Merge "Fix memory leaks in mediaanalytics" am: f7902c01f8
am: 37062b54f3
Change-Id: Ic6443bfb6a3b03691909a4b3f6678d2d09acd00a
diff --git a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
index 23a35e5..f164f28 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
@@ -62,10 +62,8 @@
secure, keyId, iv, mode, pattern, subSamples, source, offset, destination,
[&](Status_V1_2 hStatus, uint32_t hBytesWritten, hidl_string hDetailedError) {
status = toStatus_1_0(hStatus);
- if (status == Status::OK) {
- bytesWritten = hBytesWritten;
- detailedError = hDetailedError;
- }
+ bytesWritten = hBytesWritten;
+ detailedError = hDetailedError;
}
);
@@ -109,6 +107,10 @@
"destination decrypt buffer base not set");
return Void();
}
+ } else {
+ _hidl_cb(Status_V1_2::ERROR_DRM_CANNOT_HANDLE, 0,
+ "destination type not supported");
+ return Void();
}
sp<IMemory> sourceBase = mSharedBufferMap[source.bufferId];
@@ -126,38 +128,45 @@
(static_cast<void *>(sourceBase->getPointer()));
uint8_t* srcPtr = static_cast<uint8_t *>(base + source.offset + offset);
void* destPtr = NULL;
- if (destination.type == BufferType::SHARED_MEMORY) {
- const SharedBuffer& destBuffer = destination.nonsecureMemory;
- sp<IMemory> destBase = mSharedBufferMap[destBuffer.bufferId];
- if (destBase == nullptr) {
- _hidl_cb(Status_V1_2::ERROR_DRM_CANNOT_HANDLE, 0, "destination is a nullptr");
- return Void();
- }
-
- if (destBuffer.offset + destBuffer.size > destBase->getSize()) {
- _hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "invalid buffer size");
- return Void();
- }
- destPtr = static_cast<void *>(base + destination.nonsecureMemory.offset);
- } else if (destination.type == BufferType::NATIVE_HANDLE) {
- native_handle_t *handle = const_cast<native_handle_t *>(
- destination.secureMemory.getNativeHandle());
- destPtr = static_cast<void *>(handle);
+ // destination.type == BufferType::SHARED_MEMORY
+ const SharedBuffer& destBuffer = destination.nonsecureMemory;
+ sp<IMemory> destBase = mSharedBufferMap[destBuffer.bufferId];
+ if (destBase == nullptr) {
+ _hidl_cb(Status_V1_2::ERROR_DRM_CANNOT_HANDLE, 0, "destination is a nullptr");
+ return Void();
}
+ if (destBuffer.offset + destBuffer.size > destBase->getSize()) {
+ _hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "invalid buffer size");
+ return Void();
+ }
+ destPtr = static_cast<void *>(base + destination.nonsecureMemory.offset);
+
+
// Calculate the output buffer size and determine if any subsamples are
// encrypted.
size_t destSize = 0;
bool haveEncryptedSubsamples = false;
for (size_t i = 0; i < subSamples.size(); i++) {
const SubSample &subSample = subSamples[i];
- destSize += subSample.numBytesOfClearData;
- destSize += subSample.numBytesOfEncryptedData;
+ if (__builtin_add_overflow(destSize, subSample.numBytesOfClearData, &destSize)) {
+ _hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "subsample clear size overflow");
+ return Void();
+ }
+ if (__builtin_add_overflow(destSize, subSample.numBytesOfEncryptedData, &destSize)) {
+ _hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "subsample encrypted size overflow");
+ return Void();
+ }
if (subSample.numBytesOfEncryptedData > 0) {
haveEncryptedSubsamples = true;
}
}
+ if (destSize > destBuffer.size) {
+ _hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "subsample sum too large");
+ return Void();
+ }
+
if (mode == Mode::UNENCRYPTED) {
if (haveEncryptedSubsamples) {
_hidl_cb(Status_V1_2::ERROR_DRM_CANNOT_HANDLE, 0,
diff --git a/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp b/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp
index 6469735..a8a552c 100644
--- a/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp
+++ b/media/codec2/hidl/1.0/vts/functional/audio/VtsHalMediaC2V1_0TargetAudioDecTest.cpp
@@ -547,6 +547,10 @@
if (mCompName == raw) {
bitStreamInfo[0] = 8000;
bitStreamInfo[1] = 1;
+ } else if (mCompName == g711alaw || mCompName == g711mlaw) {
+ // g711 test data is all 1-channel and has no embedded config info.
