Remove dead code

mFormat is unused in resampler
mClientTid is unused
local variable pid is unused in dump

Change-Id: Ib156e38029366620bfeff2a13e73471867155a5b
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f71ba0a..fbfb0ab 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -3259,7 +3259,6 @@
         // mBufferEnd
         mFrameCount(0),
         mState(IDLE),
-        mClientTid(-1),
         mFormat(format),
         mFlags(flags & ~SYSTEM_FLAGS_MASK),
         mSessionId(sessionId)
@@ -4631,7 +4630,6 @@
     const size_t SIZE = 256;
     char buffer[SIZE];
     String8 result;
-    pid_t pid = 0;
 
     snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
     result.append(buffer);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 3f3188c..fb38404 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -376,7 +376,6 @@
             uint32_t            mFrameCount;
             // we don't really need a lock for these
             track_state         mState;
-            int                 mClientTid;
             const audio_format_t mFormat;
             uint32_t            mFlags;
             const int           mSessionId;
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index feacd96..6e17a4a3 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -130,12 +130,6 @@
     mVolume[0] = mVolume[1] = 0;
     mBuffer.frameCount = 0;
 
-    // save format for quick lookup
-    if (inChannelCount == 1) {
-        mFormat = MONO_16_BIT;
-    } else {
-        mFormat = STEREO_16_BIT;
-    }
 }
 
 AudioResampler::~AudioResampler() {
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index ffa690a..e57e2e9 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -66,7 +66,6 @@
     // multiplier to calculate fixed point phase increment
     static const double kPhaseMultiplier = 1L << kNumPhaseBits;
 
-    enum format {MONO_16_BIT, STEREO_16_BIT};
     AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
 
     // prevent copying
@@ -83,7 +82,6 @@
         uint32_t mVolumeRL;
     };
     int16_t mTargetVolume[2];
-    format mFormat;
     size_t mInputIndex;
     int32_t mPhaseIncrement;
     uint32_t mPhaseFraction;