Snap for 8426163 from b3af60eab5ae65f1303e6dbc914f76da77907cb1 to mainline-tzdata2-release
Change-Id: I28062749893f459375aaad18c050ecb7b8001b82
diff --git a/Android.bp b/Android.bp
index 5295c5e..7f2d193 100644
--- a/Android.bp
+++ b/Android.bp
@@ -12,10 +12,6 @@
// 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 9cae573..85255c8 100644
--- a/src/java/android/net/rtp/AudioCodec.java
+++ b/src/java/android/net/rtp/AudioCodec.java
@@ -33,9 +33,6 @@
* </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 3a85bbc..9d96006 100644
--- a/src/java/android/net/rtp/AudioGroup.java
+++ b/src/java/android/net/rtp/AudioGroup.java
@@ -17,10 +17,9 @@
package android.net.rtp;
import android.annotation.NonNull;
-import android.content.AttributionSource;
+import android.app.ActivityThread;
import android.content.Context;
import android.media.AudioManager;
-import android.os.Parcel;
import java.util.HashMap;
import java.util.Locale;
@@ -64,9 +63,6 @@
* 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 {
/**
@@ -118,7 +114,7 @@
/**
* Creates an empty AudioGroup.
- * @param context Context the group belongs to
+ * @param context Context used to get package name
*/
public AudioGroup(@NonNull Context context) {
mContext = context;
@@ -168,15 +164,11 @@
AudioCodec codec = stream.getCodec();
String codecSpec = String.format(Locale.US, "%d %s %s", codec.type,
codec.rtpmap, codec.fmtp);
-
- 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());
- }
+ long id = nativeAdd(stream.getMode(), stream.getSocket(),
+ stream.getRemoteAddress().getHostAddress(),
+ stream.getRemotePort(), codecSpec, stream.getDtmfType(),
+ mContext != null ? mContext.getOpPackageName()
+ : ActivityThread.currentOpPackageName());
mStreams.put(stream, id);
} catch (NullPointerException e) {
throw new IllegalStateException(e);
@@ -184,8 +176,8 @@
}
}
- private native long nativeAdd(int mode, int socket, String remoteAddress, int remotePort,
- String codecSpec, int dtmfType, Parcel attributionSource);
+ private native long nativeAdd(int mode, int socket, String remoteAddress,
+ int remotePort, String codecSpec, int dtmfType, String opPackageName);
// 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 e746e7d..5cd1abc 100644
--- a/src/java/android/net/rtp/AudioStream.java
+++ b/src/java/android/net/rtp/AudioStream.java
@@ -41,8 +41,6 @@
*
* @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 ad37455..b9d75cd 100644
--- a/src/java/android/net/rtp/RtpStream.java
+++ b/src/java/android/net/rtp/RtpStream.java
@@ -27,8 +27,6 @@
*
* <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 4d70017..363995c 100644
--- a/src/java/android/net/sip/SipAudioCall.java
+++ b/src/java/android/net/sip/SipAudioCall.java
@@ -56,8 +56,6 @@
* <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 e3f54eb..509728f 100644
--- a/src/java/android/net/sip/SipErrorCode.java
+++ b/src/java/android/net/sip/SipErrorCode.java
@@ -22,8 +22,6 @@
* {@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 b9cdb12..0339395 100644
--- a/src/java/android/net/sip/SipException.java
+++ b/src/java/android/net/sip/SipException.java
@@ -18,8 +18,6 @@
/**
* 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 b74b07d..39f66de 100644
--- a/src/java/android/net/sip/SipManager.java
+++ b/src/java/android/net/sip/SipManager.java
@@ -63,8 +63,6 @@
* <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 6f22e47..14b0334 100644
--- a/src/java/android/net/sip/SipProfile.java
+++ b/src/java/android/net/sip/SipProfile.java
@@ -45,8 +45,6 @@
* <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 0175e8a..9968cc7 100644
--- a/src/java/android/net/sip/SipRegistrationListener.java
+++ b/src/java/android/net/sip/SipRegistrationListener.java
@@ -18,8 +18,6 @@
/**
* 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 764871c..edbc66f 100644
--- a/src/java/android/net/sip/SipSession.java
+++ b/src/java/android/net/sip/SipSession.java
@@ -25,8 +25,6 @@
* <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 325d6b9..b6c8588 100644
--- a/src/jni/rtp/Android.bp
+++ b/src/jni/rtp/Android.bp
@@ -12,10 +12,6 @@
// 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: [
@@ -36,11 +32,8 @@
],
shared_libs: [
- "framework-permission-aidl-cpp",
- "libandroid_runtime",
"libaudioclient",
"libaudiofoundation",
- "libbinder",
"libcutils",
"liblog",
"libnativehelper",
@@ -49,10 +42,13 @@
],
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 e92e799..51b723d 100644
--- a/src/jni/rtp/AudioGroup.cpp
+++ b/src/jni/rtp/AudioGroup.cpp
@@ -44,8 +44,6 @@
#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>
@@ -59,8 +57,6 @@
using namespace android;
-using android::content::AttributionSourceState;
-
int gRandom = -1;
// We use a circular array to implement jitter buffer. The simplest way is doing
@@ -484,7 +480,7 @@
class AudioGroup
{
public:
- explicit AudioGroup(const AttributionSourceState &attributionSource);
+ explicit AudioGroup(const String16 &opPackageName);
~AudioGroup();
bool set(int sampleRate, int sampleCount);
@@ -509,7 +505,7 @@
int mEventQueue;
volatile int mDtmfEvent;
- const AttributionSourceState mAttributionSource;
+ String16 mOpPackageName;
int mMode;
int mSampleRate;
@@ -558,9 +554,9 @@
sp<DeviceThread> mDeviceThread;
};
-AudioGroup::AudioGroup(const AttributionSourceState &attributionSource)
- : mAttributionSource(attributionSource)
+AudioGroup::AudioGroup(const String16 &opPackageName)
{
+ mOpPackageName = opPackageName;
mMode = ON_HOLD;
mChain = NULL;
mEventQueue = -1;
@@ -822,7 +818,7 @@
// Initialize AudioTrack and AudioRecord.
