Merge tag 'android-6.0.1_r3' of https://android.googlesource.com/platform/frameworks/av into cm-13.0

Android 6.0.1 release 3

Change-Id: I2f2a1fe1b58c828e8341556996211562d6e195ab

5 files changed, 22 insertions, 5 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d7af22c..1acfaad 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1375,11 +1375,16 @@ sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId,
 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
     :   RefBase(),
         mAudioFlinger(audioFlinger),
-        // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
-        mMemoryDealer(new MemoryDealer(4100*1024, "AudioFlinger::Client")), //4MB + 1 more 4k page
         mPid(pid),
         mTimedTrackCount(0)
 {
+    size_t heapSize = kClientSharedHeapSizeBytes;
+    // Increase heap size on non low ram devices to limit risk of reconnection failure for
+    // invalidated tracks
+    if (!audioFlinger->isLowRamDevice()) {
+        heapSize *= kClientSharedHeapSizeMultiplier;
+    }
+    mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client");
 }
 
 // Client destructor must be called with AudioFlinger::mClientLock held
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 20c34ef..08fa70d 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -88,6 +88,12 @@ class ServerProxy;
 
 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
 
+
+// Max shared memory size for audio tracks and audio records per client process
+static const size_t kClientSharedHeapSizeBytes = 1024*1024;
+// Shared memory size multiplier for non low ram devices
+static const size_t kClientSharedHeapSizeMultiplier = 4;
+
 #define INCLUDING_FROM_AUDIOFLINGER_H
 
 class AudioFlinger :
@@ -423,7 +429,7 @@ private:
                             Client(const Client&);
                             Client& operator = (const Client&);
         const sp<AudioFlinger> mAudioFlinger;
-        const sp<MemoryDealer> mMemoryDealer;
+              sp<MemoryDealer> mMemoryDealer;
         const pid_t         mPid;
 
         Mutex               mTimedTrackLock;
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index 04ed08f..2493fb7 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -131,7 +131,9 @@ void FastCapture::onStateChange()
             // FIXME new may block for unbounded time at internal mutex of the heap
             //       implementation; it would be better to have normal capture thread allocate for
             //       us to avoid blocking here and to prevent possible priority inversion
-            (void)posix_memalign(&mReadBuffer, 32, frameCount * Format_frameSize(mFormat));
+            size_t bufferSize = frameCount * Format_frameSize(mFormat);
+            (void)posix_memalign(&mReadBuffer, 32, bufferSize);
+            memset(mReadBuffer, 0, bufferSize); // if posix_memalign fails, will segv here.
             mPeriodNs = (frameCount * 1000000000LL) / mSampleRate;      // 1.00
             mUnderrunNs = (frameCount * 1750000000LL) / mSampleRate;    // 1.75
             mOverrunNs = (frameCount * 500000000LL) / mSampleRate;      // 0.50
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 680ff13..e80221e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6962,6 +6962,7 @@ void AudioFlinger::RecordThread::readInputParameters_l()
     mRsmpInFrames = mFrameCount * 7;
     mRsmpInFramesP2 = roundup(mRsmpInFrames);
     free(mRsmpInBuffer);
+    mRsmpInBuffer = NULL;
 
     // TODO optimize audio capture buffer sizes ...
     // Here we calculate the size of the sliding buffer used as a source
@@ -6971,7 +6972,9 @@ void AudioFlinger::RecordThread::readInputParameters_l()
     // The current value is higher than necessary.  However it should not add to latency.
 
     // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
-    (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize);
+    size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize;
+    (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize);
+    memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here.
 
     // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
     // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 4eb13dc..f3b5375 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -717,6 +717,7 @@ status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t ev
             // But in this case we know the mixer thread (whether normal mixer or fast mixer)
             // isn't looking at this track yet:  we still hold the normal mixer thread lock,
             // and for fast tracks the track is not yet in the fast mixer thread's active set.
+            // For static tracks, this is used to acknowledge change in position or loop.
             ServerProxy::Buffer buffer;
             buffer.mFrameCount = 1;
             (void) mAudioTrackServerProxy->obtainBuffer(&buffer, true /*ackFlush*/);