Merge "Audioflinger: tracks monitor OP_PLAY_AUDIO"
diff --git a/Android.bp b/Android.bp
deleted file mode 100644
index e4f12c8..0000000
--- a/Android.bp
+++ /dev/null
@@ -1,7 +0,0 @@
-subdirs = [
- "camera",
- "drm/*",
- "media/*",
- "services/*",
- "soundtrigger",
-]
diff --git a/apex/ld.config.txt b/apex/ld.config.txt
index b342206..2daeeac 100644
--- a/apex/ld.config.txt
+++ b/apex/ld.config.txt
@@ -48,7 +48,9 @@
# TODO: replace the following when apex has a way to auto-generate this list
# namespace.default.link.platform.shared_libs = %LLNDK_LIBRARIES%
# namespace.default.link.platform.shared_libs += %SANITIZER_RUNTIME_LIBRARIES%
-namespace.default.link.platform.shared_libs = libEGL.so:libGLESv1_CM.so:libGLESv2.so:libGLESv3.so:libRS.so:libandroid_net.so:libc.so:libclang_rt.asan-aarch64-android.so:libclang_rt.asan-arm-android.so:libclang_rt.hwasan-aarch64-android.so:libclang_rt.asan-i686-android.so:libclang_rt.asan-x86_64-android.so:libdl.so:libft2.so:liblog.so:libm.so:libmediandk.so:libnativewindow.so:libneuralnetworks.so:libsync.so:libvndksupport.so:libvulkan.so
+namespace.default.link.platform.shared_libs = libEGL.so:libGLESv1_CM.so:libGLESv2.so:libGLESv3.so:libRS.so:libandroid_net.so:libc.so:libcgrouprc.so:libclang_rt.asan-aarch64-android.so:libclang_rt.asan-arm-android.so:libclang_rt.hwasan-aarch64-android.so:libclang_rt.asan-i686-android.so:libclang_rt.asan-x86_64-android.so:libdl.so:libft2.so:liblog.so:libm.so:libmediandk.so:libnativewindow.so:libneuralnetworks.so:libsync.so:libvndksupport.so:libvulkan.so
+# FIXME: b/129552044
+namespace.default.link.platform.shared_libs += libz.so
###############################################################################
# "platform" namespace
diff --git a/apex/manifest.json b/apex/manifest.json
index e2df3a3..c6c63f6 100644
--- a/apex/manifest.json
+++ b/apex/manifest.json
@@ -1,4 +1,4 @@
{
"name": "com.android.media",
- "version": 1
+ "version": 200000000
}
diff --git a/apex/manifest_codec.json b/apex/manifest_codec.json
index e2bceec..4f31b15 100644
--- a/apex/manifest_codec.json
+++ b/apex/manifest_codec.json
@@ -1,4 +1,4 @@
{
"name": "com.android.media.swcodec",
- "version": 1
+ "version": 200000000
}
diff --git a/camera/Android.bp b/camera/Android.bp
index 21588d4..2800595 100644
--- a/camera/Android.bp
+++ b/camera/Android.bp
@@ -12,8 +12,6 @@
// See the License for the specific language governing permissions and
// limitations under the License.
-subdirs = ["ndk"]
-
cc_library_shared {
name: "libcamera_client",
diff --git a/camera/CaptureResult.cpp b/camera/CaptureResult.cpp
index 928a6bc..1d8e8c4 100644
--- a/camera/CaptureResult.cpp
+++ b/camera/CaptureResult.cpp
@@ -39,6 +39,16 @@
parcel->readInt64(&frameNumber);
parcel->readInt32(&partialResultCount);
parcel->readInt32(&errorStreamId);
+ auto physicalCameraIdPresent = parcel->readBool();
+ if (physicalCameraIdPresent) {
+ String16 cameraId;
+ status_t res = OK;
+ if ((res = parcel->readString16(&cameraId)) != OK) {
+ ALOGE("%s: Failed to read camera id: %d", __FUNCTION__, res);
+ return res;
+ }
+ errorPhysicalCameraId = cameraId;
+ }
return OK;
}
@@ -56,6 +66,16 @@
parcel->writeInt64(frameNumber);
parcel->writeInt32(partialResultCount);
parcel->writeInt32(errorStreamId);
+ if (errorPhysicalCameraId.size() > 0) {
+ parcel->writeBool(true);
+ status_t res = OK;
+ if ((res = parcel->writeString16(errorPhysicalCameraId)) != OK) {
+ ALOGE("%s: Failed to write physical camera ID to parcel: %d", __FUNCTION__, res);
+ return res;
+ }
+ } else {
+ parcel->writeBool(false);
+ }
return OK;
}
diff --git a/camera/include/camera/CaptureResult.h b/camera/include/camera/CaptureResult.h
index 56fa178..ef830b5 100644
--- a/camera/include/camera/CaptureResult.h
+++ b/camera/include/camera/CaptureResult.h
@@ -70,6 +70,13 @@
int32_t errorStreamId;
/**
+ * For capture result errors, the physical camera ID in case the respective request contains
+ * a reference to physical camera device.
+ * Empty otherwise.
+ */
+ String16 errorPhysicalCameraId;
+
+ /**
* Constructor initializes object as invalid by setting requestId to be -1.
*/
CaptureResultExtras()
@@ -79,7 +86,8 @@
precaptureTriggerId(0),
frameNumber(0),
partialResultCount(0),
- errorStreamId(-1) {
+ errorStreamId(-1),
+ errorPhysicalCameraId() {
}
/**
diff --git a/camera/ndk/impl/ACameraDevice.cpp b/camera/ndk/impl/ACameraDevice.cpp
index c9db01e..25a81eb 100644
--- a/camera/ndk/impl/ACameraDevice.cpp
+++ b/camera/ndk/impl/ACameraDevice.cpp
@@ -45,6 +45,7 @@
const char* CameraDevice::kSequenceIdKey = "SequenceId";
const char* CameraDevice::kFrameNumberKey = "FrameNumber";
const char* CameraDevice::kAnwKey = "Anw";
+const char* CameraDevice::kFailingPhysicalCameraId= "FailingPhysicalCameraId";
/**
* CameraDevice Implementation
@@ -867,10 +868,19 @@
failure->wasImageCaptured = (errorCode ==
hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_RESULT);
- sp<AMessage> msg = new AMessage(kWhatCaptureFail, mHandler);
+ sp<AMessage> msg = new AMessage(cbh.mIsLogicalCameraCallback ? kWhatLogicalCaptureFail :
+ kWhatCaptureFail, mHandler);
msg->setPointer(kContextKey, cbh.mContext);
msg->setObject(kSessionSpKey, session);
- msg->setPointer(kCallbackFpKey, (void*) onError);
+ if (cbh.mIsLogicalCameraCallback) {
+ if (resultExtras.errorPhysicalCameraId.size() > 0) {
+ String8 cameraId(resultExtras.errorPhysicalCameraId);
+ msg->setString(kFailingPhysicalCameraId, cameraId.string(), cameraId.size());
+ }
+ msg->setPointer(kCallbackFpKey, (void*) cbh.mOnLogicalCameraCaptureFailed);
+ } else {
+ msg->setPointer(kCallbackFpKey, (void*) onError);
+ }
msg->setObject(kCaptureRequestKey, request);
msg->setObject(kCaptureFailureKey, failure);
postSessionMsgAndCleanup(msg);
@@ -895,6 +905,7 @@
case kWhatCaptureResult:
case kWhatLogicalCaptureResult:
case kWhatCaptureFail:
+ case kWhatLogicalCaptureFail:
case kWhatCaptureSeqEnd:
case kWhatCaptureSeqAbort:
case kWhatCaptureBufferLost:
@@ -966,6 +977,7 @@
case kWhatCaptureResult:
case kWhatLogicalCaptureResult:
case kWhatCaptureFail:
+ case kWhatLogicalCaptureFail:
case kWhatCaptureSeqEnd:
case kWhatCaptureSeqAbort:
case kWhatCaptureBufferLost:
@@ -984,6 +996,7 @@
case kWhatCaptureResult:
case kWhatLogicalCaptureResult:
case kWhatCaptureFail:
+ case kWhatLogicalCaptureFail:
case kWhatCaptureBufferLost:
found = msg->findObject(kCaptureRequestKey, &obj);
if (!found) {
@@ -1138,6 +1151,39 @@
freeACaptureRequest(request);
break;
}
+ case kWhatLogicalCaptureFail:
+ {
+ ACameraCaptureSession_logicalCamera_captureCallback_failed onFail;
+ found = msg->findPointer(kCallbackFpKey, (void**) &onFail);
+ if (!found) {
+ ALOGE("%s: Cannot find capture fail callback!", __FUNCTION__);
+ return;
+ }
+ if (onFail == nullptr) {
+ return;
+ }
+
+ found = msg->findObject(kCaptureFailureKey, &obj);
+ if (!found) {
+ ALOGE("%s: Cannot find capture failure!", __FUNCTION__);
+ return;
+ }
+ sp<CameraCaptureFailure> failureSp(
+ static_cast<CameraCaptureFailure*>(obj.get()));
+ ALogicalCameraCaptureFailure failure;
+ AString physicalCameraId;
+ found = msg->findString(kFailingPhysicalCameraId, &physicalCameraId);
+ if (found && !physicalCameraId.empty()) {
+ failure.physicalCameraId = physicalCameraId.c_str();
+ } else {
+ failure.physicalCameraId = nullptr;
+ }
+ failure.captureFailure = *failureSp;
+ ACaptureRequest* request = allocateACaptureRequest(requestSp, mId);
+ (*onFail)(context, session.get(), request, &failure);
+ freeACaptureRequest(request);
+ break;
+ }
case kWhatCaptureSeqEnd:
{
ACameraCaptureSession_captureCallback_sequenceEnd onSeqEnd;
@@ -1233,6 +1279,7 @@
if (cbs != nullptr) {
mOnCaptureCompleted = cbs->onCaptureCompleted;
+ mOnCaptureFailed = cbs->onCaptureFailed;
}
}
@@ -1248,6 +1295,7 @@
if (lcbs != nullptr) {
mOnLogicalCameraCaptureCompleted = lcbs->onLogicalCameraCaptureCompleted;
+ mOnLogicalCameraCaptureFailed = lcbs->onLogicalCameraCaptureFailed;
}
}
diff --git a/camera/ndk/impl/ACameraDevice.h b/camera/ndk/impl/ACameraDevice.h
index 56741ce..c92a95f 100644
--- a/camera/ndk/impl/ACameraDevice.h
+++ b/camera/ndk/impl/ACameraDevice.h
@@ -214,6 +214,7 @@
kWhatCaptureResult, // onCaptureProgressed, onCaptureCompleted
kWhatLogicalCaptureResult, // onLogicalCameraCaptureCompleted
kWhatCaptureFail, // onCaptureFailed
+ kWhatLogicalCaptureFail, // onLogicalCameraCaptureFailed
kWhatCaptureSeqEnd, // onCaptureSequenceCompleted
kWhatCaptureSeqAbort, // onCaptureSequenceAborted
kWhatCaptureBufferLost,// onCaptureBufferLost
@@ -233,6 +234,7 @@
static const char* kSequenceIdKey;
static const char* kFrameNumberKey;
static const char* kAnwKey;
+ static const char* kFailingPhysicalCameraId;
class CallbackHandler : public AHandler {
public:
@@ -281,6 +283,7 @@
mOnCaptureProgressed = nullptr;
mOnCaptureCompleted = nullptr;
mOnLogicalCameraCaptureCompleted = nullptr;
+ mOnLogicalCameraCaptureFailed = nullptr;
mOnCaptureFailed = nullptr;
mOnCaptureSequenceCompleted = nullptr;
mOnCaptureSequenceAborted = nullptr;
@@ -289,7 +292,6 @@
mContext = cbs->context;
mOnCaptureStarted = cbs->onCaptureStarted;
mOnCaptureProgressed = cbs->onCaptureProgressed;
- mOnCaptureFailed = cbs->onCaptureFailed;
mOnCaptureSequenceCompleted = cbs->onCaptureSequenceCompleted;
mOnCaptureSequenceAborted = cbs->onCaptureSequenceAborted;
mOnCaptureBufferLost = cbs->onCaptureBufferLost;
@@ -305,6 +307,7 @@
ACameraCaptureSession_captureCallback_result mOnCaptureProgressed;
ACameraCaptureSession_captureCallback_result mOnCaptureCompleted;
ACameraCaptureSession_logicalCamera_captureCallback_result mOnLogicalCameraCaptureCompleted;
+ ACameraCaptureSession_logicalCamera_captureCallback_failed mOnLogicalCameraCaptureFailed;
ACameraCaptureSession_captureCallback_failed mOnCaptureFailed;
ACameraCaptureSession_captureCallback_sequenceEnd mOnCaptureSequenceCompleted;
ACameraCaptureSession_captureCallback_sequenceAbort mOnCaptureSequenceAborted;
diff --git a/camera/ndk/include/camera/NdkCameraCaptureSession.h b/camera/ndk/include/camera/NdkCameraCaptureSession.h
index d13a818..07176cf 100644
--- a/camera/ndk/include/camera/NdkCameraCaptureSession.h
+++ b/camera/ndk/include/camera/NdkCameraCaptureSession.h
@@ -676,6 +676,41 @@
size_t physicalResultCount, const char** physicalCameraIds,
const ACameraMetadata** physicalResults);
+/// Struct to describe a logical camera capture failure
+typedef struct ALogicalCameraCaptureFailure {
+ /**
+ * The {@link ACameraCaptureFailure} contains information about regular logical device capture
+ * failure.
