diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index e71c66e..62e2e1e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2703,7 +2703,7 @@
     }
 
     for (size_t i=0 ; i<count ; i++) {
-        sp<Track> t = mActiveTracks[i].promote();
+        const sp<Track> t = mActiveTracks[i].promote();
         if (t == 0) {
             continue;
         }
@@ -2872,11 +2872,12 @@
         // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed
         // during last round
         size_t desiredFrames;
-        if (t->sampleRate() == mSampleRate) {
+        uint32_t sr = track->sampleRate();
+        if (sr == mSampleRate) {
             desiredFrames = mNormalFrameCount;
         } else {
             // +1 for rounding and +1 for additional sample needed for interpolation
-            desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
+            desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
             // add frames already consumed but not yet released by the resampler
             // because cblk->framesReady() will include these frames
             desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 4ea1355..3e184b4 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1172,10 +1172,12 @@
             }
         }
 
+        uint32_t sr = sampleRate();
+
         // adjust the head buffer's PTS to reflect the portion of the head buffer
         // that has already been consumed
         int64_t effectivePTS = headLocalPTS +
-                ((head.position() / mFrameSize) * mLocalTimeFreq / sampleRate());
+                ((head.position() / mFrameSize) * mLocalTimeFreq / sr);
 
         // Calculate the delta in samples between the head of the input buffer
         // queue and the start of the next output buffer that will be written.
@@ -1207,7 +1209,7 @@
         // the current output position is within this threshold, then we will
         // concatenate the next input samples to the previous output
         const int64_t kSampleContinuityThreshold =
-                (static_cast<int64_t>(sampleRate()) << 32) / 250;
+                (static_cast<int64_t>(sr) << 32) / 250;
 
         // if this is the first buffer of audio that we're emitting from this track
         // then it should be almost exactly on time.