+ bitStreamInfo[0] = 8000;
+ bitStreamInfo[1] = 1;
} else {
ASSERT_NO_FATAL_FAILURE(
getInputChannelInfo(mComponent, mCompName, bitStreamInfo));
diff --git a/media/codec2/hidl/1.0/vts/functional/common/README.md b/media/codec2/hidl/1.0/vts/functional/common/README.md
index 50e8356..f2f579c 100644
--- a/media/codec2/hidl/1.0/vts/functional/common/README.md
+++ b/media/codec2/hidl/1.0/vts/functional/common/README.md
@@ -1,31 +1,36 @@
-## Codec2 VTS Hal @ 1.0 tests ##
----
-#### master :
+# Codec2 VTS Hal @ 1.0 tests #
+
+## master :
Functionality of master is to enumerate all the Codec2 components available in C2 media service.
-usage: VtsHalMediaC2V1\_0TargetMasterTest -I default
+usage: `VtsHalMediaC2V1_0TargetMasterTest -I default`
-#### component :
+## component :
Functionality of component test is to validate common functionality across all the Codec2 components available in C2 media service. For a standard C2 component, these tests are expected to pass.
-usage: VtsHalMediaC2V1\_0TargetComponentTest -I software -C <comp name>
-example: VtsHalMediaC2V1\_0TargetComponentTest -I software -C c2.android.vorbis.decoder
+usage: `VtsHalMediaC2V1_0TargetComponentTest -I software -C <comp name>`
-#### audio :
-Functionality of audio test is to validate audio specific functionality Codec2 components. The resource files for this test are taken from media/codec2/hidl/1.0/vts/functional/res. The path to these files on the device is required to be given for bitstream tests.
+example: `VtsHalMediaC2V1_0TargetComponentTest -I software -C c2.android.vorbis.decoder`
-usage: VtsHalMediaC2V1\_0TargetAudioDecTest -I default -C <comp name> -P /sdcard/media/
-usage: VtsHalMediaC2V1\_0TargetAudioEncTest -I software -C <comp name> -P /sdcard/media/
+## audio :
+Functionality of audio test is to validate audio specific functionality of Codec2 components. The resource files for this test are taken from `frameworks/av/media/codec2/hidl/1.0/vts/functional/res`. The path to these files on the device can be specified with `-P`. (If the device path is omitted, `/data/local/tmp/media/` is the default value.)
-example: VtsHalMediaC2V1\_0TargetAudioDecTest -I software -C c2.android.flac.decoder -P /sdcard/media/
-example: VtsHalMediaC2V1\_0TargetAudioEncTest -I software -C c2.android.opus.encoder -P /sdcard/media/
+usage: `VtsHalMediaC2V1_0TargetAudioDecTest -I default -C <comp name> -P <path to resource files>`
-#### video :
-Functionality of video test is to validate video specific functionality Codec2 components. The resource files for this test are taken from media/codec2/hidl/1.0/vts/functional/res. The path to these files on the device is required to be given for bitstream tests.
+usage: `VtsHalMediaC2V1_0TargetAudioEncTest -I software -C <comp name> -P <path to resource files>`
-usage: VtsHalMediaC2V1\_0TargetVideoDecTest -I default -C <comp name> -P /sdcard/media/
-usage: VtsHalMediaC2V1\_0TargetVideoEncTest -I software -C <comp name> -P /sdcard/media/
+example: `VtsHalMediaC2V1_0TargetAudioDecTest -I software -C c2.android.flac.decoder -P /data/local/tmp/media/`
-example: VtsHalMediaC2V1\_0TargetVideoDecTest -I software -C c2.android.avc.decoder -P /sdcard/media/
-example: VtsHalMediaC2V1\_0TargetVideoEncTest -I software -C c2.android.vp9.encoder -P /sdcard/media/
+example: `VtsHalMediaC2V1_0TargetAudioEncTest -I software -C c2.android.opus.encoder -P /data/local/tmp/media/`
+
+## video :
+Functionality of video test is to validate video specific functionality of Codec2 components. The resource files for this test are taken from `frameworks/av/media/codec2/hidl/1.0/vts/functional/res`. The path to these files on the device can be specified with `-P`. (If the device path is omitted, `/data/local/tmp/media/` is the default value.)
