No newline or space at end of ALOG format string

Change-Id: I0bef580cbc818cb7c87aea23919d26f1446cec32

3 files changed, 23 insertions, 23 deletions

diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 93c91fb..d5d1b6c 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1920,7 +1920,7 @@ bool AudioFlinger::MixerThread::threadLoop()
             if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
                         mSuspended)) {
                 if (!mStandby) {
-                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+                    ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended);
                     mOutput->stream->common.standby(&mOutput->stream->common);
                     mStandby = true;
                     mBytesWritten = 0;
@@ -1934,9 +1934,9 @@ bool AudioFlinger::MixerThread::threadLoop()
 
                     releaseWakeLock_l();
                     // wait until we have something to do...
-                    ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+                    ALOGV("MixerThread %p TID %d going to sleep", this, gettid());
                     mWaitWorkCV.wait(mLock);
-                    ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
+                    ALOGV("MixerThread %p TID %d waking up", this, gettid());
                     acquireWakeLock_l();
 
                     mPrevMixerStatus = MIXER_IDLE;
@@ -2638,7 +2638,7 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
                         mSuspended)) {
                 // wait until we have something to do...
                 if (!mStandby) {
-                    ALOGV("Audio hardware entering standby, mixer %p\n", this);
+                    ALOGV("Audio hardware entering standby, mixer %p", this);
                     mOutput->stream->common.standby(&mOutput->stream->common);
                     mStandby = true;
                     mBytesWritten = 0;
@@ -2651,9 +2651,9 @@ bool AudioFlinger::DirectOutputThread::threadLoop()
                     if (exitPending()) break;
 
                     releaseWakeLock_l();
-                    ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+                    ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid());
                     mWaitWorkCV.wait(mLock);
-                    ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+                    ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid());
                     acquireWakeLock_l();
 
                     if (!mMasterMute) {
@@ -3046,9 +3046,9 @@ bool AudioFlinger::DuplicatingThread::threadLoop()
                     if (exitPending()) break;
 
                     releaseWakeLock_l();
-                    ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+                    ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid());
                     mWaitWorkCV.wait(mLock);
-                    ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+                    ALOGV("DuplicatingThread %p TID %d waking up", this, gettid());
                     acquireWakeLock_l();
 
                     mPrevMixerStatus = MIXER_IDLE;
@@ -6209,7 +6209,7 @@ sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
 
 void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
 {
-    ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
+    ALOGV("disconnect() %p handle %p", this, handle.unsafe_get());
     // keep a strong reference on this EffectModule to avoid calling the
     // destructor before we exit
     sp<EffectModule> keep(this);
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 6e17a4a..4eac032 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -184,7 +184,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
     size_t outputSampleCount = outFrameCount * 2;
     size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
-    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
 
     while (outputIndex < outputSampleCount) {
@@ -197,7 +197,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
                 goto resampleStereo16_exit;
             }
 
-            // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
             if (mBuffer.frameCount > inputIndex) break;
 
             inputIndex -= mBuffer.frameCount;
@@ -211,7 +211,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
 
         // handle boundary case
         while (inputIndex == 0) {
-            // ALOGE("boundary case\n");
+            // ALOGE("boundary case");
             out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
             out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
@@ -220,7 +220,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
         }
 
         // process input samples
-        // ALOGE("general case\n");
+        // ALOGE("general case");
 
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
         if (inputIndex + 2 < mBuffer.frameCount) {
@@ -242,7 +242,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
             Advance(&inputIndex, &phaseFraction, phaseIncrement);
         }
 
-        // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
 
         // if done with buffer, save samples
         if (inputIndex >= mBuffer.frameCount) {
@@ -259,7 +259,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount,
         }
     }
 
-    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
 
 resampleStereo16_exit:
     // save state
@@ -280,7 +280,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
     size_t outputSampleCount = outFrameCount * 2;
     size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
 
-    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
     //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
     while (outputIndex < outputSampleCount) {
         // buffer is empty, fetch a new one
@@ -292,7 +292,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
                 mPhaseFraction = phaseFraction;
                 goto resampleMono16_exit;
             }
-            // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+            // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
             if (mBuffer.frameCount >  inputIndex) break;
 
             inputIndex -= mBuffer.frameCount;
@@ -304,7 +304,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
 
         // handle boundary case
         while (inputIndex == 0) {
-            // ALOGE("boundary case\n");
+            // ALOGE("boundary case");
             int32_t sample = Interp(mX0L, in[0], phaseFraction);
             out[outputIndex++] += vl * sample;
             out[outputIndex++] += vr * sample;
@@ -314,7 +314,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         }
 
         // process input samples
-        // ALOGE("general case\n");
+        // ALOGE("general case");
 
 #ifdef ASM_ARM_RESAMP1  // asm optimisation for ResamplerOrder1
         if (inputIndex + 2 < mBuffer.frameCount) {
@@ -337,7 +337,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         }
 
 
-        // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+        // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
 
         // if done with buffer, save samples
         if (inputIndex >= mBuffer.frameCount) {
@@ -353,7 +353,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount,
         }
     }
 
-    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+    // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
 
 resampleMono16_exit:
     // save state
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 47205ba..c0e760e 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -99,7 +99,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount,
                 if (mBuffer.raw == NULL)
                     goto save_state;  // ugly, but efficient
                 in = mBuffer.i16;
-                // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+                // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
             }
 
             // advance sample state
@@ -133,7 +133,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount,
         provider->getNextBuffer(&mBuffer);
         if (mBuffer.raw == NULL)
             return;
-        // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+        // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
     }
     int16_t *in = mBuffer.i16;