SYSTEM
+    case AUDIO_STREAM_SYSTEM:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+        mAttributes.usage = AUDIO_USAGE_ASSISTANCE_SONIFICATION;
+        break;
+    case AUDIO_STREAM_RING:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+        mAttributes.usage = AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE;
+        break;
+    case AUDIO_STREAM_ALARM:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+        mAttributes.usage = AUDIO_USAGE_ALARM;
+        break;
+    case AUDIO_STREAM_NOTIFICATION:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+        mAttributes.usage = AUDIO_USAGE_NOTIFICATION;
+        break;
+    case AUDIO_STREAM_BLUETOOTH_SCO:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+        mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION;
+        mAttributes.flags |= AUDIO_FLAG_SCO;
+        break;
+    case AUDIO_STREAM_DTMF:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SONIFICATION;
+        mAttributes.usage = AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING;
+        break;
+    case AUDIO_STREAM_TTS:
+        mAttributes.content_type = AUDIO_CONTENT_TYPE_SPEECH;
+        mAttributes.usage = AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY;
+        break;
+    default:
+        ALOGE("invalid stream type %d when converting to attributes", streamType);
+    }
+}
+
+void AudioTrack::setStreamTypeFromAttributes(audio_attributes_t& aa) {
+    // flags to stream type mapping
+    if ((aa.flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+        mStreamType = AUDIO_STREAM_ENFORCED_AUDIBLE;
+        return;
+    }
+    if ((aa.flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+        mStreamType = AUDIO_STREAM_BLUETOOTH_SCO;
+        return;
+    }
+
+    // usage to stream type mapping
+    switch (aa.usage) {
+    case AUDIO_USAGE_MEDIA:
+    case AUDIO_USAGE_GAME:
+    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+    case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+        mStreamType = AUDIO_STREAM_MUSIC;
+        return;
+    case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+        mStreamType = AUDIO_STREAM_SYSTEM;
+        return;
+    case AUDIO_USAGE_VOICE_COMMUNICATION:
+        mStreamType = AUDIO_STREAM_VOICE_CALL;
+        return;
+
+    case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+        mStreamType = AUDIO_STREAM_DTMF;
+        return;
+
+    case AUDIO_USAGE_ALARM:
+        mStreamType = AUDIO_STREAM_ALARM;
+        return;
+    case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+        mStreamType = AUDIO_STREAM_RING;
+        return;
+
+    case AUDIO_USAGE_NOTIFICATION:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+    case AUDIO_USAGE_NOTIFICATION_EVENT:
+        mStreamType = AUDIO_STREAM_NOTIFICATION;
+        return;
+
+    case AUDIO_USAGE_UNKNOWN:
+    default:
+        mStreamType = AUDIO_STREAM_MUSIC;
+    }
+}
+
+bool AudioTrack::isValidAttributes(const audio_attributes_t *paa) {
+    // has flags that map to a strategy?
+    if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO)) != 0) {
+        return true;
+    }
+
+    // has known usage?
+    switch (paa->usage) {
+    case AUDIO_USAGE_UNKNOWN:
+    case AUDIO_USAGE_MEDIA:
+    case AUDIO_USAGE_VOICE_COMMUNICATION:
+    case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+    case AUDIO_USAGE_ALARM:
+    case AUDIO_USAGE_NOTIFICATION:
+    case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+    case AUDIO_USAGE_NOTIFICATION_EVENT:
+    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+    case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+    case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+    case AUDIO_USAGE_GAME:
+        break;
+    default:
+        return false;
+    }
+    return true;
+}
 // =========================================================================
 
 void AudioTrack::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
@@ -1918,7 +2106,7 @@
         ns = 1000000000LL;
         // fall through
     default:
-        LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %lld", ns);
+        LOG_ALWAYS_FATAL_IF(ns < 0, "processAudioBuffer() returned %" PRId64, ns);
         pauseInternal(ns);
         return true;
     }
diff --git a/media/libmedia/AudioTrackShared.cpp b/media/libmedia/AudioTrackShared.cpp
index 0dbfa62..eec025e 100644
--- a/media/libmedia/AudioTrackShared.cpp
+++ b/media/libmedia/AudioTrackShared.cpp
@@ -135,7 +135,7 @@
         // pipe should not be overfull
         if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
             if (mIsOut) {
-                ALOGE("Shared memory control block is corrupt (filled=%d, mFrameCount=%u); "
+                ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); "
                         "shutting down", filled, mFrameCount);
                 mIsShutdown = true;
                 status = NO_INIT;
@@ -338,7 +338,7 @@
     ssize_t filled = rear - front;
     // pipe should not be overfull
     if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
-        ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+        ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
         return 0;
     }
     return (size_t)filled;
@@ -555,7 +555,7 @@
     ssize_t filled = rear - front;
     // pipe should not already be overfull
     if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
-        ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+        ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
         mIsShutdown = true;
     }
     if (mIsShutdown) {
@@ -642,7 +642,7 @@
     }
     // FIXME AudioRecord wakeup needs to be optimized; it currently wakes up client every time
     if (!mIsOut || (mAvailToClient + stepCount >= minimum)) {
-        ALOGV("mAvailToClient=%u stepCount=%u minimum=%u", mAvailToClient, stepCount, minimum);
+        ALOGV("mAvailToClient=%zu stepCount=%zu minimum=%zu", mAvailToClient, stepCount, minimum);
         int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
         if (!(old & CBLK_FUTEX_WAKE)) {
             (void) syscall(__NR_futex, &cblk->mFutex,
@@ -675,7 +675,7 @@
     ssize_t filled = rear - cblk->u.mStreaming.mFront;
     // pipe should not already be overfull
     if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
-        ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
+        ALOGE("Shared memory control block is corrupt (filled=%zd); shutting down", filled);
         mIsShutdown = true;
         return 0;
     }
@@ -834,7 +834,7 @@
     size_t newPosition = position + stepCount;
     int32_t setFlags = 0;
     if (!(position <= newPosition && newPosition <= mFrameCount)) {
-        ALOGW("%s newPosition %u outside [%u, %u]", __func__, newPosition, position, mFrameCount);
+        ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position, mFrameCount);
         newPosition = mFrameCount;
     } else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
         if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
diff --git a/media/libmedia/CharacterEncodingDetector.cpp b/media/libmedia/CharacterEncodingDetector.cpp
index 4992798..7d1ddfd 100644
--- a/media/libmedia/CharacterEncodingDetector.cpp
+++ b/media/libmedia/CharacterEncodingDetector.cpp
@@ -112,7 +112,7 @@
         if (allprintable) {
             // since 'buf' is empty, ICU would return a UTF-8 matcher with low confidence, so
             // no need to even call it
-            ALOGV("all tags are printable, assuming ascii (%d)", strlen(buf));
+            ALOGV("all tags are printable, assuming ascii (%zu)", strlen(buf));
         } else {
             ucsdet_setText(csd, buf, strlen(buf), &status);
             int32_t matches;
@@ -267,11 +267,11 @@
     Vector<const UCharsetMatch*> matches;
     UErrorCode status = U_ZERO_ERROR;
 
-    ALOGV("%d matches", nummatches);
+    ALOGV("%zu matches", nummatches);
     for (size_t i = 0; i < nummatches; i++) {
         const char *encname = ucsdet_getName(ucma[i], &status);
         int confidence = ucsdet_getConfidence(ucma[i], &status);
-        ALOGV("%d: %s %d", i, encname, confidence);
+        ALOGV("%zu: %s %d", i, encname, confidence);
         matches.push_back(ucma[i]);
     }
 
@@ -287,7 +287,7 @@
         return matches[0];
     }
 
-    ALOGV("considering %d matches", num);
+    ALOGV("considering %zu matches", num);
 
     // keep track of how many "special" characters result when converting the input using each
     // encoding
@@ -315,7 +315,7 @@
             freqcoverage = frequent_ja_coverage;
         }
 
-        ALOGV("%d: %s %d", i, encname, confidence);
+        ALOGV("%zu: %s %d", i, encname, confidence);
         UConverter *conv = ucnv_open(encname, &status);
         const char *source = input;
         const char *sourceLimit = input + len;
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 77d131b..41a9065 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -64,7 +64,8 @@
     RELEASE_AUDIO_PATCH,
     LIST_AUDIO_PATCHES,
     SET_AUDIO_PORT_CONFIG,
-    REGISTER_CLIENT
+    REGISTER_CLIENT,
+    GET_OUTPUT_FOR_ATTR
 };
 
 class BpAudioPolicyService : public BpInterface<IAudioPolicyService>
@@ -155,6 +156,36 @@
         return static_cast <audio_io_handle_t> (reply.readInt32());
     }
 