+
+usage: `VtsHalMediaC2V1_0TargetVideoDecTest -I default -C <comp name> -P <path to resource files>`
+
+usage: `VtsHalMediaC2V1_0TargetVideoEncTest -I software -C <comp name> -P <path to resource files>`
+
+example: `VtsHalMediaC2V1_0TargetVideoDecTest -I software -C c2.android.avc.decoder -P /data/local/tmp/media/`
+
+example: `VtsHalMediaC2V1_0TargetVideoEncTest -I software -C c2.android.vp9.encoder -P /data/local/tmp/media/`
diff --git a/media/codec2/hidl/1.0/vts/functional/common/media_c2_hidl_test_common.h b/media/codec2/hidl/1.0/vts/functional/common/media_c2_hidl_test_common.h
index c577dac..db59e54 100644
--- a/media/codec2/hidl/1.0/vts/functional/common/media_c2_hidl_test_common.h
+++ b/media/codec2/hidl/1.0/vts/functional/common/media_c2_hidl_test_common.h
@@ -118,7 +118,7 @@
registerTestService<IComponentStore>();
}
- ComponentTestEnvironment() : res("/sdcard/media/") {}
+ ComponentTestEnvironment() : res("/data/local/tmp/media/") {}
void setComponent(const char* _component) { component = _component; }
diff --git a/media/libstagefright/xmlparser/api/current.txt b/media/libstagefright/xmlparser/api/current.txt
index 9d7c57d..16c8af8 100644
--- a/media/libstagefright/xmlparser/api/current.txt
+++ b/media/libstagefright/xmlparser/api/current.txt
@@ -138,7 +138,12 @@
public class Variant {
ctor public Variant();
+ method public java.util.List<media.codecs.Alias> getAlias_optional();
+ method public java.util.List<media.codecs.Quirk> getAttribute_optional();
+ method public java.util.List<media.codecs.Feature> getFeature_optional();
+ method public java.util.List<media.codecs.Limit> getLimit_optional();
method public String getName();
+ method public java.util.List<media.codecs.Quirk> getQuirk_optional();
method public void setName(String);
}
diff --git a/media/libstagefright/xmlparser/media_codecs.xsd b/media/libstagefright/xmlparser/media_codecs.xsd
index 63ec5d0..3b5681f 100644
--- a/media/libstagefright/xmlparser/media_codecs.xsd
+++ b/media/libstagefright/xmlparser/media_codecs.xsd
@@ -107,6 +107,13 @@
<xs:attribute name="value" type="xs:string"/>
</xs:complexType>
<xs:complexType name="Variant">
+ <xs:choice minOccurs="0" maxOccurs="unbounded">
+ <xs:element name="Quirk" type="Quirk"/>
+ <xs:element name="Attribute" type="Quirk"/>
+ <xs:element name="Alias" type="Alias"/>
+ <xs:element name="Limit" type="Limit"/>
+ <xs:element name="Feature" type="Feature"/>
+ </xs:choice>
<xs:attribute name="name" type="xs:string"/>
</xs:complexType>
<xs:complexType name="Setting">
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 9b0872e..4b295c9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1960,7 +1960,8 @@
&output.notificationFrameCount,
callingPid, clientUid, &output.flags,
input.clientInfo.clientTid,
- &lStatus, portId);
+ &lStatus, portId,
+ input.opPackageName);
LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
// lStatus == BAD_TYPE means FAST flag was rejected: request a new input from
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index da05dac..d5257bd 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -19,6 +19,39 @@
#error This header file should only be included from AudioFlinger.h
#endif
+// Checks and monitors OP_RECORD_AUDIO
+class OpRecordAudioMonitor : public RefBase {
+public:
+ ~OpRecordAudioMonitor() override;
+ bool hasOpRecordAudio() const;
+
+ static sp<OpRecordAudioMonitor> createIfNeeded(uid_t uid, const String16& opPackageName);
+
+private:
+ OpRecordAudioMonitor(uid_t uid, const String16& opPackageName);
+ void onFirstRef() override;
+
+ AppOpsManager mAppOpsManager;
+
+ class RecordAudioOpCallback : public BnAppOpsCallback {
+ public:
+ explicit RecordAudioOpCallback(const wp<OpRecordAudioMonitor>& monitor);
+ void opChanged(int32_t op, const String16& packageName) override;
+
+ private:
+ const wp<OpRecordAudioMonitor> mMonitor;
+ };
+
+ sp<RecordAudioOpCallback> mOpCallback;
+ // called by RecordAudioOpCallback when OP_RECORD_AUDIO is updated in AppOp callback
+ // and in onFirstRef()
+ void checkRecordAudio();