+ */
+ struct ACameraCaptureFailure captureFailure;
+
+ /**
+ * The physical camera device ID in case the capture failure comes from a capture request
+ * with configured physical camera streams for a logical camera. physicalCameraId will be set
+ * to NULL in case the capture request has no associated physical camera device.
+ *
+ */
+ const char* physicalCameraId;
+} ALogicalCameraCaptureFailure;
+
+/**
+ * The definition of logical camera capture failure callback.
+ *
+ * @param context The optional application context provided by user in
+ * {@link ACameraCaptureSession_captureCallbacks}.
+ * @param session The camera capture session of interest.
+ * @param request The capture request of interest. Note that this pointer points to a copy of
+ * capture request sent by application, so the address is different to what
+ * application sent but the content will match. This request will be freed by
+ * framework immediately after this callback returns.
+ * @param failure The {@link ALogicalCameraCaptureFailure} desribes the capture failure. The memory
+ * is managed by camera framework. Do not access this pointer after this callback
+ * returns.
+ */
+typedef void (*ACameraCaptureSession_logicalCamera_captureCallback_failed)(
+ void* context, ACameraCaptureSession* session,
+ ACaptureRequest* request, ALogicalCameraCaptureFailure* failure);
+
/**
* This has the same functionality as ACameraCaptureSession_captureCallbacks,
* with the exception that an onLogicalCameraCaptureCompleted callback is
@@ -708,9 +743,24 @@
ACameraCaptureSession_logicalCamera_captureCallback_result onLogicalCameraCaptureCompleted;
/**
+ * This callback is called instead of {@link onLogicalCameraCaptureCompleted} when the
+ * camera device failed to produce a capture result for the
+ * request.
+ *
+ * <p>Other requests are unaffected, and some or all image buffers from
+ * the capture may have been pushed to their respective output
+ * streams.</p>
+ *
+ * <p>Note that the ACaptureRequest pointer in the callback will not match what application has
+ * submitted, but the contents the ACaptureRequest will match what application submitted.</p>
+ *
+ * @see ALogicalCameraCaptureFailure
+ */
+ ACameraCaptureSession_logicalCamera_captureCallback_failed onLogicalCameraCaptureFailed;
+
+ /**
* Same as ACameraCaptureSession_captureCallbacks
*/
- ACameraCaptureSession_captureCallback_failed onCaptureFailed;
ACameraCaptureSession_captureCallback_sequenceEnd onCaptureSequenceCompleted;
ACameraCaptureSession_captureCallback_sequenceAbort onCaptureSequenceAborted;
ACameraCaptureSession_captureCallback_bufferLost onCaptureBufferLost;
diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index acf6999..99b613e 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -5695,6 +5695,8 @@
* <p>In both cases, all images generated for a particular capture request still carry the same
* timestamps, so that they can be used to look up the matching frame number and
* onCaptureStarted callback.</p>
+ * <p>This tag is only applicable if the logical camera device supports concurrent physical
+ * streams from different physical cameras.</p>
*/
ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE = // byte (acamera_metadata_enum_android_logical_multi_camera_sensor_sync_type_t)
ACAMERA_LOGICAL_MULTI_CAMERA_START + 1,
@@ -7581,14 +7583,23 @@
ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_MOTION_TRACKING = 10,
/**
- * <p>The camera device is a logical camera backed by two or more physical cameras. In
- * API level 28, the physical cameras must also be exposed to the application via
- * <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#getCameraIdList">CameraManager#getCameraIdList</a>. Starting from API
- * level 29, some or all physical cameras may not be independently exposed to the
- * application, in which case the physical camera IDs will not be available in
- * <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#getCameraIdList">CameraManager#getCameraIdList</a>. But the application
- * can still query the physical cameras' characteristics by calling
- * <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#getCameraCharacteristics">CameraManager#getCameraCharacteristics</a>.</p>
+ * <p>The camera device is a logical camera backed by two or more physical cameras.</p>
+ * <p>In API level 28, the physical cameras must also be exposed to the application via
+ * <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#getCameraIdList">CameraManager#getCameraIdList</a>.</p>
+ * <p>Starting from API level 29, some or all physical cameras may not be independently
+ * exposed to the application, in which case the physical camera IDs will not be
+ * available in <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#getCameraIdList">CameraManager#getCameraIdList</a>. But the
+ * application can still query the physical cameras' characteristics by calling
+ * <a href="https://developer.android.com/reference/android/hardware/camera2/CameraManager.html#getCameraCharacteristics">CameraManager#getCameraCharacteristics</a>. Additionally,
+ * if a physical camera is hidden from camera ID list, the mandatory stream combinations
+ * for that physical camera must be supported through the logical camera using physical
+ * streams.</p>
+ * <p>Combinations of logical and physical streams, or physical streams from different
+ * physical cameras are not guaranteed. However, if the camera device supports
+ * {@link ACameraDevice_isSessionConfigurationSupported },
+ * application must be able to query whether a stream combination involving physical
+ * streams is supported by calling
+ * {@link ACameraDevice_isSessionConfigurationSupported }.</p>
* <p>Camera application shouldn't assume that there are at most 1 rear camera and 1 front
* camera in the system. For an application that switches between front and back cameras,
* the recommendation is to switch between the first rear camera and the first front
@@ -7613,24 +7624,6 @@
* the same.</li>
* <li>The logical camera must be LIMITED or higher device.</li>
* </ul>
- * <p>Both the logical camera device and its underlying physical devices support the
- * mandatory stream combinations required for their device levels.</p>
- * <p>Additionally, for each guaranteed stream combination, the logical camera supports:</p>
- * <ul>
- * <li>For each guaranteed stream combination, the logical camera supports replacing one
- * logical {@link AIMAGE_FORMAT_YUV_420_888 YUV_420_888}
- * or raw stream with two physical streams of the same size and format, each from a
- * separate physical camera, given that the size and format are supported by both
- * physical cameras.</li>
- * <li>If the logical camera doesn't advertise RAW capability, but the underlying physical
- * cameras do, the logical camera will support guaranteed stream combinations for RAW
- * capability, except that the RAW streams will be physical streams, each from a separate
- * physical camera. This is usually the case when the physical cameras have different
- * sensor sizes.</li>
- * </ul>
- * <p>Using physical streams in place of a logical stream of the same size and format will
- * not slow down the frame rate of the capture, as long as the minimum frame duration
- * of the physical and logical streams are the same.</p>
* <p>A logical camera device's dynamic metadata may contain
* ACAMERA_LOGICAL_MULTI_CAMERA_ACTIVE_PHYSICAL_ID to notify the application of the current
* active physical camera Id. An active physical camera is the physical camera from which
diff --git a/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp b/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
index d7d774b..b18c897 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
+++ b/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
@@ -55,6 +55,7 @@
const char* CameraDevice::kSequenceIdKey = "SequenceId";
const char* CameraDevice::kFrameNumberKey = "FrameNumber";
const char* CameraDevice::kAnwKey = "Anw";
+const char* CameraDevice::kFailingPhysicalCameraId= "FailingPhysicalCameraId";
/**
* CameraDevice Implementation
@@ -299,7 +300,6 @@
OutputConfigurationWrapper outConfigW;
OutputConfiguration &outConfig = outConfigW.mOutputConfiguration;
outConfig.rotation = utils::convertToHidl(output->mRotation);
- outConfig.windowGroupId = -1; // ndk doesn't support inter OutputConfiguration buffer sharing.
