Snap for 7550640 from 15bb2977b870495211117548ea132810e8cff7c9 to mainline-wifi-releaseandroid-mainline-12.0.0_r96android-mainline-12.0.0_r83android-mainline-12.0.0_r67android-mainline-12.0.0_r57android-mainline-12.0.0_r40android-mainline-12.0.0_r20android-mainline-12.0.0_r126android-mainline-12.0.0_r114aml_wif_311811030android12-mainline-wifi-release

Change-Id: Iee2351a0600ba1f3098864aa4d30f14921ff6cf8

15 files changed, 75 insertions, 33 deletions

diff --git a/Android.bp b/Android.bp
index 7f2d193..5295c5e 100644
--- a/Android.bp
+++ b/Android.bp
@@ -12,6 +12,10 @@
 // See the License for the specific language governing permissions and
 // limitations under the License.
 
+package {
+    default_applicable_licenses: ["Android-Apache-2.0"],
+}
+
 filegroup {
     name: "opt-net-voip-srcs",
     srcs: [
diff --git a/src/java/android/net/rtp/AudioCodec.java b/src/java/android/net/rtp/AudioCodec.java
index 85255c8..9cae573 100644
--- a/src/java/android/net/rtp/AudioCodec.java
+++ b/src/java/android/net/rtp/AudioCodec.java
@@ -33,6 +33,9 @@ import java.util.Arrays;
  * </pre>
  *
  * @see AudioStream
+ *
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer
+ * supported and should not be used as the basis of future VOIP apps.
  */
 public class AudioCodec {
     /**
diff --git a/src/java/android/net/rtp/AudioGroup.java b/src/java/android/net/rtp/AudioGroup.java
index 9d96006..3a85bbc 100644
--- a/src/java/android/net/rtp/AudioGroup.java
+++ b/src/java/android/net/rtp/AudioGroup.java
@@ -17,9 +17,10 @@
 package android.net.rtp;
 
 import android.annotation.NonNull;
-import android.app.ActivityThread;
+import android.content.AttributionSource;
 import android.content.Context;
 import android.media.AudioManager;
+import android.os.Parcel;
 
 import java.util.HashMap;
 import java.util.Locale;
@@ -63,6 +64,9 @@ import java.util.Map;
  * the AudioGroups is in use.</p>
  *
  * @see AudioStream
+ *
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class AudioGroup {
     /**
@@ -114,7 +118,7 @@ public class AudioGroup {
 
     /**
      * Creates an empty AudioGroup.
-     * @param context Context used to get package name
+     * @param context Context the group belongs to
      */
     public AudioGroup(@NonNull Context context) {
         mContext = context;
@@ -164,11 +168,15 @@ public class AudioGroup {
                 AudioCodec codec = stream.getCodec();
                 String codecSpec = String.format(Locale.US, "%d %s %s", codec.type,
                         codec.rtpmap, codec.fmtp);
-                long id = nativeAdd(stream.getMode(), stream.getSocket(),
-                        stream.getRemoteAddress().getHostAddress(),
-                        stream.getRemotePort(), codecSpec, stream.getDtmfType(),
-                        mContext != null ? mContext.getOpPackageName()
-                                : ActivityThread.currentOpPackageName());
+
+                final long id;
+                try (AttributionSource.ScopedParcelState attributionSourceState = mContext
+                        .getAttributionSource().asScopedParcelState()) {
+                    id = nativeAdd(stream.getMode(), stream.getSocket(),
+                            stream.getRemoteAddress().getHostAddress(),
+                            stream.getRemotePort(), codecSpec, stream.getDtmfType(),
+                            attributionSourceState.getParcel());
+                }
                 mStreams.put(stream, id);
             } catch (NullPointerException e) {
                 throw new IllegalStateException(e);
@@ -176,8 +184,8 @@ public class AudioGroup {
         }
     }
 
-    private native long nativeAdd(int mode, int socket, String remoteAddress,
-            int remotePort, String codecSpec, int dtmfType, String opPackageName);
+    private native long nativeAdd(int mode, int socket, String remoteAddress, int remotePort,
+            String codecSpec, int dtmfType, Parcel attributionSource);
 
