Upintegrate Audio Flinger changes from ICS_AAH

Bring in changes to audio flinger made to support timed audio tracks
and HW master volume control.

Change-Id: Ide52d48809bdbed13acf35fd59b24637e35064ae
Signed-off-by: John Grossman <johngro@google.com>
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index aead9a1..74c97ed 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -80,7 +80,9 @@
 
 AudioTrack::AudioTrack()
     : mStatus(NO_INIT),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
+      mIsTimed(false),
+      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+      mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
 {
 }
 
@@ -96,7 +98,9 @@
         int notificationFrames,
         int sessionId)
     : mStatus(NO_INIT),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
+      mIsTimed(false),
+      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+      mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
@@ -134,7 +138,9 @@
         int notificationFrames,
         int sessionId)
     : mStatus(NO_INIT),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
+      mIsTimed(false),
+      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+      mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
 {
     mStatus = set(streamType, sampleRate, format, channelMask,
             0, flags, cbf, user, notificationFrames,
@@ -540,6 +546,10 @@
 {
     int afSamplingRate;
 
+    if (mIsTimed) {
+        return INVALID_OPERATION;
+    }
+
     if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
         return NO_INIT;
     }
@@ -553,6 +563,10 @@
 
 uint32_t AudioTrack::getSampleRate() const
 {
+    if (mIsTimed) {
+        return INVALID_OPERATION;
+    }
+
     AutoMutex lock(mLock);
     return mCblk->sampleRate;
 }
@@ -578,6 +592,10 @@
         return NO_ERROR;
     }
 
+    if (mIsTimed) {
+        return INVALID_OPERATION;
+    }
+
     if (loopStart >= loopEnd ||
         loopEnd - loopStart > cblk->frameCount ||
         cblk->server > loopStart) {
@@ -641,6 +659,8 @@
 
 status_t AudioTrack::setPosition(uint32_t position)
 {
+    if (mIsTimed) return INVALID_OPERATION;
+
     AutoMutex lock(mLock);
 
     if (!stopped_l()) return INVALID_OPERATION;
@@ -791,6 +811,7 @@
                                                       ((uint16_t)flags) << 16,
                                                       sharedBuffer,
                                                       output,
+                                                      mIsTimed,
                                                       &mSessionId,
                                                       &status);
 
@@ -957,6 +978,7 @@
 {
 
     if (mSharedBuffer != 0) return INVALID_OPERATION;
+    if (mIsTimed) return INVALID_OPERATION;
 
     if (ssize_t(userSize) < 0) {
         // Sanity-check: user is most-likely passing an error code, and it would
@@ -1013,6 +1035,59 @@
 