sp<AudioTrack> track = new AudioTrack();
- sp<AudioRecord> record = new AudioRecord(mGroup->mAttributionSource);
+ 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");
@@ -855,14 +851,14 @@
sp<AudioEffect> aec;
if (mode == ECHO_SUPPRESSION) {
if (mGroup->platformHasAec()) {
- aec = new AudioEffect(mGroup->mAttributionSource);
- aec->set(FX_IID_AEC,
- NULL,
- 0,
- 0,
- 0,
- record->getSessionId(),
- AUDIO_IO_HANDLE_NONE); // record sessionId is sufficient.
+ aec = new AudioEffect(FX_IID_AEC,
+ mGroup->mOpPackageName,
+ 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);
@@ -957,7 +953,7 @@
jlong add(JNIEnv *env, jobject thiz, jint mode,
jint socket, jstring jRemoteAddress, jint remotePort,
- jstring jCodecSpec, jint dtmfType, jobject jAttributionSource)
+ jstring jCodecSpec, jint dtmfType, jstring opPackageNameStr)
{
AudioCodec *codec = NULL;
AudioStream *stream = NULL;
@@ -985,9 +981,7 @@
return 0;
}
- Parcel* parcel = parcelForJavaObject(env, jAttributionSource);
- AttributionSourceState attributionSource;
- attributionSource.readFromParcel(parcel);
+ ScopedUtfChars opPackageName(env, opPackageNameStr);
// Create audio codec.
int codecType = -1;
@@ -1018,7 +1012,7 @@
group = (AudioGroup *)env->GetLongField(thiz, gNative);
if (!group) {
int mode = env->GetIntField(thiz, gMode);
- group = new AudioGroup(attributionSource);
+ group = new AudioGroup(String16(opPackageName.c_str()));
if (!group->set(8000, 256) || !group->setMode(mode)) {
jniThrowException(env, "java/lang/IllegalStateException",
"cannot initialize audio group");
@@ -1074,7 +1068,7 @@
}
JNINativeMethod gMethods[] = {
- {"nativeAdd", "(IILjava/lang/String;ILjava/lang/String;ILandroid/os/Parcel;)J", (void *)add},
+ {"nativeAdd", "(IILjava/lang/String;ILjava/lang/String;ILjava/lang/String;)J", (void *)add},
{"nativeRemove", "(J)V", (void *)remove},
{"nativeSetMode", "(I)V", (void *)setMode},
{"nativeSendDtmf", "(I)V", (void *)sendDtmf},
@@ -1084,7 +1078,7 @@
int registerAudioGroup(JNIEnv *env)
{
- gRandom = open("/dev/urandom", O_RDONLY | O_CLOEXEC);
+ gRandom = open("/dev/urandom", O_RDONLY);
if (gRandom == -1) {
ALOGE("urandom: %s", strerror(errno));
return -1;
@@ -1098,6 +1092,5 @@
ALOGE("JNI registration failed");
return -1;
}
-
return 0;
}
diff --git a/src/jni/rtp/GsmCodec.cpp b/src/jni/rtp/GsmCodec.cpp
index ff1acec..4a04bfa 100644
--- a/src/jni/rtp/GsmCodec.cpp
+++ b/src/jni/rtp/GsmCodec.cpp
@@ -16,7 +16,9 @@
#include "AudioCodec.h"
+extern "C" {
#include "gsm.h"
+}
namespace {
