rename audio policy output flags

Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 2596f07..4c41ba5 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -589,7 +589,7 @@
                                     uint32_t samplingRate,
                                     audio_format_t format,
                                     uint32_t channels,
-                                    audio_policy_output_flags_t flags)
+                                    audio_output_flags_t flags)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return 0;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 6dc6c41..092b516 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -92,7 +92,7 @@
         audio_format_t format,
         int channelMask,
         int frameCount,
-        audio_policy_output_flags_t flags,
+        audio_output_flags_t flags,
         callback_t cbf,
         void* user,
         int notificationFrames,
@@ -124,7 +124,7 @@
       mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
 {
     mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
-            frameCount, (audio_policy_output_flags_t)flags, cbf, user, notificationFrames,
+            frameCount, (audio_output_flags_t)flags, cbf, user, notificationFrames,
             0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId);
 }
 
@@ -134,7 +134,7 @@
         audio_format_t format,
         int channelMask,
         const sp<IMemory>& sharedBuffer,
-        audio_policy_output_flags_t flags,
+        audio_output_flags_t flags,
         callback_t cbf,
         void* user,
         int notificationFrames,
@@ -174,7 +174,7 @@
         audio_format_t format,
         int channelMask,
         int frameCount,
-        audio_policy_output_flags_t flags,
+        audio_output_flags_t flags,
         callback_t cbf,
         void* user,
         int notificationFrames,
@@ -222,8 +222,8 @@
 
     // force direct flag if format is not linear PCM
     if (!audio_is_linear_pcm(format)) {
-        flags = (audio_policy_output_flags_t)
-                ((flags | AUDIO_POLICY_OUTPUT_FLAG_DIRECT) & ~AUDIO_POLICY_OUTPUT_FLAG_FAST);
+        flags = (audio_output_flags_t)
+                ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
     }
 
     if (!audio_is_output_channel(channelMask)) {
@@ -735,7 +735,7 @@
         audio_format_t format,
         uint32_t channelMask,
         int frameCount,
-        audio_policy_output_flags_t flags,
+        audio_output_flags_t flags,
         const sp<IMemory>& sharedBuffer,
         audio_io_handle_t output)
 {
@@ -761,14 +761,14 @@
 
     // Client decides whether the track is TIMED (see below), but can only express a preference
     // for FAST.  Server will perform additional tests.
-    if ((flags & AUDIO_POLICY_OUTPUT_FLAG_FAST) && !(
+    if ((flags & AUDIO_OUTPUT_FLAG_FAST) && !(
             // either of these use cases:
             // use case 1: shared buffer
             (sharedBuffer != 0) ||
             // use case 2: callback handler
             (mCbf != NULL))) {
-        ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied");
-        flags = (audio_policy_output_flags_t) (flags & ~AUDIO_POLICY_OUTPUT_FLAG_FAST);
+        ALOGW("AUDIO_OUTPUT_FLAG_FAST denied");
+        flags = (audio_output_flags_t) (flags & ~AUDIO_OUTPUT_FLAG_FAST);
     }
     ALOGV("createTrack_l() output %d afFrameCount %d afLatency %d", output, afFrameCount, afLatency);
 
@@ -796,7 +796,7 @@
             if (mNotificationFramesAct > (uint32_t)frameCount/2) {
                 mNotificationFramesAct = frameCount/2;
             }
-            if (frameCount < minFrameCount && !(flags & AUDIO_POLICY_OUTPUT_FLAG_FAST)) {
+            if (frameCount < minFrameCount && !(flags & AUDIO_OUTPUT_FLAG_FAST)) {
                 // not ALOGW because it happens all the time when playing key clicks over A2DP
                 ALOGV("Minimum buffer size corrected from %d to %d",
                          frameCount, minFrameCount);
@@ -817,7 +817,7 @@
     if (mIsTimed) {
         trackFlags |= IAudioFlinger::TRACK_TIMED;
     }
-    if (flags & AUDIO_POLICY_OUTPUT_FLAG_FAST) {
+    if (flags & AUDIO_OUTPUT_FLAG_FAST) {
         trackFlags |= IAudioFlinger::TRACK_FAST;
     }
 
@@ -1033,7 +1033,7 @@
 
         size_t toWrite;
 