+    virtual audio_io_handle_t getOutputForAttr(
+                                            const audio_attributes_t *attr,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            audio_output_flags_t flags,
+                                            const audio_offload_info_t *offloadInfo)
+        {
+            Parcel data, reply;
+            data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+            if (attr == NULL) {
+                ALOGE("Writing NULL audio attributes - shouldn't happen");
+                return (audio_io_handle_t) 0;
+            }
+            data.write(attr, sizeof(audio_attributes_t));
+            data.writeInt32(samplingRate);
+            data.writeInt32(static_cast <uint32_t>(format));
+            data.writeInt32(channelMask);
+            data.writeInt32(static_cast <uint32_t>(flags));
+            // hasOffloadInfo
+            if (offloadInfo == NULL) {
+                data.writeInt32(0);
+            } else {
+                data.writeInt32(1);
+                data.write(offloadInfo, sizeof(audio_offload_info_t));
+            }
+            remote()->transact(GET_OUTPUT_FOR_ATTR, data, &reply);
+            return static_cast <audio_io_handle_t> (reply.readInt32());
+        }
+
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
                                  int session)
@@ -614,6 +645,30 @@
             return NO_ERROR;
         } break;
 
+        case GET_OUTPUT_FOR_ATTR: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_attributes_t *attr = (audio_attributes_t *) calloc(1, sizeof(audio_attributes_t));
+            data.read(attr, sizeof(audio_attributes_t));
+            uint32_t samplingRate = data.readInt32();
+            audio_format_t format = (audio_format_t) data.readInt32();
+            audio_channel_mask_t channelMask = data.readInt32();
+            audio_output_flags_t flags =
+                    static_cast <audio_output_flags_t>(data.readInt32());
+            bool hasOffloadInfo = data.readInt32() != 0;
+            audio_offload_info_t offloadInfo;
+            if (hasOffloadInfo) {
+                data.read(&offloadInfo, sizeof(audio_offload_info_t));
+            }
+            audio_io_handle_t output = getOutputForAttr(attr,
+                    samplingRate,
+                    format,
+                    channelMask,
+                    flags,
+                    hasOffloadInfo ? &offloadInfo : NULL);
+            reply->writeInt32(static_cast <int>(output));
+            return NO_ERROR;
+        } break;
+
         case START_OUTPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
             audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index 432d890..38f717c 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -15,8 +15,10 @@
 ** limitations under the License.
 */
 
+#include <inttypes.h>
 #include <stdint.h>
 #include <sys/types.h>
+
 #include <binder/Parcel.h>
 #include <media/IMediaHTTPService.h>
 #include <media/IMediaMetadataRetriever.h>
@@ -125,7 +127,7 @@
 
     sp<IMemory> getFrameAtTime(int64_t timeUs, int option)
     {
-        ALOGV("getTimeAtTime: time(%lld us) and option(%d)", timeUs, option);
+        ALOGV("getTimeAtTime: time(%" PRId64 " us) and option(%d)", timeUs, option);
         Parcel data, reply;
         data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
         data.writeInt64(timeUs);
@@ -237,7 +239,7 @@
             CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
             int64_t timeUs = data.readInt64();
             int option = data.readInt32();
-            ALOGV("getTimeAtTime: time(%lld us) and option(%d)", timeUs, option);
+            ALOGV("getTimeAtTime: time(%" PRId64 " us) and option(%d)", timeUs, option);
 #ifndef DISABLE_GROUP_SCHEDULE_HACK
             setSchedPolicy(data);
 #endif
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index 8e58162..95af006 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -17,6 +17,10 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "IMediaRecorder"
+
+#include <inttypes.h>
+#include <unistd.h>
+
 #include <utils/Log.h>
 #include <binder/Parcel.h>
 #include <camera/ICamera.h>
@@ -24,8 +28,6 @@
 #include <media/IMediaRecorder.h>
 #include <gui/Surface.h>
 #include <gui/IGraphicBufferProducer.h>
-#include <unistd.h>
-
 
 namespace android {
 
@@ -167,7 +169,7 @@
     }
 
     status_t setOutputFile(int fd, int64_t offset, int64_t length) {
-        ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+        ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
         Parcel data, reply;
         data.writeInterfaceToken(IMediaRecorder::getInterfaceDescriptor());
         data.writeFileDescriptor(fd);
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index 28238c4..e9e453b 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -475,7 +475,7 @@
 }
 
 void MediaProfiles::initRequiredProfileRefs(const Vector<int>& cameraIds) {
-    ALOGV("Number of camera ids: %d", cameraIds.size());
+    ALOGV("Number of camera ids: %zu", cameraIds.size());
     CHECK(cameraIds.size() > 0);
     mRequiredProfileRefs = new RequiredProfiles[cameraIds.size()];
     for (size_t i = 0, n = cameraIds.size(); i < n; ++i) {
@@ -602,14 +602,14 @@
 
                 int index = getCamcorderProfileIndex(cameraId, profile->mQuality);
                 if (index != -1) {
-                    ALOGV("Profile quality %d for camera %d already exists",
+                    ALOGV("Profile quality %d for camera %zu already exists",
                         profile->mQuality, cameraId);
                     CHECK(index == refIndex);
                     continue;
                 }
 
                 // Insert the new profile
-                ALOGV("Add a profile: quality %d=>%d for camera %d",
+                ALOGV("Add a profile: quality %d=>%d for camera %zu",
                         mCamcorderProfiles[info->mRefProfileIndex]->mQuality,
                         profile->mQuality, cameraId);
 
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index a55e09c..2aa0592 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -16,6 +16,9 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "SoundPool"
+
+#include <inttypes.h>
+
 #include <utils/Log.h>
 
 #define USE_SHARED_MEM_BUFFER
@@ -212,7 +215,7 @@
 
 int SoundPool::load(int fd, int64_t offset, int64_t length, int priority __unused)
 {
-    ALOGV("load: fd=%d, offset=%lld, length=%lld, priority=%d",
+    ALOGV("load: fd=%d, offset=%" PRId64 ", length=%" PRId64 ", priority=%d",
             fd, offset, length, priority);
     Mutex::Autolock lock(&mLock);
     sp<Sample> sample = new Sample(++mNextSampleID, fd, offset, length);
@@ -462,7 +465,8 @@
     mFd = dup(fd);
     mOffset = offset;
     mLength = length;
-    ALOGV("create sampleID=%d, fd=%d, offset=%lld, length=%lld", mSampleID, mFd, mLength, mOffset);
+    ALOGV("create sampleID=%d, fd=%d, offset=%" PRId64 " length=%" PRId64,
+        mSampleID, mFd, mLength, mOffset);
 }
 
 void Sample::init()
@@ -516,7 +520,7 @@
         ALOGE("Unable to load sample: %s", mUrl);
         goto error;
     }
-    ALOGV("pointer = %p, size = %u, sampleRate = %u, numChannels = %d",
+    ALOGV("pointer = %p, size = %zu, sampleRate = %u, numChannels = %d",
           mHeap->getBase(), mSize, sampleRate, numChannels);
 
     if (sampleRate > kMaxSampleRate) {
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 1d6bb6f..39a239d 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -18,6 +18,8 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MediaMetadataRetriever"
 
+#include <inttypes.h>
+
 #include <binder/IServiceManager.h>
 #include <binder/IPCThreadState.h>
 #include <media/mediametadataretriever.h>
@@ -114,7 +116,7 @@
 
 status_t MediaMetadataRetriever::setDataSource(int fd, int64_t offset, int64_t length)
 {
-    ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+    ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
     Mutex::Autolock _l(mLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
@@ -129,7 +131,7 @@
 
 sp<IMemory> MediaMetadataRetriever::getFrameAtTime(int64_t timeUs, int option)
 {
-    ALOGV("getFrameAtTime: time(%lld us) option(%d)", timeUs, option);
+    ALOGV("getFrameAtTime: time(%" PRId64 " us) option(%d)", timeUs, option);
     Mutex::Autolock _l(mLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 0be01a9..406f9f2 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -17,12 +17,14 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MediaPlayer"
-#include <utils/Log.h>
 
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <unistd.h>
 #include <fcntl.h>
+#include <inttypes.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+#include <unistd.h>
+
+#include <utils/Log.h>
 