+
+ std::atomic_bool mHasOpRecordAudio;
+ const uid_t mUid;
+ const String16 mPackage;
+};
+
// record track
class RecordTrack : public TrackBase {
public:
@@ -36,6 +69,7 @@
uid_t uid,
audio_input_flags_t flags,
track_type type,
+ const String16& opPackageName,
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
virtual ~RecordTrack();
virtual status_t initCheck() const;
@@ -68,7 +102,7 @@
{ return (mFlags & AUDIO_INPUT_FLAG_DIRECT) != 0; }
void setSilenced(bool silenced) { if (!isPatchTrack()) mSilenced = silenced; }
- bool isSilenced() const { return mSilenced; }
+ bool isSilenced() const;
status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
@@ -111,6 +145,11 @@
audio_input_flags_t mFlags;
bool mSilenced;
+
+ // used to enforce OP_RECORD_AUDIO
+ uid_t mUid;
+ String16 mOpPackageName;
+ sp<OpRecordAudioMonitor> mOpRecordAudioMonitor;
};
// playback track, used by PatchPanel
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 73292d3..ac892a5 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7303,7 +7303,7 @@
// Sanitize before releasing if the track has no access to the source data
// An idle UID receives silence from non virtual devices until active
if (activeTrack->isSilenced()) {
- memset(activeTrack->mSink.raw, 0, framesOut * mFrameSize);
+ memset(activeTrack->mSink.raw, 0, framesOut * activeTrack->frameSize());
}
activeTrack->releaseBuffer(&activeTrack->mSink);
}
@@ -7464,7 +7464,8 @@
audio_input_flags_t *flags,
pid_t tid,
status_t *status,
- audio_port_handle_t portId)
+ audio_port_handle_t portId,
+ const String16& opPackageName)
{
size_t frameCount = *pFrameCount;
size_t notificationFrameCount = *pNotificationFrameCount;
@@ -7598,7 +7599,7 @@
track = new RecordTrack(this, client, attr, sampleRate,
format, channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid, uid,
- *flags, TrackBase::TYPE_DEFAULT, portId);
+ *flags, TrackBase::TYPE_DEFAULT, opPackageName, portId);
lStatus = track->initCheck();
if (lStatus != NO_ERROR) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index acb1370..0a473d5 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1553,7 +1553,8 @@
audio_input_flags_t *flags,
pid_t tid,
status_t *status /*non-NULL*/,
- audio_port_handle_t portId);
+ audio_port_handle_t portId,
+ const String16& opPackageName);
status_t start(RecordTrack* recordTrack,
AudioSystem::sync_event_t event,
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 7a3bb0d..52e7d59 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -215,6 +215,8 @@
uint32_t channelCount() const { return mChannelCount; }
+ size_t frameSize() const { return mFrameSize; }
+
audio_channel_mask_t channelMask() const { return mChannelMask; }
virtual uint32_t sampleRate() const { return mSampleRate; }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 16a8a84..e4402bd 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -444,7 +444,7 @@
return mHasOpPlayAudio.load();
}
-// Note this method is never called (and never to be) for audio server / root track
+// Note this method is never called (and never to be) for audio server / patch record track
// - not called from constructor due to check on UID,
// - not called from PlayAudioOpCallback because the callback is not installed in this case
void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage()
@@ -1885,6 +1885,105 @@
// ----------------------------------------------------------------------------
// Record
// ----------------------------------------------------------------------------
+
+
+// ----------------------------------------------------------------------------
+// AppOp for audio recording
+// -------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AF::OpRecordAudioMonitor"
+
+// static
+sp<AudioFlinger::RecordThread::OpRecordAudioMonitor>
+AudioFlinger::RecordThread::OpRecordAudioMonitor::createIfNeeded(
+ uid_t uid, const String16& opPackageName)
+{
+ if (isServiceUid(uid)) {
+ ALOGV("not silencing record for service uid:%d pack:%s",
+ uid, String8(opPackageName).string());
+ return nullptr;