-        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
+        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
             // Divide capacity by 2 to take expansion into account
             toWrite = audioBuffer.size>>1;
             memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
@@ -1190,7 +1190,7 @@
         // Divide buffer size by 2 to take into account the expansion
         // due to 8 to 16 bit conversion: the callback must fill only half
         // of the destination buffer
-        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
+        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
             audioBuffer.size >>= 1;
         }
 
@@ -1209,7 +1209,7 @@
         }
         if (writtenSize > reqSize) writtenSize = reqSize;
 
-        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
+        if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
             // 8 to 16 bit conversion, note that source and destination are the same address
             memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
             writtenSize <<= 1;
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index 81e259a..2b5126f 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -362,7 +362,7 @@
                                          audio_format_t *pFormat,
                                          audio_channel_mask_t *pChannelMask,
                                          uint32_t *pLatencyMs,
-                                         audio_policy_output_flags_t flags)
+                                         audio_output_flags_t flags)
     {
         Parcel data, reply;
         audio_devices_t devices = pDevices ? *pDevices : (audio_devices_t)0;
@@ -855,7 +855,7 @@
             audio_format_t format = (audio_format_t) data.readInt32();
             audio_channel_mask_t channelMask = (audio_channel_mask_t)data.readInt32();
             uint32_t latency = data.readInt32();
-            audio_policy_output_flags_t flags = (audio_policy_output_flags_t) data.readInt32();
+            audio_output_flags_t flags = (audio_output_flags_t) data.readInt32();
             audio_io_handle_t output = openOutput(module,
                                                  &devices,
                                                  &samplingRate,
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 5040bd9..7aab8d6 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -124,7 +124,7 @@
                                         uint32_t samplingRate,
                                         audio_format_t format,
                                         uint32_t channels,
-                                        audio_policy_output_flags_t flags)
+                                        audio_output_flags_t flags)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -418,8 +418,8 @@
             uint32_t samplingRate = data.readInt32();
             audio_format_t format = (audio_format_t) data.readInt32();
             uint32_t channels = data.readInt32();
-            audio_policy_output_flags_t flags =
-                    static_cast <audio_policy_output_flags_t>(data.readInt32());
+            audio_output_flags_t flags =
+                    static_cast <audio_output_flags_t>(data.readInt32());
 
             audio_io_handle_t output = getOutput(stream,
                                                  samplingRate,
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 52aee49..59e538f 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -94,7 +94,7 @@
             AUDIO_FORMAT_PCM_16_BIT,
             audio_channel_out_mask_from_count(pLibConfig->numChannels),
             mTrackBufferSize,
-            AUDIO_POLICY_OUTPUT_FLAG_NONE);
+            AUDIO_OUTPUT_FLAG_NONE);
 
     // create render and playback thread
     {
diff --git a/media/libmedia/SoundPool.cpp b/media/libmedia/SoundPool.cpp
index 3b7b96a..4b318ed 100644
--- a/media/libmedia/SoundPool.cpp
+++ b/media/libmedia/SoundPool.cpp
@@ -608,10 +608,10 @@
         // do not create a new audio track if current track is compatible with sample parameters
 #ifdef USE_SHARED_MEM_BUFFER
         newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
-                channels, sample->getIMemory(), AUDIO_POLICY_OUTPUT_FLAG_NONE, callback, userData);
+                channels, sample->getIMemory(), AUDIO_OUTPUT_FLAG_NONE, callback, userData);
 #else
         newTrack = new AudioTrack(streamType, sampleRate, sample->format(),
-                channels, frameCount, AUDIO_POLICY_OUTPUT_FLAG_NONE, callback, userData,
+                channels, frameCount, AUDIO_OUTPUT_FLAG_NONE, callback, userData,
                 bufferFrames);
 #endif
         oldTrack = mAudioTrack;
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index c93b6fd..253602d 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -1024,7 +1024,7 @@
                       AUDIO_FORMAT_PCM_16_BIT,
                       AUDIO_CHANNEL_OUT_MONO,
                       0,    // frameCount
-                      AUDIO_POLICY_OUTPUT_FLAG_FAST,
+                      AUDIO_OUTPUT_FLAG_FAST,
                       audioCallback,
                       this, // user
                       0,    // notificationFrames