outConfig.windowHandles.resize(output->mSharedWindows.size() + 1);
outConfig.windowHandles[0] = output->mWindow;
outConfig.physicalCameraId = output->mPhysicalCameraId;
@@ -894,10 +894,19 @@
failure->sequenceId = sequenceId;
failure->wasImageCaptured = (errorCode == ErrorCode::CAMERA_RESULT);
- sp<AMessage> msg = new AMessage(kWhatCaptureFail, mHandler);
+ sp<AMessage> msg = new AMessage(cbh.mIsLogicalCameraCallback ? kWhatLogicalCaptureFail :
+ kWhatCaptureFail, mHandler);
msg->setPointer(kContextKey, cbh.mContext);
msg->setObject(kSessionSpKey, session);
- msg->setPointer(kCallbackFpKey, (void*) onError);
+ if (cbh.mIsLogicalCameraCallback) {
+ if (resultExtras.errorPhysicalCameraId.size() > 0) {
+ msg->setString(kFailingPhysicalCameraId, resultExtras.errorPhysicalCameraId.c_str(),
+ resultExtras.errorPhysicalCameraId.size());
+ }
+ msg->setPointer(kCallbackFpKey, (void*) cbh.mOnLogicalCameraCaptureFailed);
+ } else {
+ msg->setPointer(kCallbackFpKey, (void*) onError);
+ }
msg->setObject(kCaptureRequestKey, request);
msg->setObject(kCaptureFailureKey, failure);
postSessionMsgAndCleanup(msg);
@@ -919,6 +928,7 @@
case kWhatCaptureResult:
case kWhatLogicalCaptureResult:
case kWhatCaptureFail:
+ case kWhatLogicalCaptureFail:
case kWhatCaptureSeqEnd:
case kWhatCaptureSeqAbort:
case kWhatCaptureBufferLost:
@@ -990,6 +1000,7 @@
case kWhatCaptureResult:
case kWhatLogicalCaptureResult:
case kWhatCaptureFail:
+ case kWhatLogicalCaptureFail:
case kWhatCaptureSeqEnd:
case kWhatCaptureSeqAbort:
case kWhatCaptureBufferLost:
@@ -1009,6 +1020,7 @@
case kWhatCaptureResult:
case kWhatLogicalCaptureResult:
case kWhatCaptureFail:
+ case kWhatLogicalCaptureFail:
case kWhatCaptureBufferLost:
found = msg->findObject(kCaptureRequestKey, &obj);
if (!found) {
@@ -1161,6 +1173,39 @@
freeACaptureRequest(request);
break;
}
+ case kWhatLogicalCaptureFail:
+ {
+ ACameraCaptureSession_logicalCamera_captureCallback_failed onFail;
+ found = msg->findPointer(kCallbackFpKey, (void**) &onFail);
+ if (!found) {
+ ALOGE("%s: Cannot find capture fail callback!", __FUNCTION__);
+ return;
+ }
+ if (onFail == nullptr) {
+ return;
+ }
+
+ found = msg->findObject(kCaptureFailureKey, &obj);
+ if (!found) {
+ ALOGE("%s: Cannot find capture failure!", __FUNCTION__);
+ return;
+ }
+ sp<CameraCaptureFailure> failureSp(
+ static_cast<CameraCaptureFailure*>(obj.get()));
+ ALogicalCameraCaptureFailure failure;
+ AString physicalCameraId;
+ found = msg->findString(kFailingPhysicalCameraId, &physicalCameraId);
+ if (found && !physicalCameraId.empty()) {
+ failure.physicalCameraId = physicalCameraId.c_str();
+ } else {
+ failure.physicalCameraId = nullptr;
+ }
+ failure.captureFailure = *failureSp;
+ ACaptureRequest* request = allocateACaptureRequest(requestSp, device->getId());
+ (*onFail)(context, session.get(), request, &failure);
+ freeACaptureRequest(request);
+ break;
+ }
case kWhatCaptureSeqEnd:
{
ACameraCaptureSession_captureCallback_sequenceEnd onSeqEnd;
@@ -1256,6 +1301,7 @@
if (cbs != nullptr) {
mOnCaptureCompleted = cbs->onCaptureCompleted;
+ mOnCaptureFailed = cbs->onCaptureFailed;
}
}
@@ -1271,6 +1317,7 @@
if (lcbs != nullptr) {
mOnLogicalCameraCaptureCompleted = lcbs->onLogicalCameraCaptureCompleted;
+ mOnLogicalCameraCaptureFailed = lcbs->onLogicalCameraCaptureFailed;
}
}
@@ -1368,8 +1415,9 @@
CameraDevice::ServiceCallback::onDeviceError(
ErrorCode errorCode,
const CaptureResultExtras& resultExtras) {
- ALOGD("Device error received, code %d, frame number %" PRId64 ", request ID %d, subseq ID %d",
- errorCode, resultExtras.frameNumber, resultExtras.requestId, resultExtras.burstId);
+ ALOGD("Device error received, code %d, frame number %" PRId64 ", request ID %d, subseq ID %d"
+ " physical camera ID %s", errorCode, resultExtras.frameNumber, resultExtras.requestId,
+ resultExtras.burstId, resultExtras.errorPhysicalCameraId.c_str());
auto ret = Void();
sp<CameraDevice> dev = mDevice.promote();
if (dev == nullptr) {
diff --git a/camera/ndk/ndk_vendor/impl/ACameraDevice.h b/camera/ndk/ndk_vendor/impl/ACameraDevice.h
index 47e6f56..7036017 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraDevice.h
+++ b/camera/ndk/ndk_vendor/impl/ACameraDevice.h
@@ -242,6 +242,7 @@
kWhatCaptureResult, // onCaptureProgressed, onCaptureCompleted
kWhatLogicalCaptureResult, // onLogicalCameraCaptureCompleted
kWhatCaptureFail, // onCaptureFailed
+ kWhatLogicalCaptureFail, // onLogicalCameraCaptureFailed
kWhatCaptureSeqEnd, // onCaptureSequenceCompleted
kWhatCaptureSeqAbort, // onCaptureSequenceAborted
kWhatCaptureBufferLost,// onCaptureBufferLost
@@ -261,6 +262,7 @@
static const char* kSequenceIdKey;
static const char* kFrameNumberKey;
static const char* kAnwKey;
+ static const char* kFailingPhysicalCameraId;
class CallbackHandler : public AHandler {
public:
@@ -307,6 +309,7 @@
mOnCaptureProgressed = nullptr;
mOnCaptureCompleted = nullptr;
mOnLogicalCameraCaptureCompleted = nullptr;
+ mOnLogicalCameraCaptureFailed = nullptr;
mOnCaptureFailed = nullptr;
mOnCaptureSequenceCompleted = nullptr;
mOnCaptureSequenceAborted = nullptr;
@@ -315,7 +318,6 @@
mContext = cbs->context;
mOnCaptureStarted = cbs->onCaptureStarted;
mOnCaptureProgressed = cbs->onCaptureProgressed;
- mOnCaptureFailed = cbs->onCaptureFailed;
mOnCaptureSequenceCompleted = cbs->onCaptureSequenceCompleted;
mOnCaptureSequenceAborted = cbs->onCaptureSequenceAborted;
mOnCaptureBufferLost = cbs->onCaptureBufferLost;
@@ -332,6 +334,7 @@
ACameraCaptureSession_captureCallback_result mOnCaptureProgressed;
ACameraCaptureSession_captureCallback_result mOnCaptureCompleted;
ACameraCaptureSession_logicalCamera_captureCallback_result mOnLogicalCameraCaptureCompleted;
+ ACameraCaptureSession_logicalCamera_captureCallback_failed mOnLogicalCameraCaptureFailed;
ACameraCaptureSession_captureCallback_failed mOnCaptureFailed;
ACameraCaptureSession_captureCallback_sequenceEnd mOnCaptureSequenceCompleted;
ACameraCaptureSession_captureCallback_sequenceAbort mOnCaptureSequenceAborted;
diff --git a/camera/ndk/ndk_vendor/impl/utils.h b/camera/ndk/ndk_vendor/impl/utils.h
index 2f1006d..a03c7bc 100644
--- a/camera/ndk/ndk_vendor/impl/utils.h
+++ b/camera/ndk/ndk_vendor/impl/utils.h
@@ -99,7 +99,15 @@
return mOutputConfiguration;
}
- OutputConfigurationWrapper() = default;
+ OutputConfigurationWrapper() {
+ mOutputConfiguration.rotation = OutputConfiguration::Rotation::R0;
+ // The ndk currently doesn't support deferred surfaces
+ mOutputConfiguration.isDeferred = false;
+ mOutputConfiguration.width = 0;
+ mOutputConfiguration.height = 0;
+ // ndk doesn't support inter OutputConfiguration buffer sharing.
+ mOutputConfiguration.windowGroupId = -1;
+ };
OutputConfigurationWrapper(OutputConfiguration &outputConfiguration)
: mOutputConfiguration((outputConfiguration)) { }
diff --git a/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp b/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
index c51f93b..7368775 100644
--- a/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
+++ b/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
@@ -323,7 +323,13 @@
}
ch->mCompletedCaptureCallbackCount++;
},
- nullptr, // onCaptureFailed
+ [] (void * /*ctx*/, ACameraCaptureSession* /*session*/, ACaptureRequest* /*request*/,
+ ALogicalCameraCaptureFailure* failure) {
+ if (failure->physicalCameraId) {
+ ALOGD("%s: Physical camera id: %s result failure", __FUNCTION__,
+ failure->physicalCameraId);
+ }
+ },
nullptr, // onCaptureSequenceCompleted
nullptr, // onCaptureSequenceAborted
nullptr, // onCaptureBufferLost
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index c361690..7aa655f 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -48,6 +48,7 @@
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MediaCodecConstants.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaMuxer.h>
#include <media/stagefright/PersistentSurface.h>
@@ -83,8 +84,6 @@
using android::DISPLAY_ORIENTATION_0;
using android::DISPLAY_ORIENTATION_180;
using android::DISPLAY_ORIENTATION_90;
-using android::INFO_FORMAT_CHANGED;
-using android::INFO_OUTPUT_BUFFERS_CHANGED;
using android::INVALID_OPERATION;
using android::NAME_NOT_FOUND;
using android::NO_ERROR;
@@ -113,6 +112,7 @@
static uint32_t gVideoHeight = 0;
static uint32_t gBitRate = 20000000; // 20Mbps
static uint32_t gTimeLimitSec = kMaxTimeLimitSec;
+static uint32_t gBframes = 0;
// Set by signal handler to stop recording.
static volatile bool gStopRequested = false;
@@ -184,13 +184,18 @@
}
sp<AMessage> format = new AMessage;
- format->setInt32("width", gVideoWidth);
- format->setInt32("height", gVideoHeight);
- format->setString("mime", kMimeTypeAvc);
- format->setInt32("color-format", OMX_COLOR_FormatAndroidOpaque);
- format->setInt32("bitrate", gBitRate);
- format->setFloat("frame-rate", displayFps);
- format->setInt32("i-frame-interval", 10);
+ format->setInt32(KEY_WIDTH, gVideoWidth);
+ format->setInt32(KEY_HEIGHT, gVideoHeight);
+ format->setString(KEY_MIME, kMimeTypeAvc);
+ format->setInt32(KEY_COLOR_FORMAT, OMX_COLOR_FormatAndroidOpaque);
+ format->setInt32(KEY_BIT_RATE, gBitRate);
+ format->setFloat(KEY_FRAME_RATE, displayFps);
+ format->setInt32(KEY_I_FRAME_INTERVAL, 10);
+ format->setInt32(KEY_MAX_B_FRAMES, gBframes);
+ if (gBframes > 0) {
+ format->setInt32(KEY_PROFILE, AVCProfileMain);
+ format->setInt32(KEY_LEVEL, AVCLevel41);
+ }
sp<android::ALooper> looper = new android::ALooper;
looper->setName("screenrecord_looper");
@@ -478,7 +483,7 @@
case -EAGAIN: // INFO_TRY_AGAIN_LATER
ALOGV("Got -EAGAIN, looping");
break;
- case INFO_FORMAT_CHANGED: // INFO_OUTPUT_FORMAT_CHANGED
+ case android::INFO_FORMAT_CHANGED: // INFO_OUTPUT_FORMAT_CHANGED
{
// Format includes CSD, which we must provide to muxer.
