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| author | Xin Li <delphij@google.com> | 2020-09-08 16:55:24 -0700 |
|---|---|---|
| committer | Xin Li <delphij@google.com> | 2020-09-08 16:55:24 -0700 |
| commit | 56ed6e9c62ab59db06f6aaf165425cc0db808fa8 (patch) | |
| tree | 27d8378cdb568e851bc7790007133952fb31309a | |
| parent | 839659d9fd4df3594ac2f0526bb0a9ea8bee256f (diff) | |
| parent | aee82e345d060cac6439991cce02292a8ffeba0f (diff) | |
| download | voip-56ed6e9c62ab59db06f6aaf165425cc0db808fa8.tar.gz | |
Merge Android R
Bug: 168057903
Merged-In: I02ad49d6b2c3344b90108ee0933a82bd2244dfd8
Change-Id: I3280e5fc715ebf1fd16e52466236a56f8fb1197c
| -rw-r--r-- | src/java/com/android/server/sip/SipService.java | 4 | ||||
| -rw-r--r-- | src/jni/rtp/Android.bp | 1 | ||||
| -rw-r--r-- | src/jni/rtp/AudioGroup.cpp | 4 |
3 files changed, 7 insertions, 2 deletions
diff --git a/src/java/com/android/server/sip/SipService.java b/src/java/com/android/server/sip/SipService.java index 7c6daf8..ca91943 100644 --- a/src/java/com/android/server/sip/SipService.java +++ b/src/java/com/android/server/sip/SipService.java @@ -472,8 +472,8 @@ public final class SipService extends ISipService.Stub { mContext.enforceCallingOrSelfPermission( android.Manifest.permission.USE_SIP, message); - return mAppOps.noteOp(AppOpsManager.OP_USE_SIP, Binder.getCallingUid(), - packageName) == AppOpsManager.MODE_ALLOWED; + return mAppOps.noteOp(AppOpsManager.OPSTR_USE_SIP, Binder.getCallingUid(), + packageName, null, message) == AppOpsManager.MODE_ALLOWED; } private class SipSessionGroupExt extends SipSessionAdapter { diff --git a/src/jni/rtp/Android.bp b/src/jni/rtp/Android.bp index c938224..b6c8588 100644 --- a/src/jni/rtp/Android.bp +++ b/src/jni/rtp/Android.bp @@ -28,6 +28,7 @@ cc_library_shared { ], header_libs: [ "libmedia_headers", + "libmediametrics_headers", ], shared_libs: [ diff --git a/src/jni/rtp/AudioGroup.cpp b/src/jni/rtp/AudioGroup.cpp index 64e50f8..5f9cae8 100644 --- a/src/jni/rtp/AudioGroup.cpp +++ b/src/jni/rtp/AudioGroup.cpp @@ -819,6 +819,10 @@ bool AudioGroup::DeviceThread::threadLoop() // Initialize AudioTrack and AudioRecord. sp<AudioTrack> track = new AudioTrack(); sp<AudioRecord> record = new AudioRecord(mGroup->mOpPackageName); + // Set caller name so it can be logged in destructor. + // MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_RTP + track->setCallerName("rtp"); + record->setCallerName("rtp"); if (track->set(AUDIO_STREAM_VOICE_CALL, sampleRate, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO, output, AUDIO_OUTPUT_FLAG_NONE, NULL /*callback_t*/, NULL /*user*/, 0 /*notificationFrames*/, 0 /*sharedBuffer*/, |