 #include <binder/IServiceManager.h>
 #include <binder/IPCThreadState.h>
@@ -157,7 +159,7 @@
 
 status_t MediaPlayer::setDataSource(int fd, int64_t offset, int64_t length)
 {
-    ALOGV("setDataSource(%d, %lld, %lld)", fd, offset, length);
+    ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
     status_t err = UNKNOWN_ERROR;
     const sp<IMediaPlayerService>& service(getMediaPlayerService());
     if (service != 0) {
@@ -194,7 +196,7 @@
             (mCurrentState != MEDIA_PLAYER_STATE_ERROR) &&
             ((mCurrentState & MEDIA_PLAYER_IDLE) != MEDIA_PLAYER_IDLE);
     if ((mPlayer != NULL) && hasBeenInitialized) {
-        ALOGV("invoke %d", request.dataSize());
+        ALOGV("invoke %zu", request.dataSize());
         return  mPlayer->invoke(request, reply);
     }
     ALOGE("invoke failed: wrong state %X", mCurrentState);
@@ -818,7 +820,7 @@
                                         audio_format_t* pFormat,
                                         const sp<IMemoryHeap>& heap, size_t *pSize)
 {
-    ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
+    ALOGV("decode(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
     status_t status;
     const sp<IMediaPlayerService>& service = getMediaPlayerService();
     if (service != 0) {
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 3710e46..c8192e9 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -17,6 +17,9 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MediaRecorder"
+
+#include <inttypes.h>
+
 #include <utils/Log.h>
 #include <media/mediarecorder.h>
 #include <binder/IServiceManager.h>
@@ -286,7 +289,7 @@
 
 status_t MediaRecorder::setOutputFile(int fd, int64_t offset, int64_t length)
 {
-    ALOGV("setOutputFile(%d, %lld, %lld)", fd, offset, length);
+    ALOGV("setOutputFile(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
     if (mMediaRecorder == NULL) {
         ALOGE("media recorder is not initialized yet");
         return INVALID_OPERATION;
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index 778eb9a..76632a7 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -307,7 +307,7 @@
     return new RemoteDisplay(client, iface.string());
 }
 
-status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& args) const
+status_t MediaPlayerService::AudioCache::dump(int fd, const Vector<String16>& /*args*/) const
 {
     const size_t SIZE = 256;
     char buffer[SIZE];
@@ -673,8 +673,8 @@
 
     ALOGV("st_dev  = %llu", sb.st_dev);
     ALOGV("st_mode = %u", sb.st_mode);
-    ALOGV("st_uid  = %lu", sb.st_uid);
-    ALOGV("st_gid  = %lu", sb.st_gid);
+    ALOGV("st_uid  = %lu", static_cast<unsigned long>(sb.st_uid));
+    ALOGV("st_gid  = %lu", static_cast<unsigned long>(sb.st_gid));
     ALOGV("st_size = %llu", sb.st_size);
 
     if (offset >= sb.st_size) {
@@ -803,7 +803,7 @@
 }
 
 status_t MediaPlayerService::Client::getMetadata(
-        bool update_only, bool apply_filter, Parcel *reply)
+        bool update_only, bool /*apply_filter*/, Parcel *reply)
 {
     sp<MediaPlayerBase> player = getPlayer();
     if (player == 0) return UNKNOWN_ERROR;
@@ -1926,8 +1926,8 @@
 status_t MediaPlayerService::AudioCache::open(
         uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
         audio_format_t format, int bufferCount,
-        AudioCallback cb, void *cookie, audio_output_flags_t flags,
-        const audio_offload_info_t *offloadInfo)
+        AudioCallback cb, void *cookie, audio_output_flags_t /*flags*/,
+        const audio_offload_info_t* /*offloadInfo*/)
 {
     ALOGV("open(%u, %d, 0x%x, %d, %d)", sampleRate, channelCount, channelMask, format, bufferCount);
     if (mHeap->getHeapID() < 0) {
@@ -1994,7 +1994,7 @@
 }
 
 void MediaPlayerService::AudioCache::notify(
-        void* cookie, int msg, int ext1, int ext2, const Parcel *obj)
+        void* cookie, int msg, int ext1, int ext2, const Parcel* /*obj*/)
 {
     ALOGV("notify(%p, %d, %d, %d)", cookie, msg, ext1, ext2);
     AudioCache* p = static_cast<AudioCache*>(cookie);
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 80c7e0a..a91b0e5 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -147,8 +147,8 @@
     }
     ALOGV("st_dev  = %llu", sb.st_dev);
     ALOGV("st_mode = %u", sb.st_mode);
-    ALOGV("st_uid  = %lu", sb.st_uid);
-    ALOGV("st_gid  = %lu", sb.st_gid);
+    ALOGV("st_uid  = %lu", static_cast<unsigned long>(sb.st_uid));
+    ALOGV("st_gid  = %lu", static_cast<unsigned long>(sb.st_gid));
     ALOGV("st_size = %llu", sb.st_size);
 
     if (offset >= sb.st_size) {
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index deeddd1..749ef96 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -114,7 +114,7 @@
 }
 
 status_t MidiFile::setDataSource(
-        const sp<IMediaHTTPService> &httpService,
+        const sp<IMediaHTTPService> & /*httpService*/,
         const char* path,
         const KeyedVector<String8, String8> *) {
     ALOGV("MidiFile::setDataSource url=%s", path);
diff --git a/media/libnbaio/MonoPipe.cpp b/media/libnbaio/MonoPipe.cpp
index 4adf018..0b65861 100644
--- a/media/libnbaio/MonoPipe.cpp
+++ b/media/libnbaio/MonoPipe.cpp
@@ -14,6 +14,8 @@
  * limitations under the License.
  */
 
+#include <inttypes.h>
+
 #define LOG_TAG "MonoPipe"
 //#define LOG_NDEBUG 0
 
@@ -87,7 +89,7 @@
     static const uint64_t kUnsignedHiBitsMask = ~(0xFFFFFFFFull);
     if ((N & kSignedHiBitsMask) || (D & kUnsignedHiBitsMask)) {
         ALOGE("Cannot reduce sample rate to local clock frequency ratio to fit"
-              " in a 32/32 bit rational.  (max reduction is 0x%016llx/0x%016llx"
+              " in a 32/32 bit rational.  (max reduction is 0x%016" PRIx64 "/0x%016" PRIx64
               ").  getNextWriteTimestamp calls will be non-functional", N, D);
         return;
     }
@@ -308,7 +310,7 @@
         // error, but then zero out the ratio in the linear transform so
         // that we don't try to do any conversions from now on.  This
         // MonoPipe's getNextWriteTimestamp is now broken for good.
-        ALOGE("Overflow when attempting to convert %d audio frames to"
+        ALOGE("Overflow when attempting to convert %zu audio frames to"
               " duration in local time.  getNextWriteTimestamp will fail from"
               " now on.", audFrames);
         mSamplesToLocalTime.a_to_b_numer = 0;
diff --git a/media/libnbaio/NBAIO.cpp b/media/libnbaio/NBAIO.cpp
index ff3284c..d641e74 100644
--- a/media/libnbaio/NBAIO.cpp
+++ b/media/libnbaio/NBAIO.cpp
@@ -137,7 +137,7 @@
 ssize_t NBAIO_Port::negotiate(const NBAIO_Format offers[], size_t numOffers,
                                   NBAIO_Format counterOffers[], size_t& numCounterOffers)
 {
-    ALOGV("negotiate offers=%p numOffers=%u countersOffers=%p numCounterOffers=%u",
+    ALOGV("negotiate offers=%p numOffers=%zu countersOffers=%p numCounterOffers=%zu",
             offers, numOffers, counterOffers, numCounterOffers);
     if (Format_isValid(mFormat)) {
         for (size_t i = 0; i < numOffers; ++i) {
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 6130084..79bdfe8 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -54,6 +54,7 @@
                 return mStreamType;
             }
             bool        isOffloaded() const { return (mFlags & IAudioFlinger::TRACK_OFFLOAD) != 0; }
+            bool        isDirect() const { return (mFlags & IAudioFlinger::TRACK_DIRECT) != 0; }
             status_t    setParameters(const String8& keyValuePairs);
             status_t    attachAuxEffect(int EffectId);
             void        setAuxBuffer(int EffectId, int32_t *buffer);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d6390b1..8ce819c 100755
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -1660,7 +1660,7 @@
     track->mState = TrackBase::STOPPED;
     if (!trackActive) {
         removeTrack_l(track);
-    } else if (track->isFastTrack() || track->isOffloaded()) {
+    } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) {
         track->mState = TrackBase::STOPPING_1;
     }
 
@@ -1868,7 +1868,9 @@
     }
     mNormalFrameCount = multiplier * mFrameCount;
     // round up to nearest 16 frames to satisfy AudioMixer
-    mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+    if (mType == MIXER || mType == DUPLICATING) {
+        mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
+    }
     ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount,
             mNormalFrameCount);
 
