diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 4c1fbd7..6c7cdde 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -66,7 +66,8 @@
 // ---------------------------------------------------------------------------
 
 AudioRecord::AudioRecord(const String16 &opPackageName)
-    : mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName), mSessionId(AUDIO_SESSION_ALLOCATE),
+    : mActive(false), mStatus(NO_INIT), mOpPackageName(opPackageName),
+      mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
       mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), mPortId(AUDIO_PORT_HANDLE_NONE)
 {
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 1908f0e..5cd2789 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -554,7 +554,8 @@
                     "channel mask %#x frameCount %zu frameCountHAL %zu deviceId %d",
                     event == AUDIO_OUTPUT_CONFIG_CHANGED ? "output" : "input",
                     ioDesc->mIoHandle, ioDesc->mSamplingRate, ioDesc->mFormat,
-                    ioDesc->mChannelMask, ioDesc->mFrameCount, ioDesc->mFrameCountHAL, ioDesc->getDeviceId());
+                    ioDesc->mChannelMask, ioDesc->mFrameCount, ioDesc->mFrameCountHAL,
+                    ioDesc->getDeviceId());
 
         } break;
         }
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index f878be9..d590cb7 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -917,7 +917,8 @@
     }
 
     // Check resampler ratios are within bounds
-    if ((uint64_t)effectiveRate > (uint64_t)mSampleRate * (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
+    if ((uint64_t)effectiveRate > (uint64_t)mSampleRate *
+            (uint64_t)AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
         ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
                 playbackRate.mSpeed, playbackRate.mPitch);
         return BAD_VALUE;
@@ -1274,9 +1275,10 @@
                                            mFlags, mSelectedDeviceId, &mPortId);
 
     if (status != NO_ERROR || output == AUDIO_IO_HANDLE_NONE) {
-        ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u, format %#x,"
-              " channel mask %#x, flags %#x",
-              mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask, mFlags);
+        ALOGE("Could not get audio output for session %d, stream type %d, usage %d, sample rate %u,"
+              " format %#x, channel mask %#x, flags %#x",
+              mSessionId, streamType, mAttributes.usage, mSampleRate, mFormat, mChannelMask,
+              mFlags);
         return BAD_VALUE;
     }
     {
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3d1f268..fb18b05 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -289,7 +289,8 @@
                                             sessionId,
                                             &streamType, client.clientUid,
                                             &fullConfig,
-                                            (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT),
+                                            (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
+                                                    AUDIO_OUTPUT_FLAG_DIRECT),
                                             *deviceId, &portId);
     } else {
         ret = AudioSystem::getInputForAttr(attr, &io,
@@ -1979,7 +1980,8 @@
                                   uint32_t *latencyMs,
                                   audio_output_flags_t flags)
 {
-    ALOGI("openOutput() this %p, module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x",
+    ALOGI("openOutput() this %p, module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, "
+              "flags %x",
               this, module,
               (devices != NULL) ? *devices : 0,
               config->sample_rate,
@@ -2077,13 +2079,13 @@
             if (mPlaybackThreads.size()) {
                 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0));
                 if (dstThread != NULL) {
-                    // audioflinger lock is held here so the acquisition order of thread locks does not
-                    // matter
+                    // audioflinger lock is held so order of thread lock acquisition doesn't matter
                     Mutex::Autolock _dl(dstThread->mLock);
                     Mutex::Autolock _sl(playbackThread->mLock);
                     Vector< sp<EffectChain> > effectChains = playbackThread->getEffectChains_l();
                     for (size_t i = 0; i < effectChains.size(); i ++) {
-                        moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(), dstThread, true);
+                        moveEffectChain_l(effectChains[i]->sessionId(), playbackThread.get(),
+                                dstThread, true);
                     }
                 }
             }
@@ -2255,7 +2257,8 @@
                                           inHwDev, inputStream,
                                           primaryOutputDevice_l(), devices, mSystemReady);
             mMmapThreads.add(*input, thread);
-            ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input, thread.get());
+            ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
+                    thread.get());
             return thread;
         } else {
 #ifdef TEE_SINK
@@ -2361,7 +2364,7 @@
                 }
             }
             if (chain != 0) {
-                // first check if a record thread is already opened with a client on the same session.
+                // first check if a record thread is already opened with a client on same session.
                 // This should only happen in case of overlap between one thread tear down and the
                 // creation of its replacement
                 size_t i;
@@ -2378,7 +2381,7 @@
                         break;
                     }
                 }
-                // put the chain aside if we could not find a record thread with the same session id.
+                // put the chain aside if we could not find a record thread with the same session id
                 if (i == mRecordThreads.size()) {
                     putOrphanEffectChain_l(chain);
                 }
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 44fd512..420f7e8 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -561,7 +561,8 @@
         virtual status_t createMmapBuffer(int32_t minSizeFrames,
                                           struct audio_mmap_buffer_info *info);
         virtual status_t getMmapPosition(struct audio_mmap_position *position);
-        virtual status_t start(const MmapStreamInterface::Client& client, audio_port_handle_t *handle);
+        virtual status_t start(const MmapStreamInterface::Client& client,
+                                         audio_port_handle_t *handle);
         virtual status_t stop(audio_port_handle_t handle);
 
     private:
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b10e42c..cf7ba43 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5599,7 +5599,7 @@
                         ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
                                 track->name());
                         tracksToRemove->add(track);
-                        // indicate to client process that the track was disabled because of underrun;
+                        // tell client process that the track was disabled because of underrun;
                         // it will then automatically call start() when data is available
                         track->disable();
                     }
@@ -7215,7 +7215,8 @@
     // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
     mRsmpInFramesOA = mRsmpInFramesP2 + mFrameCount - 1;
     (void)posix_memalign(&mRsmpInBuffer, 32, mRsmpInFramesOA * mFrameSize);
-    memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize); // if posix_memalign fails, will segv here.
+    // if posix_memalign fails, will segv here.
+    memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize);
 
     // 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?
@@ -7459,7 +7460,8 @@
     return mThread->getMmapPosition(position);
 }
 
-status_t AudioFlinger::MmapThreadHandle::start(const MmapStreamInterface::Client& client, audio_port_handle_t *handle)
+status_t AudioFlinger::MmapThreadHandle::start(const MmapStreamInterface::Client& client,
+        audio_port_handle_t *handle)
 
 {
     if (mThread == 0) {
@@ -8051,8 +8053,8 @@
         return BAD_VALUE;
     }
     if (isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
-        ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap thread",
-              desc->name);
+        ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap "
+              "thread", desc->name);
         return BAD_VALUE;
     }
 
@@ -8284,7 +8286,8 @@
 {
     MmapThread::dumpInternals(fd, args);
 
-    dprintf(fd, "  Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n", mStreamType, mStreamVolume, mHalVolFloat, mStreamMute);
+    dprintf(fd, "  Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
+            mStreamType, mStreamVolume, mHalVolFloat, mStreamMute);
     dprintf(fd, "  Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute);
 }