ALOGV("Encoder format changed");
@@ -495,7 +500,7 @@
}
}
break;
- case INFO_OUTPUT_BUFFERS_CHANGED: // INFO_OUTPUT_BUFFERS_CHANGED
+ case android::INFO_OUTPUT_BUFFERS_CHANGED: // INFO_OUTPUT_BUFFERS_CHANGED
// Not expected for an encoder; handle it anyway.
ALOGV("Encoder buffers changed");
err = encoder->getOutputBuffers(&buffers);
@@ -960,6 +965,7 @@
{ "codec-name", required_argument, NULL, 'N' },
{ "monotonic-time", no_argument, NULL, 'm' },
{ "persistent-surface", no_argument, NULL, 'p' },
+ { "bframes", required_argument, NULL, 'B' },
{ NULL, 0, NULL, 0 }
};
@@ -1052,6 +1058,11 @@
case 'p':
gPersistentSurface = true;
break;
+ case 'B':
+ if (parseValueWithUnit(optarg, &gBframes) != NO_ERROR) {
+ return 2;
+ }
+ break;
default:
if (ic != '?') {
fprintf(stderr, "getopt_long returned unexpected value 0x%x\n", ic);
diff --git a/drm/libdrmframework/Android.bp b/drm/libdrmframework/Android.bp
index 43ba72b..940c17d 100644
--- a/drm/libdrmframework/Android.bp
+++ b/drm/libdrmframework/Android.bp
@@ -39,4 +39,3 @@
cflags: ["-Werror"],
}
-subdirs = ["plugins/*"]
diff --git a/drm/libdrmframework/plugins/common/Android.bp b/drm/libdrmframework/plugins/common/Android.bp
deleted file mode 100644
index 213e57f..0000000
--- a/drm/libdrmframework/plugins/common/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["util"]
diff --git a/drm/libdrmframework/plugins/forward-lock/Android.bp b/drm/libdrmframework/plugins/forward-lock/Android.bp
deleted file mode 100644
index f884c14..0000000
--- a/drm/libdrmframework/plugins/forward-lock/Android.bp
+++ /dev/null
@@ -1,4 +0,0 @@
-subdirs = [
- "FwdLockEngine",
- "internal-format",
-]
diff --git a/drm/libdrmframework/plugins/forward-lock/internal-format/Android.bp b/drm/libdrmframework/plugins/forward-lock/internal-format/Android.bp
deleted file mode 100644
index 9f58e26..0000000
--- a/drm/libdrmframework/plugins/forward-lock/internal-format/Android.bp
+++ /dev/null
@@ -1,5 +0,0 @@
-subdirs = [
- "common",
- "converter",
- "decoder",
-]
diff --git a/drm/mediadrm/Android.bp b/drm/mediadrm/Android.bp
deleted file mode 100644
index b9f07f1..0000000
--- a/drm/mediadrm/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["plugins/*"]
diff --git a/drm/mediadrm/plugins/clearkey/default/Android.bp b/drm/mediadrm/plugins/clearkey/default/Android.bp
index 7ba5708..9803d32 100644
--- a/drm/mediadrm/plugins/clearkey/default/Android.bp
+++ b/drm/mediadrm/plugins/clearkey/default/Android.bp
@@ -61,7 +61,3 @@
},
}
-//########################################################################
-// Build unit tests
-
-subdirs = ["tests"]
diff --git a/include/soundtrigger/ISoundTriggerHwService.h b/include/soundtrigger/ISoundTriggerHwService.h
index ae0cb01..1faeb0f 100644
--- a/include/soundtrigger/ISoundTriggerHwService.h
+++ b/include/soundtrigger/ISoundTriggerHwService.h
@@ -33,12 +33,14 @@
DECLARE_META_INTERFACE(SoundTriggerHwService);
- virtual status_t listModules(struct sound_trigger_module_descriptor *modules,
+ virtual status_t listModules(const String16& opPackageName,
+ struct sound_trigger_module_descriptor *modules,
uint32_t *numModules) = 0;
- virtual status_t attach(const sound_trigger_module_handle_t handle,
- const sp<ISoundTriggerClient>& client,
- sp<ISoundTrigger>& module) = 0;
+ virtual status_t attach(const String16& opPackageName,
+ const sound_trigger_module_handle_t handle,
+ const sp<ISoundTriggerClient>& client,
+ sp<ISoundTrigger>& module) = 0;
virtual status_t setCaptureState(bool active) = 0;
};
diff --git a/include/soundtrigger/SoundTrigger.h b/include/soundtrigger/SoundTrigger.h
index 2e2ff7a..ccc61dc 100644
--- a/include/soundtrigger/SoundTrigger.h
+++ b/include/soundtrigger/SoundTrigger.h
@@ -36,10 +36,12 @@
virtual ~SoundTrigger();
- static status_t listModules(struct sound_trigger_module_descriptor *modules,
+ static status_t listModules(const String16& opPackageName,
+ struct sound_trigger_module_descriptor *modules,
uint32_t *numModules);
- static sp<SoundTrigger> attach(const sound_trigger_module_handle_t module,
- const sp<SoundTriggerCallback>& callback);
+ static sp<SoundTrigger> attach(const String16& opPackageName,
+ const sound_trigger_module_handle_t module,
+ const sp<SoundTriggerCallback>& callback);
static status_t setCaptureState(bool active);
diff --git a/media/codec2/components/aac/C2SoftAacEnc.cpp b/media/codec2/components/aac/C2SoftAacEnc.cpp
index 137e775..71eb1ac 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.cpp
+++ b/media/codec2/components/aac/C2SoftAacEnc.cpp
@@ -153,7 +153,7 @@
mSentCodecSpecificData(false),
mInputTimeSet(false),
mInputSize(0),
- mInputTimeUs(-1ll),
+ mInputTimeUs(0),
mSignalledError(false),
mOutIndex(0u) {
}
@@ -179,7 +179,7 @@
mSentCodecSpecificData = false;
mInputTimeSet = false;
mInputSize = 0u;
- mInputTimeUs = -1ll;
+ mInputTimeUs = 0;
mSignalledError = false;
return C2_OK;
}
@@ -197,6 +197,7 @@
mSentCodecSpecificData = false;
mInputTimeSet = false;
mInputSize = 0u;
+ mInputTimeUs = 0;
return C2_OK;
}
@@ -558,6 +559,7 @@
mSentCodecSpecificData = false;
mInputTimeSet = false;
mInputSize = 0u;
+ mInputTimeUs = 0;
// TODO: we don't have any pending work at this time to drain.
return C2_OK;
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 55a525e..d1fa920 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -1525,6 +1525,7 @@
mPending.splice(mPending.end(), mStash);
mDepth = depth;
}
+
void CCodecBufferChannel::ReorderStash::setKey(C2Config::ordinal_key_t key) {
mPending.splice(mPending.end(), mStash);
mKey = key;
@@ -1547,13 +1548,25 @@
int64_t timestamp,
int32_t flags,
const C2WorkOrdinalStruct &ordinal) {
- auto it = mStash.begin();
- for (; it != mStash.end(); ++it) {
- if (less(ordinal, it->ordinal)) {
- break;
+ bool eos = flags & MediaCodec::BUFFER_FLAG_EOS;
+ if (!buffer && eos) {
+ // TRICKY: we may be violating ordering of the stash here. Because we
+ // don't expect any more emplace() calls after this, the ordering should
+ // not matter.
+ mStash.emplace_back(buffer, timestamp, flags, ordinal);
+ } else {
+ flags = flags & ~MediaCodec::BUFFER_FLAG_EOS;
+ auto it = mStash.begin();
+ for (; it != mStash.end(); ++it) {
+ if (less(ordinal, it->ordinal)) {
+ break;
+ }
+ }
+ mStash.emplace(it, buffer, timestamp, flags, ordinal);
+ if (eos) {
+ mStash.back().flags = mStash.back().flags | MediaCodec::BUFFER_FLAG_EOS;
}
}
- mStash.emplace(it, buffer, timestamp, flags, ordinal);
while (!mStash.empty() && mStash.size() > mDepth) {
mPending.push_back(mStash.front());
mStash.pop_front();
@@ -2815,8 +2828,9 @@
outBuffer->meta()->setInt64("timeUs", entry.timestamp);
outBuffer->meta()->setInt32("flags", entry.flags);
- ALOGV("[%s] sendOutputBuffers: out buffer index = %zu [%p] => %p + %zu",
- mName, index, outBuffer.get(), outBuffer->data(), outBuffer->size());
+ ALOGV("[%s] sendOutputBuffers: out buffer index = %zu [%p] => %p + %zu (%lld)",
+ mName, index, outBuffer.get(), outBuffer->data(), outBuffer->size(),
+ (long long)entry.timestamp);
mCallback->onOutputBufferAvailable(index, outBuffer);
}
}
diff --git a/media/extractors/Android.bp b/media/extractors/Android.bp
deleted file mode 100644
index e8176cf..0000000
--- a/media/extractors/Android.bp
+++ /dev/null
@@ -1,3 +0,0 @@
-subdirs = [
- "*",
-]
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index 5ff1c59..5a31c58 100755
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -24,6 +24,7 @@
#include <stdlib.h>
#include <string.h>
+#include <log/log.h>
#include <utils/Log.h>
#include "AC4Parser.h"
@@ -81,7 +82,7 @@
const Trex *trex,
off64_t firstMoofOffset,
const sp<ItemTable> &itemTable,
- int32_t elstShiftStartTicks);
+ uint64_t elstShiftStartTicks);
virtual status_t init();
virtual media_status_t start();
@@ -147,7 +148,7 @@
// Start offset from composition time to presentation time.
// Support shift only for video tracks through mElstShiftStartTicks for now.
