Miscellaneous code cleanup in audio framework

Changes:
 - Move declaration of kClassPathName to top of file so it can be used
   in more than one place, instead of "android/media/AudioSystem".
 - Make private methods static.
 - Add comment to stream_type, audio_mode, force_use types that they must match
   values in AudioSystem.java.
 - Add comment about unused types mp3_sub_format and vorbis_sub_format.
 - Fix typos.
 - Use @ in javadoc comments.
 - Delete dead APIs setMode, getMode, setRouting, getRouting in AudioSystem.java
   (they are all hidden, deprecated, and unused by rest of framework)
 - Delete unused private log method.
 - Fix pathname for android_media_AudioSystem.cpp.
 - Improve code formatting for space after == and !=.
 - Add logging of delta for changing audio policy manager ref count.

Change-Id: I18037c7beb8ab76d1fda08c11e589f6e591d36e1
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 2dc4beb..edf4b8b 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -33,6 +33,7 @@
 {
 public:
 
+    // must match android/media/AudioSystem.java STREAM_* constants
     enum stream_type {
         DEFAULT          =-1,
         VOICE_CALL       = 0,
@@ -54,6 +55,8 @@
         PCM_SUB_8_BIT           = 0x2, // must be 2 for backward compatibility
     };
 
+    // FIXME These sub_format enums are currently unused
+
     // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify
     // bit rate, stereo mode, version...
     enum mp3_sub_format {
@@ -100,7 +103,7 @@
     };
 
 
-    // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java
+    // Channel mask definitions must be kept in sync with values in /media/java/android/media/AudioFormat.java
     enum audio_channels {
         // output channels
         CHANNEL_OUT_FRONT_LEFT = 0x4,
@@ -150,6 +153,7 @@
                 CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK)
     };
 
+    // must match android/media/AudioSystem.java MODE_* values
     enum audio_mode {
         MODE_INVALID = -2,
         MODE_CURRENT = -1,
@@ -189,6 +193,7 @@
     // set/get master volume
     static status_t setMasterVolume(float value);
     static status_t getMasterVolume(float* volume);
+
     // mute/unmute audio outputs
     static status_t setMasterMute(bool mute);
     static status_t getMasterMute(bool* mute);
@@ -234,7 +239,7 @@
     static status_t setVoiceVolume(float volume);
 
     // return the number of audio frames written by AudioFlinger to audio HAL and
-    // audio dsp to DAC since the output on which the specificed stream is playing
+    // audio dsp to DAC since the output on which the specified stream is playing
     // has exited standby.
     // returned status (from utils/Errors.h) can be:
     // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data
@@ -321,7 +326,7 @@
         FORCE_DEFAULT = FORCE_NONE
     };
 
-    // usages used for setForceUse()
+    // usages used for setForceUse(), must match AudioSystem.java
     enum force_use {
         FOR_COMMUNICATION,
         FOR_MEDIA,
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 2702242..52135e7 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -5712,7 +5712,7 @@
         const uint32_t i = 31 - __builtin_clz(device);
         device &= ~(1 << i);
         if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) {
-            LOGE("device convertion error for AudioSystem device 0x%08x", device);
+            LOGE("device conversion error for AudioSystem device 0x%08x", device);
             return 0;
         }
         deviceOut |= (uint32_t)sDeviceConvTable[i];
diff --git a/services/audioflinger/AudioPolicyManagerBase.cpp b/services/audioflinger/AudioPolicyManagerBase.cpp
index f653dc5..c0ac669 100644
--- a/services/audioflinger/AudioPolicyManagerBase.cpp
+++ b/services/audioflinger/AudioPolicyManagerBase.cpp
@@ -542,7 +542,7 @@
     }
 
 
-    LOGW_IF((output ==0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
+    LOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d, format %d, channels %x, flags %x",
                 stream, samplingRate, format, channels, flags);
 
     return output;
@@ -2107,7 +2107,7 @@
                                     uint32_t device)
 {
    return ((flags & AudioSystem::OUTPUT_FLAG_DIRECT) ||
-          (format !=0 && !AudioSystem::isLinearPCM(format)));
+          (format != 0 && !AudioSystem::isLinearPCM(format)));
 }
 
 uint32_t AudioPolicyManagerBase::getMaxEffectsCpuLoad()
@@ -2159,7 +2159,7 @@
         return;
     }
     mRefCount[stream] += delta;
-    LOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+    LOGV("changeRefCount() delta %d, stream %d, refCount %d", delta, stream, mRefCount[stream]);
 }
 
 uint32_t AudioPolicyManagerBase::AudioOutputDescriptor::refCount()
@@ -2215,7 +2215,8 @@
 
 AudioPolicyManagerBase::AudioInputDescriptor::AudioInputDescriptor()
     : mSamplingRate(0), mFormat(0), mChannels(0),
-     mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0)
+      mAcoustics((AudioSystem::audio_in_acoustics)0), mDevice(0), mRefCount(0),
+      mInputSource(0)
 {
 }