     // Package-private method used by AudioStream.join().
     synchronized void remove(AudioStream stream) {
diff --git a/src/java/android/net/rtp/AudioStream.java b/src/java/android/net/rtp/AudioStream.java
index 5cd1abc..e746e7d 100644
--- a/src/java/android/net/rtp/AudioStream.java
+++ b/src/java/android/net/rtp/AudioStream.java
@@ -41,6 +41,8 @@ import java.net.SocketException;
  *
  * @see RtpStream
  * @see AudioGroup
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class AudioStream extends RtpStream {
     private AudioCodec mCodec;
diff --git a/src/java/android/net/rtp/RtpStream.java b/src/java/android/net/rtp/RtpStream.java
index b9d75cd..ad37455 100644
--- a/src/java/android/net/rtp/RtpStream.java
+++ b/src/java/android/net/rtp/RtpStream.java
@@ -27,6 +27,8 @@ import java.net.SocketException;
  *
  * <p class="note">Using this class requires
  * {@link android.Manifest.permission#INTERNET} permission.</p>
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class RtpStream {
     /**
diff --git a/src/java/android/net/sip/SipAudioCall.java b/src/java/android/net/sip/SipAudioCall.java
index 363995c..4d70017 100644
--- a/src/java/android/net/sip/SipAudioCall.java
+++ b/src/java/android/net/sip/SipAudioCall.java
@@ -56,6 +56,8 @@ import java.net.UnknownHostException;
  * <a href="{@docRoot}guide/topics/network/sip.html">Session Initiation Protocol</a>
  * developer guide.</p>
  * </div>
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class SipAudioCall {
     private static final String LOG_TAG = SipAudioCall.class.getSimpleName();
diff --git a/src/java/android/net/sip/SipErrorCode.java b/src/java/android/net/sip/SipErrorCode.java
index 509728f..e3f54eb 100644
--- a/src/java/android/net/sip/SipErrorCode.java
+++ b/src/java/android/net/sip/SipErrorCode.java
@@ -22,6 +22,8 @@ package android.net.sip;
  * {@link SipSession.Listener#onError onError()},
  * {@link SipSession.Listener#onCallChangeFailed onCallChangeFailed()} and
  * {@link SipSession.Listener#onRegistrationFailed onRegistrationFailed()}.
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class SipErrorCode {
     /** Not an error. */
diff --git a/src/java/android/net/sip/SipException.java b/src/java/android/net/sip/SipException.java
index 0339395..b9cdb12 100644
--- a/src/java/android/net/sip/SipException.java
+++ b/src/java/android/net/sip/SipException.java
@@ -18,6 +18,8 @@ package android.net.sip;
 
 /**
  * Indicates a general SIP-related exception.
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class SipException extends Exception {
     public SipException() {
diff --git a/src/java/android/net/sip/SipManager.java b/src/java/android/net/sip/SipManager.java
index 39f66de..b74b07d 100644
--- a/src/java/android/net/sip/SipManager.java
+++ b/src/java/android/net/sip/SipManager.java
@@ -63,6 +63,8 @@ import java.util.List;
  * <a href="{@docRoot}guide/topics/network/sip.html">Session Initiation Protocol</a>
  * developer guide.</p>
  * </div>
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class SipManager {
     /**
diff --git a/src/java/android/net/sip/SipProfile.java b/src/java/android/net/sip/SipProfile.java
index 14b0334..6f22e47 100644
--- a/src/java/android/net/sip/SipProfile.java
+++ b/src/java/android/net/sip/SipProfile.java
@@ -45,6 +45,8 @@ import javax.sip.address.URI;
  * <a href="{@docRoot}guide/topics/network/sip.html">Session Initiation Protocol</a>
  * developer guide.</p>
  * </div>
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public class SipProfile implements Parcelable, Serializable, Cloneable {
     private static final long serialVersionUID = 1L;
diff --git a/src/java/android/net/sip/SipRegistrationListener.java b/src/java/android/net/sip/SipRegistrationListener.java
index 9968cc7..0175e8a 100644
--- a/src/java/android/net/sip/SipRegistrationListener.java
+++ b/src/java/android/net/sip/SipRegistrationListener.java
@@ -18,6 +18,8 @@ package android.net.sip;
 