@@ -2656,7 +2658,7 @@
     if (mNormalSink != 0) {
         return mNormalSink->getTimestamp(timestamp);
     }
-    if (mType == OFFLOAD && mOutput->stream->get_presentation_position) {
+    if ((mType == OFFLOAD || mType == DIRECT) && mOutput->stream->get_presentation_position) {
         uint64_t position64;
         int ret = mOutput->stream->get_presentation_position(
                                                 mOutput->stream, &position64, &timestamp.mTime);
@@ -3947,14 +3949,16 @@
         // The first time a track is added we wait
         // for all its buffers to be filled before processing it
         uint32_t minFrames;
-        if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) {
+        if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing()) {
             minFrames = mNormalFrameCount;
         } else {
             minFrames = 1;
         }
 
-        if ((track->framesReady() >= minFrames) && track->isReady() &&
-                !track->isPaused() && !track->isTerminated())
+        ALOGI("prepareTracks_l minFrames %d state %d frames ready %d, ",
+              minFrames, track->mState, track->framesReady());
+        if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
+                !track->isStopping_2() && !track->isStopped())
         {
             ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer);
 
@@ -3981,17 +3985,26 @@
             if (!mEffectChains.isEmpty() && last) {
                 mEffectChains[0]->clearInputBuffer();
             }
-
-            ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer);
-            if ((track->sharedBuffer() != 0) || track->isTerminated() ||
-                    track->isStopped() || track->isPaused()) {
+            if (track->isStopping_1()) {
+                track->mState = TrackBase::STOPPING_2;
+            }
+            if ((track->sharedBuffer() != 0) || track->isStopped() ||
+                    track->isStopping_2() || track->isPaused()) {
                 // We have consumed all the buffers of this track.
                 // Remove it from the list of active tracks.
-                // TODO: implement behavior for compressed audio
-                size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
+                size_t audioHALFrames;
+                if (audio_is_linear_pcm(mFormat)) {
+                    audioHALFrames = (latency_l() * mSampleRate) / 1000;
+                } else {
+                    audioHALFrames = 0;
+                }
+
                 size_t framesWritten = mBytesWritten / mFrameSize;
                 if (mStandby || !last ||
                         track->presentationComplete(framesWritten, audioHALFrames)) {
+                    if (track->isStopping_2()) {
+                        track->mState = TrackBase::STOPPED;
+                    }
                     if (track->isStopped()) {
                         track->reset();
                     }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1e906ad..4fbb973 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -706,7 +706,7 @@
             if (playbackThread->mActiveTracks.indexOf(this) < 0) {
                 reset();
                 mState = STOPPED;
-            } else if (!isFastTrack() && !isOffloaded()) {
+            } else if (!isFastTrack() && !isOffloaded() && !isDirect()) {
                 mState = STOPPED;
             } else {
                 // For fast tracks prepareTracks_l() will set state to STOPPING_2
@@ -860,7 +860,7 @@
     }
     Mutex::Autolock _l(thread->mLock);
     PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
-    if (!isOffloaded()) {
+    if (!isOffloaded() && !isDirect()) {
         if (!playbackThread->mLatchQValid) {
             mPreviousValid = false;
             return INVALID_OPERATION;
@@ -980,8 +980,6 @@
     }
 
     if (framesWritten >= mPresentationCompleteFrames || isOffloaded()) {
-        ALOGV("presentationComplete() session %d complete: framesWritten %d",
-                  mSessionId, framesWritten);
         triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE);
         mAudioTrackServerProxy->setStreamEndDone();
         return true;
diff --git a/services/audioflinger/tests/Android.mk b/services/audioflinger/tests/Android.mk
index 874f18f..f365637 100644
--- a/services/audioflinger/tests/Android.mk
+++ b/services/audioflinger/tests/Android.mk
@@ -1,5 +1,8 @@
 # Build the unit tests for audioflinger
 
+#
+# resampler unit test
+#
 LOCAL_PATH:= $(call my-dir)
 include $(CLEAR_VARS)
 
@@ -20,6 +23,7 @@
 	bionic/libstdc++/include \
 	external/gtest/include \
 	external/stlport/stlport \
+	$(call include-path-for, audio-utils) \
 	frameworks/av/services/audioflinger
 
 LOCAL_SRC_FILES := \
diff --git a/services/audioflinger/tests/resampler_tests.cpp b/services/audioflinger/tests/resampler_tests.cpp
index 8f9c270..4a67d0b 100644
--- a/services/audioflinger/tests/resampler_tests.cpp
+++ b/services/audioflinger/tests/resampler_tests.cpp
@@ -33,200 +33,7 @@
 #include <gtest/gtest.h>
 #include <media/AudioBufferProvider.h>
 #include "AudioResampler.h"
-
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-template<typename T, typename U>
-struct is_same
-{
-    static const bool value = false;
-};
-
-template<typename T>
-struct is_same<T, T>  // partial specialization
-{
-    static const bool value = true;
-};
-
-template<typename T>
-static inline T convertValue(double val)
-{
-    if (is_same<T, int16_t>::value) {
-        return floor(val * 32767.0 + 0.5);
-    } else if (is_same<T, int32_t>::value) {
-        return floor(val * (1UL<<31) + 0.5);
-    }
-    return val; // assume float or double
-}
-
-/* Creates a type-independent audio buffer provider from
- * a buffer base address, size, framesize, and input increment array.
- *
- * No allocation or deallocation of the provided buffer is done.
- */
-class TestProvider : public android::AudioBufferProvider {
-public:
-    TestProvider(const void* addr, size_t frames, size_t frameSize,
-            const std::vector<size_t>& inputIncr)
-    : mAddr(addr),
-      mNumFrames(frames),
-      mFrameSize(frameSize),
-      mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
-    {
-    }
-
-    virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS )
-    {
-        size_t requestedFrames = buffer->frameCount;
-        if (requestedFrames > mNumFrames - mNextFrame) {
-            buffer->frameCount = mNumFrames - mNextFrame;
-        }
-        if (!mInputIncr.empty()) {
-            size_t provided = mInputIncr[mNextIdx++];
-            ALOGV("getNextBuffer() mValue[%d]=%u not %u",
-                    mNextIdx-1, provided, buffer->frameCount);
-            if (provided < buffer->frameCount) {
-                buffer->frameCount = provided;
-            }
-            if (mNextIdx >= mInputIncr.size()) {
-                mNextIdx = 0;
-            }
-        }
-        ALOGV("getNextBuffer() requested %u frames out of %u frames available"
-                " and returned %u frames\n",
-                requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
-        mUnrel = buffer->frameCount;
-        if (buffer->frameCount > 0) {
-            buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
-            return android::NO_ERROR;
-        } else {
-            buffer->raw = NULL;
-            return android::NOT_ENOUGH_DATA;
-        }
-    }
-
-    virtual void releaseBuffer(Buffer* buffer)
-    {
-        if (buffer->frameCount > mUnrel) {
-            ALOGE("releaseBuffer() released %u frames but only %u available "
-                    "to release\n", buffer->frameCount, mUnrel);
-            mNextFrame += mUnrel;
-            mUnrel = 0;
-        } else {
-
-            ALOGV("releaseBuffer() released %u frames out of %u frames available "
-                    "to release\n", buffer->frameCount, mUnrel);
-            mNextFrame += buffer->frameCount;
-            mUnrel -= buffer->frameCount;
-        }
-        buffer->frameCount = 0;
-        buffer->raw = NULL;
-    }
-
-    void reset()
-    {
-        mNextFrame = 0;
-    }
-
-    size_t getNumFrames()
-    {
-        return mNumFrames;
-    }
-
-    void setIncr(const std::vector<size_t> inputIncr)
-    {
-        mNextIdx = 0;
-        mInputIncr = inputIncr;
-    }
-
-protected:
-    const void* mAddr;   // base address
-    size_t mNumFrames;   // total frames
-    int mFrameSize;      // frame size (# channels * bytes per sample)
-    size_t mNextFrame;   // index of next frame to provide
-    size_t mUnrel;       // number of frames not yet released
-    std::vector<size_t> mInputIncr; // number of frames provided per call
-    size_t mNextIdx;     // index of next entry in mInputIncr to use
-};
-
-/* Creates a buffer filled with a sine wave.
- *
- * Returns a pair consisting of the sine signal buffer and the number of frames.
- * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
- */
-template<typename T>
-static std::pair<T*, size_t> createSine(size_t channels,
-        double freq, double samplingRate, double time)
-{
-    double tscale = 1. / samplingRate;
-    size_t frames = static_cast<size_t>(samplingRate * time);
-    T* buffer = new T[frames * channels];
-    for (size_t i = 0; i < frames; ++i) {
-        double t = i * tscale;
-        double y = sin(2. * M_PI * freq * t);
-        T yt = convertValue<T>(y);
-
-        for (size_t j = 0; j < channels; ++j) {
-            buffer[i*channels + j] = yt / (j + 1);
-        }
-    }
-    return std::make_pair(buffer, frames);
-}
-
-/* Creates a buffer filled with a chirp signal (a sine wave sweep).
- *
- * Returns a pair consisting of the chirp signal buffer and the number of frames.
- * The caller must delete[] the buffer when no longer needed (no shared_ptr<>).
- *
- * When creating the Chirp, note that the frequency is the true sinusoidal
- * frequency not the sampling rate.
- *
- * http://en.wikipedia.org/wiki/Chirp
- */
-template<typename T>
-static std::pair<T*, size_t> createChirp(size_t channels,
-        double minfreq, double maxfreq, double samplingRate, double time)
-{
-    double tscale = 1. / samplingRate;
-    size_t frames = static_cast<size_t>(samplingRate * time);
-    T *buffer = new T[frames * channels];
-    // note the chirp constant k has a divide-by-two.
-    double k = (maxfreq - minfreq) / (2. * time);
-    for (size_t i = 0; i < frames; ++i) {
-        double t = i * tscale;
-        double y = sin(2. * M_PI * (k * t + minfreq) * t);
-        T yt = convertValue<T>(y);
-
-        for (size_t j = 0; j < channels; ++j) {
-            buffer[i*channels + j] = yt / (j + 1);
-        }
-    }
-    return std::make_pair(buffer, frames);
-}
-
-/* This derived class creates a buffer provider of datatype T,
- * consisting of an input signal, e.g. from createChirp().
- * The number of frames can be obtained from the base class
- * TestProvider::getNumFrames().
- */
-template <typename T>
-class SignalProvider : public TestProvider {
-public:
-    SignalProvider(const std::pair<T*, size_t>& bufferInfo, size_t channels,
-            const std::vector<size_t>& values)
-    : TestProvider(bufferInfo.first, bufferInfo.second, channels * sizeof(T), values),
-      mManagedPtr(bufferInfo.first)
-    {
-    }
-
-    virtual ~SignalProvider()
-    {
-        delete[] mManagedPtr;
-    }
-
-protected:
-    T* mManagedPtr;
-};
+#include "test_utils.h"
 