- int32_t mElstShiftStartTicks;
+ uint64_t mElstShiftStartTicks;
size_t parseNALSize(const uint8_t *data) const;
status_t parseChunk(off64_t *offset);
@@ -397,7 +398,6 @@
while (track) {
Track *next = track->next;
- AMediaFormat_delete(track->meta);
delete track;
track = next;
}
@@ -672,7 +672,6 @@
ALOGV("adding HEIF image track %u", imageIndex);
Track *track = new Track;
- track->next = NULL;
if (mLastTrack != NULL) {
mLastTrack->next = track;
} else {
@@ -682,10 +681,7 @@
track->meta = meta;
AMediaFormat_setInt32(track->meta, AMEDIAFORMAT_KEY_TRACK_ID, imageIndex);
- track->includes_expensive_metadata = false;
- track->skipTrack = false;
track->timescale = 1000000;
- track->elstShiftStartTicks = 0;
}
}
@@ -967,7 +963,6 @@
ALOGV("adding new track");
Track *track = new Track;
- track->next = NULL;
if (mLastTrack) {
mLastTrack->next = track;
} else {
@@ -975,15 +970,9 @@
}
mLastTrack = track;
- track->includes_expensive_metadata = false;
- track->skipTrack = false;
- track->timescale = 0;
track->meta = AMediaFormat_new();
AMediaFormat_setString(track->meta,
AMEDIAFORMAT_KEY_MIME, "application/octet-stream");
- track->has_elst = false;
- track->subsample_encryption = false;
- track->elstShiftStartTicks = 0;
}
off64_t stop_offset = *offset + chunk_size;
@@ -1033,6 +1022,7 @@
mLastTrack->skipTrack = true;
}
+
if (mLastTrack->skipTrack) {
ALOGV("skipping this track...");
Track *cur = mFirstTrack;
@@ -1053,6 +1043,21 @@
return OK;
}
+
+ // place things we built elsewhere into their final locations
+
+ // put aggregated tx3g data into the metadata
+ if (mLastTrack->mTx3gFilled > 0) {
+ ALOGV("Putting %zu bytes of tx3g data into meta data",
+ mLastTrack->mTx3gFilled);
+ AMediaFormat_setBuffer(mLastTrack->meta,
+ AMEDIAFORMAT_KEY_TEXT_FORMAT_DATA,
+ mLastTrack->mTx3gBuffer, mLastTrack->mTx3gFilled);
+ // drop it now to reduce our footprint
+ free(mLastTrack->mTx3gBuffer);
+ mLastTrack->mTx3gBuffer = NULL;
+ }
+
} else if (chunk_type == FOURCC("moov")) {
mInitCheck = OK;
@@ -2553,41 +2558,55 @@
if (mLastTrack == NULL)
return ERROR_MALFORMED;
- void *data;
- size_t size = 0;
- if (!AMediaFormat_getBuffer(mLastTrack->meta,
- AMEDIAFORMAT_KEY_TEXT_FORMAT_DATA, &data, &size)) {
- size = 0;
- }
-
- if ((chunk_size > SIZE_MAX) || (SIZE_MAX - chunk_size <= size)) {
+ // complain about ridiculous chunks
+ if (chunk_size > kMaxAtomSize) {
return ERROR_MALFORMED;
}
- uint8_t *buffer = new (std::nothrow) uint8_t[size + chunk_size];
- if (buffer == NULL) {
+ // complain about empty atoms
+ if (chunk_data_size <= 0) {
+ ALOGE("b/124330204");
+ android_errorWriteLog(0x534e4554, "124330204");
return ERROR_MALFORMED;
}
- if (size > 0) {
- memcpy(buffer, data, size);
+ // should fill buffer based on "data_offset" and "chunk_data_size"
+ // instead of *offset and chunk_size;
+ // but we've been feeding the extra data to consumers for multiple releases and
+ // if those apps are compensating for it, we'd break them with such a change
+ //
+
+ if (mLastTrack->mTx3gSize - mLastTrack->mTx3gFilled < chunk_size) {
+ size_t growth = kTx3gGrowth;
+ if (growth < chunk_size) {
+ growth = chunk_size;
+ }
+ // although this disallows 2 tx3g atoms of nearly kMaxAtomSize...
+ if ((uint64_t) mLastTrack->mTx3gSize + growth > kMaxAtomSize) {
+ ALOGE("b/124330204 - too much space");
+ android_errorWriteLog(0x534e4554, "124330204");
+ return ERROR_MALFORMED;
+ }
+ uint8_t *updated = (uint8_t *)realloc(mLastTrack->mTx3gBuffer,
+ mLastTrack->mTx3gSize + growth);
+ if (updated == NULL) {
+ return ERROR_MALFORMED;
+ }
+ mLastTrack->mTx3gBuffer = updated;
+ mLastTrack->mTx3gSize += growth;
}
- if ((size_t)(mDataSource->readAt(*offset, buffer + size, chunk_size))
+ if ((size_t)(mDataSource->readAt(*offset,
+ mLastTrack->mTx3gBuffer + mLastTrack->mTx3gFilled,
+ chunk_size))
< chunk_size) {
- delete[] buffer;
- buffer = NULL;
// advance read pointer so we don't end up reading this again
*offset += chunk_size;
return ERROR_IO;
}
- AMediaFormat_setBuffer(mLastTrack->meta,
- AMEDIAFORMAT_KEY_TEXT_FORMAT_DATA, buffer, size + chunk_size);
-
- delete[] buffer;
-
+ mLastTrack->mTx3gFilled += chunk_size;
*offset += chunk_size;
break;
}
@@ -4040,7 +4059,7 @@
if (track->has_elst and !strncasecmp("video/", mime, 6) and track->elst_media_time > 0) {
track->elstShiftStartTicks = track->elst_media_time;
- ALOGV("video track->elstShiftStartTicks :%" PRId64, track->elst_media_time);
+ ALOGV("video track->elstShiftStartTicks :%" PRIu64, track->elstShiftStartTicks);
}
MPEG4Source *source = new MPEG4Source(
@@ -4450,7 +4469,7 @@
const Trex *trex,
off64_t firstMoofOffset,
const sp<ItemTable> &itemTable,
- int32_t elstShiftStartTicks)
+ uint64_t elstShiftStartTicks)
: mFormat(format),
mDataSource(dataSource),
mTimescale(timeScale),
@@ -4576,7 +4595,7 @@
// Start offset should be less or equal to composition time of first sample.
// ISO : sample_composition_time_offset, version 0 (unsigned) for major brands.
mElstShiftStartTicks = std::min(mElstShiftStartTicks,
- (*mCurrentSamples.begin()).compositionOffset);
+ (uint64_t)(*mCurrentSamples.begin()).compositionOffset);
}
return err;
}
@@ -4586,7 +4605,7 @@
err = mSampleTable->getMetaDataForSample(0, NULL, NULL, &firstSampleCTS);
// Start offset should be less or equal to composition time of first sample.
// Composition time stamp of first sample cannot be negative.
- mElstShiftStartTicks = std::min(mElstShiftStartTicks, (int32_t)firstSampleCTS);
+ mElstShiftStartTicks = std::min(mElstShiftStartTicks, firstSampleCTS);
}
return err;
@@ -5239,8 +5258,30 @@
sampleCtsOffset = 0;
}
- if (size < (off64_t)sampleCount * bytesPerSample) {
- return -EINVAL;
+ if (bytesPerSample != 0) {
+ if (size < (off64_t)sampleCount * bytesPerSample) {
+ return -EINVAL;
+ }
+ } else {
+ if (sampleDuration == 0) {
+ ALOGW("b/123389881 sampleDuration == 0");
+ android_errorWriteLog(0x534e4554, "124389881 zero");
+ return -EINVAL;
+ }
+
+ // apply some sanity (vs strict legality) checks
+ //
+ // clamp the count of entries in the trun box, to avoid spending forever parsing
+ // this box. Clamping (vs error) lets us play *something*.
+ // 1 million is about 400 msecs on a Pixel3, should be no more than a couple seconds
+ // on the slowest devices.
+ static constexpr uint32_t kMaxTrunSampleCount = 1000000;
+ if (sampleCount > kMaxTrunSampleCount) {
+ ALOGW("b/123389881 clamp sampleCount(%u) @ kMaxTrunSampleCount(%u)",
+ sampleCount, kMaxTrunSampleCount);
+ android_errorWriteLog(0x534e4554, "124389881 count");
+
+ }
}
Sample tmp;
@@ -5496,7 +5537,11 @@
err = mSampleTable->getMetaDataForSample(
mCurrentSampleIndex, &offset, &size, &cts, &isSyncSample, &stts);
if(err == OK) {
- cts -= mElstShiftStartTicks;
+ /* Composition Time Stamp cannot be negative. Some files have video Sample
+ * Time(STTS)delta with zero value(b/117402420). Hence subtract only
+ * min(cts, mElstShiftStartTicks), so that audio tracks can be played.
+ */
+ cts -= std::min(cts, mElstShiftStartTicks);
}
} else {
@@ -5780,8 +5825,8 @@
if (options && options->getSeekTo(&seekTimeUs, &mode)) {
seekTimeUs += ((long double)mElstShiftStartTicks * 1000000) / mTimescale;
- ALOGV("shifted seekTimeUs :%" PRId64 ", mElstShiftStartTicks:%" PRId32, seekTimeUs,
- mElstShiftStartTicks);
+ ALOGV("shifted seekTimeUs :%" PRId64 ", mElstShiftStartTicks:%" PRIu64, seekTimeUs,
+ mElstShiftStartTicks);
int numSidxEntries = mSegments.size();
if (numSidxEntries != 0) {
@@ -5837,7 +5882,7 @@
off64_t offset = 0;
size_t size = 0;
- uint32_t cts = 0;
+ uint64_t cts = 0;
bool isSyncSample = false;
bool newBuffer = false;
if (mBuffer == NULL || mCurrentSampleIndex >= mCurrentSamples.size()) {
@@ -5869,7 +5914,11 @@
offset = smpl->offset;
size = smpl->size;
cts = mCurrentTime + smpl->compositionOffset;
- cts -= mElstShiftStartTicks;
+ /* Composition Time Stamp cannot be negative. Some files have video Sample
+ * Time(STTS)delta with zero value(b/117402420). Hence subtract only
+ * min(cts, mElstShiftStartTicks), so that audio tracks can be played.
+ */
+ cts -= std::min(cts, mElstShiftStartTicks);
mCurrentTime += smpl->duration;
isSyncSample = (mCurrentSampleIndex == 0);
diff --git a/media/extractors/mp4/MPEG4Extractor.h b/media/extractors/mp4/MPEG4Extractor.h
index fadfb50..031e793 100644
--- a/media/extractors/mp4/MPEG4Extractor.h
+++ b/media/extractors/mp4/MPEG4Extractor.h
@@ -83,12 +83,47 @@
bool includes_expensive_metadata;
bool skipTrack;
bool has_elst;
+ /* signed int, ISO Spec allows media_time = -1 for other use cases.