 // -------------------------------------------------------------------------
 
+TimedAudioTrack::TimedAudioTrack() {
+    mIsTimed = true;
+}
+
+status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
+{
+    status_t result = UNKNOWN_ERROR;
+
+    // If the track is not invalid already, try to allocate a buffer.  alloc
+    // fails indicating that the server is dead, flag the track as invalid so
+    // we can attempt to restore in in just a bit.
+    if (!(mCblk->flags & CBLK_INVALID_MSK)) {
+        result = mAudioTrack->allocateTimedBuffer(size, buffer);
+        if (result == DEAD_OBJECT) {
+            android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
+        }
+    }
+
+    // If the track is invalid at this point, attempt to restore it. and try the
+    // allocation one more time.
+    if (mCblk->flags & CBLK_INVALID_MSK) {
+        mCblk->lock.lock();
+        result = restoreTrack_l(mCblk, false);
+        mCblk->lock.unlock();
+
+        if (result == OK)
+            result = mAudioTrack->allocateTimedBuffer(size, buffer);
+    }
+
+    return result;
+}
+
+status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
+                                           int64_t pts)
+{
+    // restart track if it was disabled by audioflinger due to previous underrun
+    if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
+        android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
+        ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
+        mAudioTrack->start(0);
+    }
+
+    return mAudioTrack->queueTimedBuffer(buffer, pts);
+}
+
+status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
+                                                TargetTimeline target)
+{
+    return mAudioTrack->setMediaTimeTransform(xform, target);
+}
+
+// -------------------------------------------------------------------------
+
 bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
 {
     Buffer audioBuffer;
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 4507e5d..ebadbfa 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -90,6 +90,7 @@
                                 uint32_t flags,
                                 const sp<IMemory>& sharedBuffer,
                                 audio_io_handle_t output,
+                                bool isTimed,
                                 int *sessionId,
                                 status_t *status)
     {
@@ -105,6 +106,7 @@
         data.writeInt32(flags);
         data.writeStrongBinder(sharedBuffer->asBinder());
         data.writeInt32((int32_t) output);
+        data.writeInt32(isTimed);
         int lSessionId = 0;
         if (sessionId != NULL) {
             lSessionId = *sessionId;
@@ -689,11 +691,12 @@
             uint32_t flags = data.readInt32();
             sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
             audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+            bool isTimed = data.readInt32();
             int sessionId = data.readInt32();
             status_t status;
             sp<IAudioTrack> track = createTrack(pid,
                     (audio_stream_type_t) streamType, sampleRate, format,
-                    channelCount, bufferCount, flags, buffer, output, &sessionId, &status);
+                    channelCount, bufferCount, flags, buffer, output, isTimed, &sessionId, &status);
             reply->writeInt32(sessionId);
             reply->writeInt32(status);
             reply->writeStrongBinder(track->asBinder());
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index a7958de..28ebbbf 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -35,7 +35,10 @@
     FLUSH,
     MUTE,
     PAUSE,
-    ATTACH_AUX_EFFECT
+    ATTACH_AUX_EFFECT,
+    ALLOCATE_TIMED_BUFFER,
+    QUEUE_TIMED_BUFFER,
+    SET_MEDIA_TIME_TRANSFORM,
 };
 
 class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -114,6 +117,52 @@
         }
         return status;
     }
+
+    virtual status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        data.writeInt32(size);
+        status_t status = remote()->transact(ALLOCATE_TIMED_BUFFER,
+                                             data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+            if (status == NO_ERROR) {
+                *buffer = interface_cast<IMemory>(reply.readStrongBinder());
+            }
+        }
+        return status;
+    }
+
+    virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+                                      int64_t pts) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        data.writeStrongBinder(buffer->asBinder());
+        data.writeInt64(pts);
+        status_t status = remote()->transact(QUEUE_TIMED_BUFFER,
+                                             data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+        }
+        return status;
+    }
+
+    virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+                                           int target) {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+        data.writeInt64(xform.a_zero);
+        data.writeInt64(xform.b_zero);
+        data.writeInt32(xform.a_to_b_numer);
+        data.writeInt32(xform.a_to_b_denom);
+        data.writeInt32(target);
+        status_t status = remote()->transact(SET_MEDIA_TIME_TRANSFORM,
+                                             data, &reply);
+        if (status == NO_ERROR) {
+            status = reply.readInt32();
+        }
+        return status;
+    }
 };
 
 IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -159,10 +208,38 @@
             reply->writeInt32(attachAuxEffect(data.readInt32()));
             return NO_ERROR;
         } break;
+        case ALLOCATE_TIMED_BUFFER: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            sp<IMemory> buffer;
+            status_t status = allocateTimedBuffer(data.readInt32(), &buffer);
+            reply->writeInt32(status);
+            if (status == NO_ERROR) {
+                reply->writeStrongBinder(buffer->asBinder());
+            }
+            return NO_ERROR;
+        } break;
+        case QUEUE_TIMED_BUFFER: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            sp<IMemory> buffer = interface_cast<IMemory>(
+                data.readStrongBinder());
+            uint64_t pts = data.readInt64();
+            reply->writeInt32(queueTimedBuffer(buffer, pts));
+            return NO_ERROR;
+        } break;
+        case SET_MEDIA_TIME_TRANSFORM: {
+            CHECK_INTERFACE(IAudioTrack, data, reply);
+            LinearTransform xform;
+            xform.a_zero = data.readInt64();
+            xform.b_zero = data.readInt64();
+            xform.a_to_b_numer = data.readInt32();
+            xform.a_to_b_denom = data.readInt32();
+            int target = data.readInt32();
+            reply->writeInt32(setMediaTimeTransform(xform, target));
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
 }
 
 }; // namespace android
-