 void resample(void *output, size_t outputFrames, const std::vector<size_t> &outputIncr,
         android::AudioBufferProvider *provider, android::AudioResampler *resampler)
@@ -261,10 +68,11 @@
         enum android::AudioResampler::src_quality quality)
 {
     // create the provider
-    std::vector<size_t> inputIncr;
-    SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
-            0., outputFreq/2., outputFreq, outputFreq/2000.),
-            channels, inputIncr);
+    std::vector<int> inputIncr;
+    SignalProvider provider;
+    provider.setChirp<int16_t>(channels,
+            0., outputFreq/2., outputFreq, outputFreq/2000.);
+    provider.setIncr(inputIncr);
 
     // calculate the output size
     size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
@@ -339,10 +147,11 @@
         enum android::AudioResampler::src_quality quality)
 {
     // create the provider
-    std::vector<size_t> inputIncr;
-    SignalProvider<int16_t> provider(createChirp<int16_t>(channels,
-            0., inputFreq/2., inputFreq, inputFreq/2000.),
-            channels, inputIncr);
+    std::vector<int> inputIncr;
+    SignalProvider provider;
+    provider.setChirp<int16_t>(channels,
+            0., inputFreq/2., inputFreq, inputFreq/2000.);
+    provider.setIncr(inputIncr);
 
     // calculate the output size
     size_t outputFrames = ((int64_t) provider.getNumFrames() * outputFreq) / inputFreq;
diff --git a/services/audioflinger/tests/test_utils.h b/services/audioflinger/tests/test_utils.h
new file mode 100644
index 0000000..f954292
--- /dev/null
+++ b/services/audioflinger/tests/test_utils.h
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2014 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_TEST_UTILS_H
+#define ANDROID_AUDIO_TEST_UTILS_H
+
+#include <audio_utils/sndfile.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+template<typename T, typename U>
+struct is_same
+{
+    static const bool value = false;
+};
+
+template<typename T>
+struct is_same<T, T>  // partial specialization
+{
+    static const bool value = true;
+};
+
+template<typename T>
+static inline T convertValue(double val)
+{
+    if (is_same<T, int16_t>::value) {
+        return floor(val * 32767.0 + 0.5);
+    } else if (is_same<T, int32_t>::value) {
+        return floor(val * (1UL<<31) + 0.5);
+    }
+    return val; // assume float or double
+}
+
+// Convert a list of integers in CSV format to a Vector of those values.
+// Returns the number of elements in the list, or -1 on error.
+static inline int parseCSV(const char *string, std::vector<int>& values)
+{
+    // pass 1: count the number of values and do syntax check
+    size_t numValues = 0;
+    bool hadDigit = false;
+    for (const char *p = string; ; ) {
+        switch (*p++) {
+        case '0': case '1': case '2': case '3': case '4':
+        case '5': case '6': case '7': case '8': case '9':
+            hadDigit = true;
+            break;
+        case '\0':
+            if (hadDigit) {
+                // pass 2: allocate and initialize vector of values
+                values.resize(++numValues);
+                values[0] = atoi(p = string);
+                for (size_t i = 1; i < numValues; ) {
+                    if (*p++ == ',') {
+                        values[i++] = atoi(p);
+                    }
+                }
+                return numValues;
+            }
+            // fall through
+        case ',':
+            if (hadDigit) {
+                hadDigit = false;
+                numValues++;
+                break;
+            }
+            // fall through
+        default:
+            return -1;
+        }
+    }
+}
+
+/* Creates a type-independent audio buffer provider from
+ * a buffer base address, size, framesize, and input increment array.
+ *
+ * No allocation or deallocation of the provided buffer is done.
+ */
+class TestProvider : public android::AudioBufferProvider {
+public:
+    TestProvider(void* addr, size_t frames, size_t frameSize,
+            const std::vector<int>& inputIncr)
+    : mAddr(addr),
+      mNumFrames(frames),
+      mFrameSize(frameSize),
+      mNextFrame(0), mUnrel(0), mInputIncr(inputIncr), mNextIdx(0)
+    {
+    }
+
+    TestProvider()
+    : mAddr(NULL), mNumFrames(0), mFrameSize(0),
+      mNextFrame(0), mUnrel(0), mNextIdx(0)
+    {
+    }
+
+    void setIncr(const std::vector<int>& inputIncr) {
+        mInputIncr = inputIncr;
+        mNextIdx = 0;
+    }
+
+    virtual android::status_t getNextBuffer(Buffer* buffer, int64_t pts __unused = kInvalidPTS)
+    {
+        size_t requestedFrames = buffer->frameCount;
+        if (requestedFrames > mNumFrames - mNextFrame) {
+            buffer->frameCount = mNumFrames - mNextFrame;
+        }
+        if (!mInputIncr.empty()) {
+            size_t provided = mInputIncr[mNextIdx++];
+            ALOGV("getNextBuffer() mValue[%d]=%u not %u",
+                    mNextIdx-1, provided, buffer->frameCount);
+            if (provided < buffer->frameCount) {
+                buffer->frameCount = provided;
+            }
+            if (mNextIdx >= mInputIncr.size()) {
+                mNextIdx = 0;
+            }
+        }
+        ALOGV("getNextBuffer() requested %u frames out of %u frames available"
+                " and returned %u frames\n",
+                requestedFrames, mNumFrames - mNextFrame, buffer->frameCount);
+        mUnrel = buffer->frameCount;
+        if (buffer->frameCount > 0) {
+            buffer->raw = (char *)mAddr + mFrameSize * mNextFrame;
+            return android::NO_ERROR;
+        } else {
+            buffer->raw = NULL;
+            return android::NOT_ENOUGH_DATA;
+        }
+    }
+
+    virtual void releaseBuffer(Buffer* buffer)
+    {
+        if (buffer->frameCount > mUnrel) {
+            ALOGE("releaseBuffer() released %u frames but only %u available "
+                    "to release\n", buffer->frameCount, mUnrel);
+            mNextFrame += mUnrel;
+            mUnrel = 0;
+        } else {
+
+            ALOGV("releaseBuffer() released %u frames out of %u frames available "
+                    "to release\n", buffer->frameCount, mUnrel);
+            mNextFrame += buffer->frameCount;
+            mUnrel -= buffer->frameCount;
+        }
+        buffer->frameCount = 0;
+        buffer->raw = NULL;
+    }
+
+    void reset()
+    {
+        mNextFrame = 0;
+    }
+
+    size_t getNumFrames()
+    {
+        return mNumFrames;
+    }
+
+
+protected:
+    void* mAddr;   // base address
+    size_t mNumFrames;   // total frames
+    int mFrameSize;      // frame size (# channels * bytes per sample)
+    size_t mNextFrame;   // index of next frame to provide
+    size_t mUnrel;       // number of frames not yet released
+    std::vector<int> mInputIncr; // number of frames provided per call
+    size_t mNextIdx;     // index of next entry in mInputIncr to use
+};
+
+/* Creates a buffer filled with a sine wave.
+ */
+template<typename T>
+static void createSine(void *vbuffer, size_t frames,
+        size_t channels, double sampleRate, double freq)
+{
+    double tscale = 1. / sampleRate;
+    T* buffer = reinterpret_cast<T*>(vbuffer);
+    for (size_t i = 0; i < frames; ++i) {
+        double t = i * tscale;
+        double y = sin(2. * M_PI * freq * t);
+        T yt = convertValue<T>(y);
+
+        for (size_t j = 0; j < channels; ++j) {
+            buffer[i*channels + j] = yt / (j + 1);
+        }
+    }
+}
+
+/* Creates a buffer filled with a chirp signal (a sine wave sweep).
+ *
+ * When creating the Chirp, note that the frequency is the true sinusoidal
+ * frequency not the sampling rate.
+ *
+ * http://en.wikipedia.org/wiki/Chirp
+ */
+template<typename T>
+static void createChirp(void *vbuffer, size_t frames,
+        size_t channels, double sampleRate,  double minfreq, double maxfreq)
+{
+    double tscale = 1. / sampleRate;
+    T *buffer = reinterpret_cast<T*>(vbuffer);
+    // note the chirp constant k has a divide-by-two.
+    double k = (maxfreq - minfreq) / (2. * tscale * frames);
+    for (size_t i = 0; i < frames; ++i) {
+        double t = i * tscale;
+        double y = sin(2. * M_PI * (k * t + minfreq) * t);
+        T yt = convertValue<T>(y);
+
+        for (size_t j = 0; j < channels; ++j) {
+            buffer[i*channels + j] = yt / (j + 1);
+        }
+    }
+}
+
+/* This derived class creates a buffer provider of datatype T,
+ * consisting of an input signal, e.g. from createChirp().
+ * The number of frames can be obtained from the base class
+ * TestProvider::getNumFrames().
+ */
+
+class SignalProvider : public TestProvider {
+public:
+    SignalProvider()
+    : mSampleRate(0),
+      mChannels(0)
+    {
+    }
+
+    virtual ~SignalProvider()
+    {
+        free(mAddr);
+        mAddr = NULL;
+    }
+
+    template <typename T>
+    void setChirp(size_t channels, double minfreq, double maxfreq, double sampleRate, double time)
+    {
+        createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+        createChirp<T>(mAddr, mNumFrames, mChannels, mSampleRate, minfreq, maxfreq);
+    }
+
+    template <typename T>
+    void setSine(size_t channels,
+            double freq, double sampleRate, double time)
+    {
+        createBufferByFrames<T>(channels, sampleRate, sampleRate*time);
+        createSine<T>(mAddr, mNumFrames,  mChannels, mSampleRate, freq);
+    }
+
+    template <typename T>
+    void setFile(const char *file_in)
+    {
+        SF_INFO info;
+        info.format = 0;
+        SNDFILE *sf = sf_open(file_in, SFM_READ, &info);
+        if (sf == NULL) {
+            perror(file_in);
+            return;
+        }
+        createBufferByFrames<T>(info.channels, info.samplerate, info.frames);
+        if (is_same<T, float>::value) {
+            (void) sf_readf_float(sf, (float *) mAddr, mNumFrames);
+        } else if (is_same<T, short>::value) {
+            (void) sf_readf_short(sf, (short *) mAddr, mNumFrames);
+        }
+        sf_close(sf);
+    }
+
+    template <typename T>
+    void createBufferByFrames(size_t channels, uint32_t sampleRate, size_t frames)
+    {
+        mNumFrames = frames;
+        mChannels = channels;
+        mFrameSize = mChannels * sizeof(T);
+        free(mAddr);
+        mAddr = malloc(mFrameSize * mNumFrames);
+        mSampleRate = sampleRate;
+    }
+
+    uint32_t getSampleRate() const {
+        return mSampleRate;
+    }
+
+    uint32_t getNumChannels() const {
+        return mChannels;
+    }
+
+protected:
+    uint32_t mSampleRate;
+    uint32_t mChannels;
+};
+
+#endif // ANDROID_AUDIO_TEST_UTILS_H
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index c025a45..33e4397 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -90,6 +90,12 @@
                                         audio_channel_mask_t channelMask,
                                         audio_output_flags_t flags,
                                         const audio_offload_info_t *offloadInfo) = 0;
+    virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+                                                uint32_t samplingRate,
+                                                audio_format_t format,
+                                                audio_channel_mask_t channelMask,
+                                                audio_output_flags_t flags,
+                                                const audio_offload_info_t *offloadInfo) = 0;
     // indicates to the audio policy manager that the output starts being used by corresponding stream.
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
index 8cc386a..6342d8f 100644
--- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/AudioPolicyInterfaceImpl.cpp
@@ -131,6 +131,22 @@
                                     format, channelMask, flags, offloadInfo);
 }
 