+ * but we don't support empty edits for now.
+ */
int64_t elst_media_time;
uint64_t elst_segment_duration;
- int32_t elstShiftStartTicks;
+ // unsigned int, shift start offset only when media_time > 0.
+ uint64_t elstShiftStartTicks;
bool subsample_encryption;
+
+ uint8_t *mTx3gBuffer;
+ size_t mTx3gSize, mTx3gFilled;
+
+
+ Track() {
+ next = NULL;
+ meta = NULL;
+ timescale = 0;
+ includes_expensive_metadata = false;
+ skipTrack = false;
+ has_elst = false;
+ elst_media_time = 0;
+ elstShiftStartTicks = 0;
+ subsample_encryption = false;
+ mTx3gBuffer = NULL;
+ mTx3gSize = mTx3gFilled = 0;
+ }
+ ~Track() {
+ if (meta) {
+ AMediaFormat_delete(meta);
+ meta = NULL;
+ }
+ free (mTx3gBuffer);
+ mTx3gBuffer = NULL;
+ }
+
+ private:
+ DISALLOW_EVIL_CONSTRUCTORS(Track);
};
+ static const int kTx3gGrowth = 16 * 1024;
+
Vector<SidxEntry> mSidxEntries;
off64_t mMoofOffset;
bool mMoofFound;
diff --git a/media/libaaudio/Android.bp b/media/libaaudio/Android.bp
index 4857008..16958f9 100644
--- a/media/libaaudio/Android.bp
+++ b/media/libaaudio/Android.bp
@@ -35,4 +35,3 @@
version_script: "libaaudio.map.txt",
}
-subdirs = ["*"]
diff --git a/media/libaaudio/examples/Android.bp b/media/libaaudio/examples/Android.bp
index 639fab2..49bd5ee 100644
--- a/media/libaaudio/examples/Android.bp
+++ b/media/libaaudio/examples/Android.bp
@@ -1,5 +1,3 @@
-subdirs = ["*"]
-
cc_library_headers {
name: "libaaudio_example_utils",
export_include_dirs: ["utils"],
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index c7c42eb..49d921f 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -40,7 +40,8 @@
// V0.4.01 = add -h hang option
// fix -n option to set output buffer for -tm
// plot first glitch
-#define APP_VERSION "0.4.01"
+// V0.4.02 = allow -n0 for minimal buffer size
+#define APP_VERSION "0.4.02"
// Tag for machine readable results as property = value pairs
#define RESULT_TAG "RESULT: "
@@ -535,7 +536,7 @@
printf("INPUT stream ----------------------------------------\n");
// Use different parameters for the input.
argParser.setDeviceId(requestedInputDeviceId);
- argParser.setNumberOfBursts(AAUDIO_UNSPECIFIED);
+ argParser.setNumberOfBursts(AAudioParameters::kDefaultNumberOfBursts);
argParser.setFormat(requestedInputFormat);
argParser.setPerformanceMode(inputPerformanceLevel);
argParser.setChannelCount(requestedInputChannelCount);
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index f5ed7aa..9115778 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -36,11 +36,14 @@
aaudio_content_type_t contentType) = nullptr;
static void (*s_setInputPreset)(AAudioStreamBuilder* builder,
aaudio_input_preset_t inputPreset) = nullptr;
+static void (*s_setAllowedCapturePolicy)(AAudioStreamBuilder* builder,
+ aaudio_allowed_capture_policy_t usage) = nullptr;
static bool s_loadAttempted = false;
static aaudio_usage_t (*s_getUsage)(AAudioStream *stream) = nullptr;
static aaudio_content_type_t (*s_getContentType)(AAudioStream *stream) = nullptr;
static aaudio_input_preset_t (*s_getInputPreset)(AAudioStream *stream) = nullptr;
+static aaudio_allowed_capture_policy_t (*s_getAllowedCapturePolicy)(AAudioStream *stream) = nullptr;
// Link to test functions in shared library.
static void loadFutureFunctions() {
@@ -61,6 +64,10 @@
dlsym(handle, "AAudioStreamBuilder_setInputPreset");
if (s_setInputPreset == nullptr) goto error;
+ s_setAllowedCapturePolicy = (void (*)(AAudioStreamBuilder *, aaudio_input_preset_t))
+ dlsym(handle, "AAudioStreamBuilder_setAllowedCapturePolicy");
+ if (s_setAllowedCapturePolicy == nullptr) goto error;
+
s_getUsage = (aaudio_usage_t (*)(AAudioStream *))
dlsym(handle, "AAudioStream_getUsage");
if (s_getUsage == nullptr) goto error;
@@ -72,6 +79,10 @@
s_getInputPreset = (aaudio_input_preset_t (*)(AAudioStream *))
dlsym(handle, "AAudioStream_getInputPreset");
if (s_getInputPreset == nullptr) goto error;
+
+ s_getAllowedCapturePolicy = (aaudio_input_preset_t (*)(AAudioStream *))
+ dlsym(handle, "AAudioStream_getAllowedCapturePolicy");
+ if (s_getAllowedCapturePolicy == nullptr) goto error;
}
return;
@@ -169,6 +180,14 @@
mInputPreset = inputPreset;
}
+ aaudio_allowed_capture_policy_t getAllowedCapturePolicy() const {
+ return mAllowedCapturePolicy;
+ }
+
+ void setAllowedCapturePolicy(aaudio_allowed_capture_policy_t policy) {
+ mAllowedCapturePolicy = policy;
+ }
+
int32_t getDeviceId() const {
return mDeviceId;
}
@@ -223,8 +242,17 @@
} else if (mUsage != AAUDIO_UNSPECIFIED){
printf("WARNING: setInputPreset not supported");
}
+
+ // Call Q functions if supported.
+ if (s_setAllowedCapturePolicy != nullptr) {
+ s_setAllowedCapturePolicy(builder, mAllowedCapturePolicy);
+ } else if (mAllowedCapturePolicy != AAUDIO_UNSPECIFIED){
+ printf("WARNING: setAllowedCapturePolicy not supported");
+ }
}
+ static constexpr int32_t kDefaultNumberOfBursts = 2;
+
private:
int32_t mChannelCount = AAUDIO_UNSPECIFIED;
aaudio_format_t mFormat = AAUDIO_FORMAT_UNSPECIFIED;
@@ -238,8 +266,9 @@
aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
aaudio_input_preset_t mInputPreset = AAUDIO_UNSPECIFIED;
+ aaudio_allowed_capture_policy_t mAllowedCapturePolicy = AAUDIO_UNSPECIFIED;
- int32_t mNumberOfBursts = AAUDIO_UNSPECIFIED;
+ int32_t mNumberOfBursts = kDefaultNumberOfBursts;
int32_t mFramesPerCallback = AAUDIO_UNSPECIFIED;
};
@@ -267,6 +296,9 @@
case 'c':
setChannelCount(atoi(&arg[2]));
break;
+ case 'C':
+ setAllowedCapturePolicy(parseAllowedCapturePolicy(arg[2]));
+ break;
case 'd':
setDeviceId(atoi(&arg[2]));
break;
@@ -341,6 +373,10 @@
printf(" Default values are UNSPECIFIED unless otherwise stated.\n");
printf(" -b{bufferCapacity} frames\n");
printf(" -c{channels} for example 2 for stereo\n");
+ printf(" -C{a|s|n} set playback capture policy\n");
+ printf(" a = _ALL (default)\n");
+ printf(" s = _SYSTEM\n");
+ printf(" n = _NONE\n");
printf(" -d{deviceId} default is %d\n", AAUDIO_UNSPECIFIED);
printf(" -f{0|1|2} set format\n");
printf(" 0 = UNSPECIFIED\n");
@@ -352,7 +388,7 @@
printf(" 1 = _NEVER, never use MMAP\n");
printf(" 2 = _AUTO, use MMAP if available, default for -m with no number\n");
printf(" 3 = _ALWAYS, use MMAP or fail\n");
- printf(" -n{numberOfBursts} for setBufferSize\n");
+ printf(" -n{numberOfBursts} for setBufferSize, default %d\n", kDefaultNumberOfBursts);
printf(" -p{performanceMode} set output AAUDIO_PERFORMANCE_MODE*, default NONE\n");
printf(" n for _NONE\n");
printf(" l for _LATENCY\n");
@@ -365,6 +401,25 @@
printf(" -z{callbackSize} or block size, in frames, default = 0\n");
}
+ static aaudio_performance_mode_t parseAllowedCapturePolicy(char c) {
+ aaudio_allowed_capture_policy_t policy = AAUDIO_ALLOW_CAPTURE_BY_ALL;
+ switch (c) {
+ case 'a':
+ policy = AAUDIO_ALLOW_CAPTURE_BY_ALL;
+ break;
+ case 's':
+ policy = AAUDIO_ALLOW_CAPTURE_BY_SYSTEM;
+ break;
+ case 'n':
+ policy = AAUDIO_ALLOW_CAPTURE_BY_NONE;
+ break;
+ default:
+ printf("ERROR: invalid playback capture policy %c\n", c);
+ break;
+ }
+ return policy;
+ }
+
static aaudio_performance_mode_t parsePerformanceMode(char c) {
aaudio_performance_mode_t mode = AAUDIO_PERFORMANCE_MODE_NONE;
switch (c) {
@@ -407,17 +462,28 @@
getFormat(), AAudioStream_getFormat(stream));
int32_t framesPerBurst = AAudioStream_getFramesPerBurst(stream);
- int32_t sizeFrames = AAudioStream_getBufferSizeInFrames(stream);
printf(" Buffer: burst = %d\n", framesPerBurst);
+
+ int32_t sizeFrames = AAudioStream_getBufferSizeInFrames(stream);
if (framesPerBurst > 0) {
- printf(" Buffer: size = %d = (%d * %d) + %d\n",
+ int32_t requestedSize = getNumberOfBursts() * framesPerBurst;
+ printf(" BufferSize: requested = %4d, actual = %4d = (%d * %d) + %d\n",
+ requestedSize,
sizeFrames,
(sizeFrames / framesPerBurst),
framesPerBurst,
(sizeFrames % framesPerBurst));
+ } else {
+ printf(" BufferSize: %d\n", sizeFrames);
}
- printf(" Capacity: requested = %d, actual = %d\n", getBufferCapacity(),
- AAudioStream_getBufferCapacityInFrames(stream));
+
+ int32_t capacityFrames = AAudioStream_getBufferCapacityInFrames(stream);
+ printf(" Capacity: requested = %4d, actual = %4d = (%d * %d) + %d\n",
+ getBufferCapacity(),
+ capacityFrames,
+ (capacityFrames / framesPerBurst),
+ framesPerBurst,
+ (capacityFrames % framesPerBurst));
printf(" CallbackSize: requested = %d, actual = %d\n", getFramesPerCallback(),
AAudioStream_getFramesPerDataCallback(stream));
@@ -449,6 +515,11 @@
printf(" Is MMAP used? %s\n", AAudioStream_isMMapUsed(stream)
? "yes" : "no");
+ if (s_getAllowedCapturePolicy != nullptr) {
+ printf(" ContentType: requested = %d, actual = %d\n",
+ getAllowedCapturePolicy(), s_getAllowedCapturePolicy(stream));
+ }
+
}
int32_t getDurationSeconds() const {
diff --git a/media/libaaudio/examples/utils/AAudioSimplePlayer.h b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
index 4373fa9..fd1fc45 100644
--- a/media/libaaudio/examples/utils/AAudioSimplePlayer.h
+++ b/media/libaaudio/examples/utils/AAudioSimplePlayer.h
@@ -120,10 +120,9 @@
if (result == AAUDIO_OK) {
int32_t sizeInBursts = parameters.getNumberOfBursts();
- if (sizeInBursts > 0) {
- int32_t framesPerBurst = AAudioStream_getFramesPerBurst(mStream);
- AAudioStream_setBufferSizeInFrames(mStream, sizeInBursts * framesPerBurst);
- }
+ int32_t framesPerBurst = AAudioStream_getFramesPerBurst(mStream);
+ int32_t bufferSizeFrames = sizeInBursts * framesPerBurst;
+ AAudioStream_setBufferSizeInFrames(mStream, bufferSizeFrames);
}
AAudioStreamBuilder_delete(builder);
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
index 2b05f10..ca60233 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
@@ -31,7 +31,7 @@
#include "AAudioSimplePlayer.h"
#include "AAudioArgsParser.h"
-#define APP_VERSION "0.1.6"
+#define APP_VERSION "0.1.7"
constexpr int32_t kDefaultHangTimeMSec = 10;
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 8e36c77..ee5d089 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -37,7 +37,7 @@
/**
* This is used to represent a value that has not been specified.