 /**
  * Listener for SIP registration events.
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public interface SipRegistrationListener {
     /**
diff --git a/src/java/android/net/sip/SipSession.java b/src/java/android/net/sip/SipSession.java
index edbc66f..764871c 100644
--- a/src/java/android/net/sip/SipSession.java
+++ b/src/java/android/net/sip/SipSession.java
@@ -25,6 +25,8 @@ import android.telephony.Rlog;
  * <p>You can get a {@link SipSession} from {@link SipManager} with {@link
  * SipManager#createSipSession createSipSession()} (when initiating calls) or {@link
  * SipManager#getSessionFor getSessionFor()} (when receiving calls).</p>
+ * @deprecated {@link android.net.sip.SipManager} and associated classes are no longer supported and
+ * should not be used as the basis of future VOIP apps.
  */
 public final class SipSession {
     private static final String TAG = "SipSession";
diff --git a/src/jni/rtp/Android.bp b/src/jni/rtp/Android.bp
index b6c8588..325d6b9 100644
--- a/src/jni/rtp/Android.bp
+++ b/src/jni/rtp/Android.bp
@@ -12,6 +12,10 @@
 // See the License for the specific language governing permissions and
 // limitations under the License.
 
+package {
+    default_applicable_licenses: ["Android-Apache-2.0"],
+}
+
 cc_library_shared {
     name: "librtp_jni",
     srcs: [
@@ -32,8 +36,11 @@ cc_library_shared {
     ],
 
     shared_libs: [
+        "framework-permission-aidl-cpp",
+        "libandroid_runtime",
         "libaudioclient",
         "libaudiofoundation",
+        "libbinder",
         "libcutils",
         "liblog",
         "libnativehelper",
@@ -42,13 +49,10 @@ cc_library_shared {
     ],
     static_libs: [
         "libgsm",
+        "framework-permission-aidl-cpp",
         "libstagefright_amrnbdec",
         "libstagefright_amrnbenc",
     ],
-    include_dirs: [
-        "frameworks/av/media/libstagefright/codecs/amrnb/enc/src",
-        "frameworks/av/media/libstagefright/codecs/amrnb/dec/src",
-    ],
     cflags: [
         "-fvisibility=hidden",
         "-Wall",
diff --git a/src/jni/rtp/AudioGroup.cpp b/src/jni/rtp/AudioGroup.cpp
index 51b723d..e92e799 100644
--- a/src/jni/rtp/AudioGroup.cpp
+++ b/src/jni/rtp/AudioGroup.cpp
@@ -44,6 +44,8 @@
 #include <system/audio.h>
 
 #include <nativehelper/ScopedUtfChars.h>
+#include <android/content/AttributionSourceState.h>
+#include <android_os_Parcel.h>
 
 #include "jni.h"
 #include <nativehelper/JNIHelp.h>
@@ -57,6 +59,8 @@ namespace {
 
 using namespace android;
 
+using android::content::AttributionSourceState;
+
 int gRandom = -1;
 
 // We use a circular array to implement jitter buffer. The simplest way is doing
@@ -480,7 +484,7 @@ void AudioStream::decode(int tick)
 class AudioGroup
 {
 public:
-    explicit AudioGroup(const String16 &opPackageName);
+    explicit AudioGroup(const AttributionSourceState &attributionSource);
     ~AudioGroup();
     bool set(int sampleRate, int sampleCount);
 
@@ -505,7 +509,7 @@ private:
     int mEventQueue;
     volatile int mDtmfEvent;
 
-    String16 mOpPackageName;
+    const AttributionSourceState mAttributionSource;
 
     int mMode;
     int mSampleRate;
@@ -554,9 +558,9 @@ private:
     sp<DeviceThread> mDeviceThread;
 };
 
-AudioGroup::AudioGroup(const String16 &opPackageName)
+AudioGroup::AudioGroup(const AttributionSourceState &attributionSource)
+        : mAttributionSource(attributionSource)
 {
-    mOpPackageName = opPackageName;
     mMode = ON_HOLD;
     mChain = NULL;
     mEventQueue = -1;
@@ -818,7 +822,7 @@ bool AudioGroup::DeviceThread::threadLoop()
 