+audio_io_handle_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_output_flags_t flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    if (mAudioPolicyManager == NULL) {
+        return 0;
+    }
+    ALOGV("getOutput()");
+    Mutex::Autolock _l(mLock);
+    return mAudioPolicyManager->getOutputForAttr(attr, samplingRate,
+                                    format, channelMask, flags, offloadInfo);
+}
+
 status_t AudioPolicyService::startOutput(audio_io_handle_t output,
                                          audio_stream_type_t stream,
                                          int session)
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/AudioPolicyManager.cpp
index d4c9374..74c6207 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/AudioPolicyManager.cpp
@@ -122,6 +122,11 @@
     STRING_TO_ENUM(AUDIO_FORMAT_MP3),
     STRING_TO_ENUM(AUDIO_FORMAT_AAC),
     STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+    STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+    STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+    STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+    STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
 };
 
 const StringToEnum sOutChannelsNameToEnumTable[] = {
@@ -623,13 +628,53 @@
                                     audio_output_flags_t flags,
                                     const audio_offload_info_t *offloadInfo)
 {
-    audio_io_handle_t output = 0;
-    uint32_t latency = 0;
+
     routing_strategy strategy = getStrategy(stream);
     audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
     ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
           device, stream, samplingRate, format, channelMask, flags);
 
+    return getOutputForDevice(device, stream, samplingRate,format, channelMask, flags,
+            offloadInfo);
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr,
+                                    uint32_t samplingRate,
+                                    audio_format_t format,
+                                    audio_channel_mask_t channelMask,
+                                    audio_output_flags_t flags,
+                                    const audio_offload_info_t *offloadInfo)
+{
+    if (attr == NULL) {
+        ALOGE("getOutputForAttr() called with NULL audio attributes");
+        return 0;
+    }
+    ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s",
+            attr->usage, attr->content_type, attr->tags);
+
+    // TODO this is where filtering for custom policies (rerouting, dynamic sources) will go
+    routing_strategy strategy = (routing_strategy) getStrategyForAttr(attr);
+    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    ALOGV("getOutputForAttr() device %d, samplingRate %d, format %x, channelMask %x, flags %x",
+          device, samplingRate, format, channelMask, flags);
+
+    audio_stream_type_t stream = streamTypefromAttributesInt(attr);
+    return getOutputForDevice(device, stream, samplingRate, format, channelMask, flags,
+                offloadInfo);
+}
+
+audio_io_handle_t AudioPolicyManager::getOutputForDevice(
+        audio_devices_t device,
+        audio_stream_type_t stream,
+        uint32_t samplingRate,
+        audio_format_t format,
+        audio_channel_mask_t channelMask,
+        audio_output_flags_t flags,
+        const audio_offload_info_t *offloadInfo)
+{
+    audio_io_handle_t output = 0;
+    uint32_t latency = 0;
+
 #ifdef AUDIO_POLICY_TEST
     if (mCurOutput != 0) {
         ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
@@ -1515,7 +1560,7 @@
     snprintf(buffer, SIZE,
              " Stream  Can be muted  Index Min  Index Max  Index Cur [device : index]...\n");
     write(fd, buffer, strlen(buffer));
-    for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+    for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) {
         snprintf(buffer, SIZE, " %02zu      ", i);
         write(fd, buffer, strlen(buffer));
         mStreams[i].dump(fd);
@@ -1653,7 +1698,7 @@
         }
     }
     *generation = curAudioPortGeneration();
-    ALOGV("listAudioPorts() got %d ports needed %d", portsWritten, *num_ports);
+    ALOGV("listAudioPorts() got %zu ports needed %d", portsWritten, *num_ports);
     return NO_ERROR;
 }
 