- * For example, an application could use AAUDIO_UNSPECIFIED to indicate
+ * For example, an application could use {@link #AAUDIO_UNSPECIFIED} to indicate
* that is did not not care what the specific value of a parameter was
* and would accept whatever it was given.
*/
@@ -232,7 +232,8 @@
* This information is used by certain platforms or routing policies
* to make more refined volume or routing decisions.
*
- * Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ * Note that these match the equivalent values in {@link android.media.AudioAttributes}
+ * in the Android Java API.
*
* Added in API level 28.
*/
@@ -303,12 +304,13 @@
/**
* The CONTENT_TYPE attribute describes "what" you are playing.
* It expresses the general category of the content. This information is optional.
- * But in case it is known (for instance {@link #AAUDIO_CONTENT_TYPE_MOVIE} for a
- * movie streaming service or {@link #AAUDIO_CONTENT_TYPE_SPEECH} for
+ * But in case it is known (for instance AAUDIO_CONTENT_TYPE_MOVIE for a
+ * movie streaming service or AAUDIO_CONTENT_TYPE_SPEECH for
* an audio book application) this information might be used by the audio framework to
* enforce audio focus.
*
- * Note that these match the equivalent values in AudioAttributes in the Android Java API.
+ * Note that these match the equivalent values in {@link android.media.AudioAttributes}
+ * in the Android Java API.
*
* Added in API level 28.
*/
@@ -384,6 +386,48 @@
typedef int32_t aaudio_input_preset_t;
/**
+ * Specifying if audio may or may not be captured by other apps or the system.
+ *
+ * Note that these match the equivalent values in {@link android.media.AudioAttributes}
+ * in the Android Java API.
+ *
+ * Added in API level 29.
+ */
+enum {
+ /**
+ * Indicates that the audio may be captured by any app.
+ *
+ * For privacy, the following usages can not be recorded: AAUDIO_VOICE_COMMUNICATION*,
+ * AAUDIO_USAGE_NOTIFICATION*, AAUDIO_USAGE_ASSISTANCE* and {@link #AAUDIO_USAGE_ASSISTANT}.
+ *
+ * On {@link android.os.Build.VERSION_CODES#Q}, this means only {@link #AAUDIO_USAGE_MEDIA}
+ * and {@link #AAUDIO_USAGE_GAME} may be captured.
+ *
+ * See {@link android.media.AudioAttributes#ALLOW_CAPTURE_BY_ALL}.
+ */
+ AAUDIO_ALLOW_CAPTURE_BY_ALL = 1,
+ /**
+ * Indicates that the audio may only be captured by system apps.
+ *
+ * System apps can capture for many purposes like accessibility, user guidance...
+ * but have strong restriction. See
+ * {@link android.media.AudioAttributes#ALLOW_CAPTURE_BY_SYSTEM} for what the system apps
+ * can do with the capture audio.
+ */
+ AAUDIO_ALLOW_CAPTURE_BY_SYSTEM = 2,
+ /**
+ * Indicates that the audio may not be recorded by any app, even if it is a system app.
+ *
+ * It is encouraged to use {@link #AAUDIO_ALLOW_CAPTURE_BY_SYSTEM} instead of this value as system apps
+ * provide significant and useful features for the user (eg. accessibility).
+ * See {@link android.media.AudioAttributes#ALLOW_CAPTURE_BY_NONE}.
+ */
+ AAUDIO_ALLOW_CAPTURE_BY_NONE = 3,
+};
+
+typedef int32_t aaudio_allowed_capture_policy_t;
+
+/**
* These may be used with AAudioStreamBuilder_setSessionId().
*
* Added in API level 28.
@@ -452,8 +496,8 @@
*
* The deviceId is initially unspecified, meaning that the current default device will be used.
*
- * The default direction is AAUDIO_DIRECTION_OUTPUT.
- * The default sharing mode is AAUDIO_SHARING_MODE_SHARED.
+ * The default direction is {@link #AAUDIO_DIRECTION_OUTPUT}.
+ * The default sharing mode is {@link #AAUDIO_SHARING_MODE_SHARED}.
* The data format, samplesPerFrames and sampleRate are unspecified and will be
* chosen by the device when it is opened.
*
@@ -466,11 +510,11 @@
* Request an audio device identified device using an ID.
* On Android, for example, the ID could be obtained from the Java AudioManager.
*
- * The default, if you do not call this function, is AAUDIO_UNSPECIFIED,
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED},
* in which case the primary device will be used.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param deviceId device identifier or AAUDIO_UNSPECIFIED
+ * @param deviceId device identifier or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API void AAudioStreamBuilder_setDeviceId(AAudioStreamBuilder* builder,
int32_t deviceId) __INTRODUCED_IN(26);
@@ -478,7 +522,7 @@
/**
* Request a sample rate in Hertz.
*
- * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
* An optimal value will then be chosen when the stream is opened.
* After opening a stream with an unspecified value, the application must
* query for the actual value, which may vary by device.
@@ -495,7 +539,7 @@
/**
* Request a number of channels for the stream.
*
- * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
* An optimal value will then be chosen when the stream is opened.
* After opening a stream with an unspecified value, the application must
* query for the actual value, which may vary by device.
@@ -519,9 +563,9 @@
int32_t samplesPerFrame) __INTRODUCED_IN(26);
/**
- * Request a sample data format, for example AAUDIO_FORMAT_PCM_I16.
+ * Request a sample data format, for example {@link #AAUDIO_FORMAT_PCM_I16}.
*
- * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
* An optimal value will then be chosen when the stream is opened.
* After opening a stream with an unspecified value, the application must
* query for the actual value, which may vary by device.
@@ -530,7 +574,8 @@
* If a stream cannot be opened with the specified value then the open will fail.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param format common formats are AAUDIO_FORMAT_PCM_FLOAT and AAUDIO_FORMAT_PCM_I16.
+ * @param format common formats are {@link #AAUDIO_FORMAT_PCM_FLOAT} and
+ * {@link #AAUDIO_FORMAT_PCM_I16}.
*/
AAUDIO_API void AAudioStreamBuilder_setFormat(AAudioStreamBuilder* builder,
aaudio_format_t format) __INTRODUCED_IN(26);
@@ -538,13 +583,13 @@
/**
* Request a mode for sharing the device.
*
- * The default, if you do not call this function, is AAUDIO_SHARING_MODE_SHARED.
+ * The default, if you do not call this function, is {@link #AAUDIO_SHARING_MODE_SHARED}.
*
* The requested sharing mode may not be available.
* The application can query for the actual mode after the stream is opened.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param sharingMode AAUDIO_SHARING_MODE_SHARED or AAUDIO_SHARING_MODE_EXCLUSIVE
+ * @param sharingMode {@link #AAUDIO_SHARING_MODE_SHARED} or {@link #AAUDIO_SHARING_MODE_EXCLUSIVE}
*/
AAUDIO_API void AAudioStreamBuilder_setSharingMode(AAudioStreamBuilder* builder,
aaudio_sharing_mode_t sharingMode) __INTRODUCED_IN(26);
@@ -552,10 +597,10 @@
/**
* Request the direction for a stream.
*
- * The default, if you do not call this function, is AAUDIO_DIRECTION_OUTPUT.
+ * The default, if you do not call this function, is {@link #AAUDIO_DIRECTION_OUTPUT}.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param direction AAUDIO_DIRECTION_OUTPUT or AAUDIO_DIRECTION_INPUT
+ * @param direction {@link #AAUDIO_DIRECTION_OUTPUT} or {@link #AAUDIO_DIRECTION_INPUT}
*/
AAUDIO_API void AAudioStreamBuilder_setDirection(AAudioStreamBuilder* builder,
aaudio_direction_t direction) __INTRODUCED_IN(26);
@@ -564,10 +609,10 @@
* Set the requested buffer capacity in frames.
* The final AAudioStream capacity may differ, but will probably be at least this big.
*
- * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param numFrames the desired buffer capacity in frames or AAUDIO_UNSPECIFIED
+ * @param numFrames the desired buffer capacity in frames or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API void AAudioStreamBuilder_setBufferCapacityInFrames(AAudioStreamBuilder* builder,
int32_t numFrames) __INTRODUCED_IN(26);
@@ -575,16 +620,17 @@
/**
* Set the requested performance mode.
*
- * Supported modes are AAUDIO_PERFORMANCE_MODE_NONE, AAUDIO_PERFORMANCE_MODE_POWER_SAVING
- * and AAUDIO_PERFORMANCE_MODE_LOW_LATENCY.
+ * Supported modes are {@link #AAUDIO_PERFORMANCE_MODE_NONE},
+ * {@link #AAUDIO_PERFORMANCE_MODE_POWER_SAVING} * and {@link #AAUDIO_PERFORMANCE_MODE_LOW_LATENCY}.