     // Initialize AudioTrack and AudioRecord.
     sp<AudioTrack> track = new AudioTrack();
-    sp<AudioRecord> record = new AudioRecord(mGroup->mOpPackageName);
+    sp<AudioRecord> record = new AudioRecord(mGroup->mAttributionSource);
     // Set caller name so it can be logged in destructor.
     // MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_RTP
     track->setCallerName("rtp");
@@ -851,14 +855,14 @@ bool AudioGroup::DeviceThread::threadLoop()
     sp<AudioEffect> aec;
     if (mode == ECHO_SUPPRESSION) {
         if (mGroup->platformHasAec()) {
-            aec = new AudioEffect(FX_IID_AEC,
-                                    mGroup->mOpPackageName,
-                                    NULL,
-                                    0,
-                                    0,
-                                    0,
-                                    record->getSessionId(),
-                                    AUDIO_IO_HANDLE_NONE); // record sessionId is sufficient.
+            aec = new AudioEffect(mGroup->mAttributionSource);
+            aec->set(FX_IID_AEC,
+                     NULL,
+                     0,
+                     0,
+                     0,
+                     record->getSessionId(),
+                     AUDIO_IO_HANDLE_NONE); // record sessionId is sufficient.
             status_t status = aec->initCheck();
             if (status == NO_ERROR || status == ALREADY_EXISTS) {
                 aec->setEnabled(true);
@@ -953,7 +957,7 @@ static jfieldID gMode;
 
 jlong add(JNIEnv *env, jobject thiz, jint mode,
     jint socket, jstring jRemoteAddress, jint remotePort,
-    jstring jCodecSpec, jint dtmfType, jstring opPackageNameStr)
+    jstring jCodecSpec, jint dtmfType, jobject jAttributionSource)
 {
     AudioCodec *codec = NULL;
     AudioStream *stream = NULL;
@@ -981,7 +985,9 @@ jlong add(JNIEnv *env, jobject thiz, jint mode,
         return 0;
     }
 
-    ScopedUtfChars opPackageName(env, opPackageNameStr);
+    Parcel* parcel = parcelForJavaObject(env, jAttributionSource);
+    AttributionSourceState attributionSource;
+    attributionSource.readFromParcel(parcel);
 
     // Create audio codec.
     int codecType = -1;
@@ -1012,7 +1018,7 @@ jlong add(JNIEnv *env, jobject thiz, jint mode,
     group = (AudioGroup *)env->GetLongField(thiz, gNative);
     if (!group) {
         int mode = env->GetIntField(thiz, gMode);
-        group = new AudioGroup(String16(opPackageName.c_str()));
+        group = new AudioGroup(attributionSource);
         if (!group->set(8000, 256) || !group->setMode(mode)) {
             jniThrowException(env, "java/lang/IllegalStateException",
                 "cannot initialize audio group");
@@ -1068,7 +1074,7 @@ void sendDtmf(JNIEnv *env, jobject thiz, jint event)
 }
 
 JNINativeMethod gMethods[] = {
-    {"nativeAdd", "(IILjava/lang/String;ILjava/lang/String;ILjava/lang/String;)J", (void *)add},
+    {"nativeAdd", "(IILjava/lang/String;ILjava/lang/String;ILandroid/os/Parcel;)J", (void *)add},
     {"nativeRemove", "(J)V", (void *)remove},
     {"nativeSetMode", "(I)V", (void *)setMode},
     {"nativeSendDtmf", "(I)V", (void *)sendDtmf},
@@ -1078,7 +1084,7 @@ JNINativeMethod gMethods[] = {
 
 int registerAudioGroup(JNIEnv *env)
 {
-    gRandom = open("/dev/urandom", O_RDONLY);
+    gRandom = open("/dev/urandom", O_RDONLY | O_CLOEXEC);
     if (gRandom == -1) {
         ALOGE("urandom: %s", strerror(errno));
         return -1;
@@ -1092,5 +1098,6 @@ int registerAudioGroup(JNIEnv *env)
         ALOGE("JNI registration failed");
         return -1;
     }
+
     return 0;
 }
diff --git a/src/jni/rtp/GsmCodec.cpp b/src/jni/rtp/GsmCodec.cpp
index 4a04bfa..ff1acec 100644
--- a/src/jni/rtp/GsmCodec.cpp
+++ b/src/jni/rtp/GsmCodec.cpp
@@ -16,9 +16,7 @@
 
 #include "AudioCodec.h"
 
-extern "C" {
 #include "gsm.h"
-}
 
 namespace {