@@ -1997,7 +2042,7 @@
             generation == NULL) {
         return BAD_VALUE;
     }
-    ALOGV("listAudioPatches() num_patches %d patches %p available patches %d",
+    ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu",
           *num_patches, patches, mAudioPatches.size());
     if (patches == NULL) {
         *num_patches = 0;
@@ -2009,13 +2054,13 @@
             i  < mAudioPatches.size() && patchesWritten < patchesMax; i++) {
         patches[patchesWritten] = mAudioPatches[i]->mPatch;
         patches[patchesWritten++].id = mAudioPatches[i]->mHandle;
-        ALOGV("listAudioPatches() patch %d num_sources %d num_sinks %d",
+        ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
               i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks);
     }
     *num_patches = mAudioPatches.size();
 
     *generation = curAudioPortGeneration();
-    ALOGV("listAudioPatches() got %d patches needed %d", patchesWritten, *num_patches);
+    ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches);
     return NO_ERROR;
 }
 
@@ -2779,7 +2824,7 @@
             {
                 if (mHwModules[module_idx]->mInputProfiles[profile_index]->mSupportedDevices.types()
                         & (device & ~AUDIO_DEVICE_BIT_IN)) {
-                    ALOGV("checkInputsForDevice(): adding profile %d from module %d",
+                    ALOGV("checkInputsForDevice(): adding profile %zu from module %zu",
                           profile_index, module_idx);
                     profiles.add(mHwModules[module_idx]->mInputProfiles[profile_index]);
                 }
@@ -2903,7 +2948,7 @@
                  profile_index++) {
                 sp<IOProfile> profile = mHwModules[module_index]->mInputProfiles[profile_index];
                 if (profile->mSupportedDevices.types() & device) {
-                    ALOGV("checkInputsForDevice(): clearing direct input profile %d on module %d",
+                    ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %zu",
                           profile_index, module_index);
                     if (profile->mSamplingRates[0] == 0) {
                         profile->mSamplingRates.clear();
@@ -3239,6 +3284,44 @@
     }
 }
 
+uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) {
+    // flags to strategy mapping
+    if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+        return (uint32_t) STRATEGY_ENFORCED_AUDIBLE;
+    }
+
+    // usage to strategy mapping
+    switch (attr->usage) {
+    case AUDIO_USAGE_MEDIA:
+    case AUDIO_USAGE_GAME:
+    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+    case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+    case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+        return (uint32_t) STRATEGY_MEDIA;
+
+    case AUDIO_USAGE_VOICE_COMMUNICATION:
+        return (uint32_t) STRATEGY_PHONE;
+
+    case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+        return (uint32_t) STRATEGY_DTMF;
+
+    case AUDIO_USAGE_ALARM:
+    case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+        return (uint32_t) STRATEGY_SONIFICATION;
+
+    case AUDIO_USAGE_NOTIFICATION:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+    case AUDIO_USAGE_NOTIFICATION_EVENT:
+        return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL;
+
+    case AUDIO_USAGE_UNKNOWN:
+    default:
+        return (uint32_t) STRATEGY_MEDIA;
+    }
+}
+
 void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
     switch(stream) {
     case AUDIO_STREAM_MUSIC:
@@ -4911,7 +4994,7 @@
     }
     port->num_formats = i;
 
-    ALOGV("AudioPort::toAudioPort() num gains %d", mGains.size());
+    ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
 
     for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
         port->gains[i] = mGains[i]->mGain;
@@ -5609,7 +5692,7 @@
 {
     sp<DeviceDescriptor> device;
     for (size_t i = 0; i < size(); i++) {
-        ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%d)->mId %d", id, i, itemAt(i)->mId);
+        ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId);
         if (itemAt(i)->mId == id) {
             device = itemAt(i);
             break;
@@ -5916,4 +5999,46 @@
     mHwModules.add(module);
 }
 
+audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
+{
+    // flags to stream type mapping
+    if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) {
+        return AUDIO_STREAM_ENFORCED_AUDIBLE;
+    }
+    if ((attr->flags & AUDIO_FLAG_SCO) == AUDIO_FLAG_SCO) {
+        return AUDIO_STREAM_BLUETOOTH_SCO;
+    }
+
+    // usage to stream type mapping
+    switch (attr->usage) {
+    case AUDIO_USAGE_MEDIA:
+    case AUDIO_USAGE_GAME:
+    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+    case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+        return AUDIO_STREAM_MUSIC;
+    case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+        return AUDIO_STREAM_SYSTEM;
+    case AUDIO_USAGE_VOICE_COMMUNICATION:
+        return AUDIO_STREAM_VOICE_CALL;
+
+    case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+        return AUDIO_STREAM_DTMF;
+
+    case AUDIO_USAGE_ALARM:
+        return AUDIO_STREAM_ALARM;
+    case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+        return AUDIO_STREAM_RING;
+
+    case AUDIO_USAGE_NOTIFICATION:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+    case AUDIO_USAGE_NOTIFICATION_EVENT:
+        return AUDIO_STREAM_NOTIFICATION;
+
+    case AUDIO_USAGE_UNKNOWN:
+    default:
+        return AUDIO_STREAM_MUSIC;
+    }
+}
 }; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
index 1abeb6a..c23d994 100644
--- a/services/audiopolicy/AudioPolicyManager.h
+++ b/services/audiopolicy/AudioPolicyManager.h
@@ -84,6 +84,12 @@
                                             audio_channel_mask_t channelMask,
                                             audio_output_flags_t flags,
                                             const audio_offload_info_t *offloadInfo);
+        virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+                                            uint32_t samplingRate,
+                                            audio_format_t format,
+                                            audio_channel_mask_t channelMask,
+                                            audio_output_flags_t flags,
+                                            const audio_offload_info_t *offloadInfo);
         virtual status_t startOutput(audio_io_handle_t output,
                                      audio_stream_type_t stream,
                                      int session = 0);
@@ -116,6 +122,8 @@
 
         // return the strategy corresponding to a given stream type
         virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
+        // return the strategy corresponding to the given audio attributes
+        virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
 
         // return the enabled output devices for the given stream type
         virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
@@ -755,6 +763,17 @@
         uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
         // converts device address to string sent to audio HAL via setParameters
         static String8 addressToParameter(audio_devices_t device, const String8 address);
+        // internal method to return the output handle for the given device and format
+        audio_io_handle_t getOutputForDevice(
+                audio_devices_t device,
+                audio_stream_type_t stream,
+                uint32_t samplingRate,
+                audio_format_t format,
+                audio_channel_mask_t channelMask,
+                audio_output_flags_t flags,
+                const audio_offload_info_t *offloadInfo);
+        // internal function to derive a stream type value from audio attributes
+        audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
 };
 
 };
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/AudioPolicyService.h
index 66d9cad..69673cd 100644
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/AudioPolicyService.h
@@ -70,6 +70,12 @@
                                         audio_output_flags_t flags =
                                                 AUDIO_OUTPUT_FLAG_NONE,
                                         const audio_offload_info_t *offloadInfo = NULL);
+    virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr,
+                                            uint32_t samplingRate = 0,
+                                            audio_format_t format = AUDIO_FORMAT_DEFAULT,
+                                            audio_channel_mask_t channelMask = 0,
+                                            audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                                            const audio_offload_info_t *offloadInfo = NULL);
     virtual status_t startOutput(audio_io_handle_t output,
                                  audio_stream_type_t stream,
                                  int session = 0);
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 73eccbf..109b891 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -659,7 +659,8 @@
                                       int clientUid,
                                       int callingPid,
                                       /*out*/
-                                      sp<Client>& client) {
+                                      sp<Client>& client,
+                                      int halVersion) {
 
     int facing = -1;
     int deviceVersion = getDeviceVersion(cameraId, &facing);
@@ -672,28 +673,48 @@
                      cameraId);
     }
 