+ }
+
+ if (opPackageName.size() == 0) {
+ Vector<String16> packages;
+ // no package name, happens with SL ES clients
+ // query package manager to find one
+ PermissionController permissionController;
+ permissionController.getPackagesForUid(uid, packages);
+ if (packages.isEmpty()) {
+ return nullptr;
+ } else {
+ ALOGV("using pack:%s for uid:%d", String8(packages[0]).string(), uid);
+ return new OpRecordAudioMonitor(uid, packages[0]);
+ }
+ }
+
+ return new OpRecordAudioMonitor(uid, opPackageName);
+}
+
+AudioFlinger::RecordThread::OpRecordAudioMonitor::OpRecordAudioMonitor(
+ uid_t uid, const String16& opPackageName)
+ : mHasOpRecordAudio(true), mUid(uid), mPackage(opPackageName)
+{
+}
+
+AudioFlinger::RecordThread::OpRecordAudioMonitor::~OpRecordAudioMonitor()
+{
+ if (mOpCallback != 0) {
+ mAppOpsManager.stopWatchingMode(mOpCallback);
+ }
+ mOpCallback.clear();
+}
+
+void AudioFlinger::RecordThread::OpRecordAudioMonitor::onFirstRef()
+{
+ checkRecordAudio();
+ mOpCallback = new RecordAudioOpCallback(this);
+ ALOGV("start watching OP_RECORD_AUDIO for pack:%s", String8(mPackage).string());
+ mAppOpsManager.startWatchingMode(AppOpsManager::OP_RECORD_AUDIO, mPackage, mOpCallback);
+}
+
+bool AudioFlinger::RecordThread::OpRecordAudioMonitor::hasOpRecordAudio() const {
+ return mHasOpRecordAudio.load();
+}
+
+// Called by RecordAudioOpCallback when OP_RECORD_AUDIO is updated in AppOp callback
+// and in onFirstRef()
+// Note this method is never called (and never to be) for audio server / root track
+// due to the UID in createIfNeeded(). As a result for those record track, it's:
+// - not called from constructor,
+// - not called from RecordAudioOpCallback because the callback is not installed in this case
+void AudioFlinger::RecordThread::OpRecordAudioMonitor::checkRecordAudio()
+{
+ const int32_t mode = mAppOpsManager.checkOp(AppOpsManager::OP_RECORD_AUDIO,
+ mUid, mPackage);
+ const bool hasIt = (mode == AppOpsManager::MODE_ALLOWED);
+ // verbose logging only log when appOp changed
+ ALOGI_IF(hasIt != mHasOpRecordAudio.load(),
+ "OP_RECORD_AUDIO missing, %ssilencing record uid%d pack:%s",
+ hasIt ? "un" : "", mUid, String8(mPackage).string());
+ mHasOpRecordAudio.store(hasIt);
+}
+
+AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::RecordAudioOpCallback(
+ const wp<OpRecordAudioMonitor>& monitor) : mMonitor(monitor)
+{ }
+
+void AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::opChanged(int32_t op,
+ const String16& packageName) {
+ UNUSED(packageName);
+ if (op != AppOpsManager::OP_RECORD_AUDIO) {
+ return;
+ }
+ sp<OpRecordAudioMonitor> monitor = mMonitor.promote();
+ if (monitor != NULL) {
+ monitor->checkRecordAudio();
+ }
+}
+
+
+
#undef LOG_TAG
#define LOG_TAG "AF::RecordHandle"
@@ -1956,6 +2055,7 @@
uid_t uid,
audio_input_flags_t flags,
track_type type,
+ const String16& opPackageName,
audio_port_handle_t portId)
: TrackBase(thread, client, attr, sampleRate, format,
channelMask, frameCount, buffer, bufferSize, sessionId,
@@ -1969,7 +2069,8 @@
mResamplerBufferProvider(NULL), // initialize in case of early constructor exit
mRecordBufferConverter(NULL),
mFlags(flags),
- mSilenced(false)
+ mSilenced(false),
+ mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(uid, opPackageName))
{
if (mCblk == NULL) {
return;
@@ -2220,6 +2321,14 @@
mServerLatencyMs.store(latencyMs);
}
+bool AudioFlinger::RecordThread::RecordTrack::isSilenced() const {
+ if (mSilenced) {
+ return true;
+ }
+ // The monitor is only created for record tracks that can be silenced.
+ return mOpRecordAudioMonitor ? !mOpRecordAudioMonitor->hasOpRecordAudio() : false;
+}
+
status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
std::vector<media::MicrophoneInfo>* activeMicrophones)
{
@@ -2270,7 +2379,7 @@
audio_attributes_t{} /* currently unused for patch track */,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER,
- flags, TYPE_PATCH),
+ flags, TYPE_PATCH, String16()),
PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true),
*recordThread, timeout)
{