-    switch(deviceVersion) {
-      case CAMERA_DEVICE_API_VERSION_1_0:
-        client = new CameraClient(this, cameraClient,
-                clientPackageName, cameraId,
-                facing, callingPid, clientUid, getpid());
-        break;
-      case CAMERA_DEVICE_API_VERSION_2_0:
-      case CAMERA_DEVICE_API_VERSION_2_1:
-      case CAMERA_DEVICE_API_VERSION_3_0:
-      case CAMERA_DEVICE_API_VERSION_3_1:
-      case CAMERA_DEVICE_API_VERSION_3_2:
-        client = new Camera2Client(this, cameraClient,
-                clientPackageName, cameraId,
-                facing, callingPid, clientUid, getpid(),
-                deviceVersion);
-        break;
-      case -1:
-        ALOGE("Invalid camera id %d", cameraId);
-        return BAD_VALUE;
-      default:
-        ALOGE("Unknown camera device HAL version: %d", deviceVersion);
-        return INVALID_OPERATION;
+    if (halVersion < 0 || halVersion == deviceVersion) {
+        // Default path: HAL version is unspecified by caller, create CameraClient
+        // based on device version reported by the HAL.
+        switch(deviceVersion) {
+          case CAMERA_DEVICE_API_VERSION_1_0:
+            client = new CameraClient(this, cameraClient,
+                    clientPackageName, cameraId,
+                    facing, callingPid, clientUid, getpid());
+            break;
+          case CAMERA_DEVICE_API_VERSION_2_0:
+          case CAMERA_DEVICE_API_VERSION_2_1:
+          case CAMERA_DEVICE_API_VERSION_3_0:
+          case CAMERA_DEVICE_API_VERSION_3_1:
+          case CAMERA_DEVICE_API_VERSION_3_2:
+            client = new Camera2Client(this, cameraClient,
+                    clientPackageName, cameraId,
+                    facing, callingPid, clientUid, getpid(),
+                    deviceVersion);
+            break;
+          case -1:
+            ALOGE("Invalid camera id %d", cameraId);
+            return BAD_VALUE;
+          default:
+            ALOGE("Unknown camera device HAL version: %d", deviceVersion);
+            return INVALID_OPERATION;
+        }
+    } else {
+        // A particular HAL version is requested by caller. Create CameraClient
+        // based on the requested HAL version.
+        if (deviceVersion > CAMERA_DEVICE_API_VERSION_1_0 &&
+            halVersion == CAMERA_DEVICE_API_VERSION_1_0) {
+            // Only support higher HAL version device opened as HAL1.0 device.
+            client = new CameraClient(this, cameraClient,
+                    clientPackageName, cameraId,
+                    facing, callingPid, clientUid, getpid());
+        } else {
+            // Other combinations (e.g. HAL3.x open as HAL2.x) are not supported yet.
+            ALOGE("Invalid camera HAL version %x: HAL %x device can only be"
+                    " opened as HAL %x device", halVersion, deviceVersion,
+                    CAMERA_DEVICE_API_VERSION_1_0);
+            return INVALID_OPERATION;
+        }
     }
 
     status_t status = connectFinishUnsafe(client, client->getRemote());
@@ -762,6 +783,70 @@
     return OK;
 }
 
+status_t CameraService::connectLegacy(
+        const sp<ICameraClient>& cameraClient,
+        int cameraId, int halVersion,
+        const String16& clientPackageName,
+        int clientUid,
+        /*out*/
+        sp<ICamera>& device) {
+
+    if (halVersion != CAMERA_HAL_API_VERSION_UNSPECIFIED &&
+            mModule->common.module_api_version < CAMERA_MODULE_API_VERSION_2_3) {
+        /*
+         * Either the HAL version is unspecified in which case this just creates
+         * a camera client selected by the latest device version, or
+         * it's a particular version in which case the HAL must supported
+         * the open_legacy call
+         */
+        ALOGE("%s: camera HAL module version %x doesn't support connecting to legacy HAL devices!",
+                __FUNCTION__, mModule->common.module_api_version);
+        return INVALID_OPERATION;
+    }
+
+    String8 clientName8(clientPackageName);
+    int callingPid = getCallingPid();
+
+    LOG1("CameraService::connect legacy E (pid %d \"%s\", id %d)", callingPid,
+            clientName8.string(), cameraId);
+
+    status_t status = validateConnect(cameraId, /*inout*/clientUid);
+    if (status != OK) {
+        return status;
+    }
+
+    sp<Client> client;
+    {
+        Mutex::Autolock lock(mServiceLock);
+        sp<BasicClient> clientTmp;
+        if (!canConnectUnsafe(cameraId, clientPackageName,
+                              cameraClient->asBinder(),
+                              /*out*/clientTmp)) {
+            return -EBUSY;
+        } else if (client.get() != NULL) {
+            device = static_cast<Client*>(clientTmp.get());
+            return OK;
+        }
+
+        status = connectHelperLocked(cameraClient,
+                                     cameraId,
+                                     clientPackageName,
+                                     clientUid,
+                                     callingPid,
+                                     client,
+                                     halVersion);
+        if (status != OK) {
+            return status;
+        }
+
+    }
+    // important: release the mutex here so the client can call back
+    //    into the service from its destructor (can be at the end of the call)
+
+    device = client;
+    return OK;
+}
+
 status_t CameraService::connectFinishUnsafe(const sp<BasicClient>& client,
                                             const sp<IBinder>& remoteCallback) {
     status_t status = client->initialize(mModule);
@@ -1196,6 +1281,7 @@
         case BnCameraService::CONNECT:
         case BnCameraService::CONNECT_PRO:
         case BnCameraService::CONNECT_DEVICE:
+        case BnCameraService::CONNECT_LEGACY:
             const int pid = getCallingPid();
             const int self_pid = getpid();
             if (pid != self_pid) {
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index b2b65b8..28590eb 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -83,6 +83,11 @@
             /*out*/
             sp<ICamera>& device);
 
+    virtual status_t connectLegacy(const sp<ICameraClient>& cameraClient, int cameraId,
+            int halVersion, const String16& clientPackageName, int clientUid,
+            /*out*/
+            sp<ICamera>& device);
+
     virtual status_t connectPro(const sp<IProCameraCallbacks>& cameraCb,
             int cameraId, const String16& clientPackageName, int clientUid,
             /*out*/
@@ -450,7 +455,8 @@
                                       int clientUid,
                                       int callingPid,
                                       /*out*/
-                                      sp<Client>& client);
+                                      sp<Client>& client,
+                                      int halVersion = CAMERA_HAL_API_VERSION_UNSPECIFIED);
 };
 
 } // namespace android
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index 30b7bb8..517226d 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -79,7 +79,7 @@
         ALOGE("%s: Camera %d: unable to initialize device: %s (%d)",
                 __FUNCTION__, mCameraId, strerror(-res), res);
         mHardware.clear();
-        return NO_INIT;
+        return res;
     }
 
     mHardware->setCallbacks(notifyCallback,
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 8154e59..544f736 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -102,7 +102,7 @@
 status_t CameraDeviceClient::submitRequestList(List<sp<CaptureRequest> > requests,
                                                bool streaming, int64_t* lastFrameNumber) {
     ATRACE_CALL();
-    ALOGV("%s-start of function. Request list size %d", __FUNCTION__, requests.size());
+    ALOGV("%s-start of function. Request list size %zu", __FUNCTION__, requests.size());
 
     status_t res;
     if ( (res = checkPid(__FUNCTION__) ) != OK) return res;
@@ -177,7 +177,7 @@
 
         metadata.update(ANDROID_REQUEST_ID, &requestId, /*size*/1);
         loopCounter++; // loopCounter starts from 1
-        ALOGV("%s: Camera %d: Creating request with ID %d (%d of %d)",
+        ALOGV("%s: Camera %d: Creating request with ID %d (%d of %zu)",
               __FUNCTION__, mCameraId, requestId, loopCounter, requests.size());
 
         metadataRequestList.push_back(metadata);
diff --git a/services/camera/libcameraservice/device1/CameraHardwareInterface.h b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
index 87b2807..925b645 100644
--- a/services/camera/libcameraservice/device1/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/device1/CameraHardwareInterface.h
@@ -92,8 +92,22 @@
     status_t initialize(hw_module_t *module)
     {
         ALOGI("Opening camera %s", mName.string());
-        int rc = module->methods->open(module, mName.string(),
-                                       (hw_device_t **)&mDevice);
+        camera_module_t *cameraModule = reinterpret_cast<camera_module_t *>(module);
+        camera_info info;
+        status_t res = cameraModule->get_camera_info(atoi(mName.string()), &info);
+        if (res != OK) return res;
+
+        int rc = OK;
+        if (module->module_api_version >= CAMERA_MODULE_API_VERSION_2_3 &&
+            info.device_version > CAMERA_DEVICE_API_VERSION_1_0) {
+            // Open higher version camera device as HAL1.0 device.
+            rc = cameraModule->open_legacy(module, mName.string(),
+                                               CAMERA_DEVICE_API_VERSION_1_0,
+                                               (hw_device_t **)&mDevice);
+        } else {
+            rc = module->methods->open(module, mName.string(),
+                                           (hw_device_t **)&mDevice);
+        }
         if (rc != OK) {
             ALOGE("Could not open camera %s: %d", mName.string(), rc);
             return rc;