*
- * The default, if you do not call this function, is AAUDIO_PERFORMANCE_MODE_NONE.
+ * The default, if you do not call this function, is {@link #AAUDIO_PERFORMANCE_MODE_NONE}.
*
* You may not get the mode you requested.
- * You can call AAudioStream_getPerformanceMode() to find out the final mode for the stream.
+ * You can call AAudioStream_getPerformanceMode()
+ * to find out the final mode for the stream.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param mode the desired performance mode, eg. AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
+ * @param mode the desired performance mode, eg. {@link #AAUDIO_PERFORMANCE_MODE_LOW_LATENCY}
*/
AAUDIO_API void AAudioStreamBuilder_setPerformanceMode(AAudioStreamBuilder* builder,
aaudio_performance_mode_t mode) __INTRODUCED_IN(26);
@@ -596,12 +642,12 @@
* behavior of the stream.
* This could, for example, affect how volume and focus is handled for the stream.
*
- * The default, if you do not call this function, is AAUDIO_USAGE_MEDIA.
+ * The default, if you do not call this function, is {@link #AAUDIO_USAGE_MEDIA}.
*
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param usage the desired usage, eg. AAUDIO_USAGE_GAME
+ * @param usage the desired usage, eg. {@link #AAUDIO_USAGE_GAME}
*/
AAUDIO_API void AAudioStreamBuilder_setUsage(AAudioStreamBuilder* builder,
aaudio_usage_t usage) __INTRODUCED_IN(28);
@@ -613,12 +659,12 @@
* behavior of the stream.
* This could, for example, affect whether a stream is paused when a notification occurs.
*
- * The default, if you do not call this function, is AAUDIO_CONTENT_TYPE_MUSIC.
+ * The default, if you do not call this function, is {@link #AAUDIO_CONTENT_TYPE_MUSIC}.
*
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param contentType the type of audio data, eg. AAUDIO_CONTENT_TYPE_SPEECH
+ * @param contentType the type of audio data, eg. {@link #AAUDIO_CONTENT_TYPE_SPEECH}
*/
AAUDIO_API void AAudioStreamBuilder_setContentType(AAudioStreamBuilder* builder,
aaudio_content_type_t contentType) __INTRODUCED_IN(28);
@@ -631,7 +677,7 @@
* This could, for example, affect which microphones are used and how the
* recorded data is processed.
*
- * The default, if you do not call this function, is AAUDIO_INPUT_PRESET_VOICE_RECOGNITION.
+ * The default, if you do not call this function, is {@link #AAUDIO_INPUT_PRESET_VOICE_RECOGNITION}.
* That is because VOICE_RECOGNITION is the preset with the lowest latency
* on many platforms.
*
@@ -643,14 +689,30 @@
AAUDIO_API void AAudioStreamBuilder_setInputPreset(AAudioStreamBuilder* builder,
aaudio_input_preset_t inputPreset) __INTRODUCED_IN(28);
+/**
+ * Specify whether this stream audio may or may not be captured by other apps or the system.
+ *
+ * The default is {@link #AAUDIO_ALLOW_CAPTURE_BY_ALL}.
+ *
+ * Note that an application can also set its global policy, in which case the most restrictive
+ * policy is always applied. See {@link android.media.AudioAttributes#setAllowedCapturePolicy(int)}
+ *
+ * Added in API level 29.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param inputPreset the desired level of opt-out from being captured.
+ */
+AAUDIO_API void AAudioStreamBuilder_setAllowedCapturePolicy(AAudioStreamBuilder* builder,
+ aaudio_allowed_capture_policy_t capturePolicy) __INTRODUCED_IN(29);
+
/** Set the requested session ID.
*
* The session ID can be used to associate a stream with effects processors.
* The effects are controlled using the Android AudioEffect Java API.
*
- * The default, if you do not call this function, is AAUDIO_SESSION_ID_NONE.
+ * The default, if you do not call this function, is {@link #AAUDIO_SESSION_ID_NONE}.
*
- * If set to AAUDIO_SESSION_ID_ALLOCATE then a session ID will be allocated
+ * If set to {@link #AAUDIO_SESSION_ID_ALLOCATE} then a session ID will be allocated
* when the stream is opened.
*
* The allocated session ID can be obtained by calling AAudioStream_getSessionId()
@@ -668,7 +730,7 @@
* Added in API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param sessionId an allocated sessionID or AAUDIO_SESSION_ID_ALLOCATE
+ * @param sessionId an allocated sessionID or {@link #AAUDIO_SESSION_ID_ALLOCATE}
*/
AAUDIO_API void AAudioStreamBuilder_setSessionId(AAudioStreamBuilder* builder,
aaudio_session_id_t sessionId) __INTRODUCED_IN(28);
@@ -751,15 +813,16 @@
*
* Note that when using this callback, the audio data will be passed in or out
* of the function as an argument.
- * So you cannot call AAudioStream_write() or AAudioStream_read() on the same stream
- * that has an active data callback.
+ * So you cannot call AAudioStream_write() or AAudioStream_read()
+ * on the same stream that has an active data callback.
*
- * The callback function will start being called after AAudioStream_requestStart() is called.
+ * The callback function will start being called after AAudioStream_requestStart()
+ * is called.
* It will stop being called after AAudioStream_requestPause() or
* AAudioStream_requestStop() is called.
*
* This callback function will be called on a real-time thread owned by AAudio. See
- * {@link AAudioStream_dataCallback} for more information.
+ * {@link #AAudioStream_dataCallback} for more information.
*
* Note that the AAudio callbacks will never be called simultaneously from multiple threads.
*
@@ -773,9 +836,9 @@
/**
* Set the requested data callback buffer size in frames.
- * See {@link AAudioStream_dataCallback}.
+ * See {@link #AAudioStream_dataCallback}.
*
- * The default, if you do not call this function, is AAUDIO_UNSPECIFIED.
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
*
* For the lowest possible latency, do not call this function. AAudio will then
* call the dataProc callback function with whatever size is optimal.
@@ -792,7 +855,7 @@
* half the buffer capacity, to allow double buffering.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @param numFrames the desired buffer size in frames or AAUDIO_UNSPECIFIED
+ * @param numFrames the desired buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API void AAudioStreamBuilder_setFramesPerDataCallback(AAudioStreamBuilder* builder,
int32_t numFrames) __INTRODUCED_IN(26);
@@ -853,12 +916,12 @@
/**
* Open a stream based on the options in the StreamBuilder.
*
- * AAudioStream_close must be called when finished with the stream to recover
+ * AAudioStream_close() must be called when finished with the stream to recover
* the memory and to free the associated resources.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param stream pointer to a variable to receive the new stream reference
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStreamBuilder_openStream(AAudioStreamBuilder* builder,
AAudioStream** stream) __INTRODUCED_IN(26);
@@ -867,7 +930,7 @@
* Delete the resources associated with the StreamBuilder.
*
* @param builder reference provided by AAudio_createStreamBuilder()
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStreamBuilder_delete(AAudioStreamBuilder* builder)
__INTRODUCED_IN(26);
@@ -880,7 +943,7 @@
* Free the resources associated with a stream created by AAudioStreamBuilder_openStream()
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_close(AAudioStream* stream) __INTRODUCED_IN(26);
@@ -888,24 +951,26 @@
* Asynchronously request to start playing the stream. For output streams, one should
* write to the stream to fill the buffer before starting.
* Otherwise it will underflow.
- * After this call the state will be in AAUDIO_STREAM_STATE_STARTING or AAUDIO_STREAM_STATE_STARTED.
+ * After this call the state will be in {@link #AAUDIO_STREAM_STATE_STARTING} or
+ * {@link #AAUDIO_STREAM_STATE_STARTED}.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestStart(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Asynchronous request for the stream to pause.
* Pausing a stream will freeze the data flow but not flush any buffers.
- * Use AAudioStream_Start() to resume playback after a pause.
- * After this call the state will be in AAUDIO_STREAM_STATE_PAUSING or AAUDIO_STREAM_STATE_PAUSED.
+ * Use AAudioStream_requestStart() to resume playback after a pause.
+ * After this call the state will be in {@link #AAUDIO_STREAM_STATE_PAUSING} or
+ * {@link #AAUDIO_STREAM_STATE_PAUSED}.
*
- * This will return AAUDIO_ERROR_UNIMPLEMENTED for input streams.
+ * This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
* For input streams use AAudioStream_requestStop().
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestPause(AAudioStream* stream) __INTRODUCED_IN(26);
@@ -914,32 +979,34 @@
* Flushing will discard any pending data.
* This call only works if the stream is pausing or paused. TODO review
* Frame counters are not reset by a flush. They may be advanced.
- * After this call the state will be in AAUDIO_STREAM_STATE_FLUSHING or AAUDIO_STREAM_STATE_FLUSHED.
+ * After this call the state will be in {@link #AAUDIO_STREAM_STATE_FLUSHING} or
+ * {@link #AAUDIO_STREAM_STATE_FLUSHED}.
*
- * This will return AAUDIO_ERROR_UNIMPLEMENTED for input streams.
+ * This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestFlush(AAudioStream* stream) __INTRODUCED_IN(26);
/**
* Asynchronous request for the stream to stop.
* The stream will stop after all of the data currently buffered has been played.
- * After this call the state will be in AAUDIO_STREAM_STATE_STOPPING or AAUDIO_STREAM_STATE_STOPPED.
+ * After this call the state will be in {@link #AAUDIO_STREAM_STATE_STOPPING} or
+ * {@link #AAUDIO_STREAM_STATE_STOPPED}.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_requestStop(AAudioStream* stream) __INTRODUCED_IN(26);
/**
- * Query the current state of the client, eg. AAUDIO_STREAM_STATE_PAUSING
+ * Query the current state of the client, eg. {@link #AAUDIO_STREAM_STATE_PAUSING}
*
* This function will immediately return the state without updating the state.
* If you want to update the client state based on the server state then
* call AAudioStream_waitForStateChange() with currentState
- * set to AAUDIO_STREAM_STATE_UNKNOWN and a zero timeout.
+ * set to {@link #AAUDIO_STREAM_STATE_UNKNOWN} and a zero timeout.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
@@ -965,7 +1032,7 @@
* @param inputState The state we want to avoid.
* @param nextState Pointer to a variable that will be set to the new state.
* @param timeoutNanoseconds Maximum number of nanoseconds to wait for completion.
- * @return AAUDIO_OK or a negative error.
+ * @return {@link #AAUDIO_OK} or a negative error.
*/
AAUDIO_API aaudio_result_t AAudioStream_waitForStateChange(AAudioStream* stream,
aaudio_stream_state_t inputState, aaudio_stream_state_t *nextState,
@@ -1082,17 +1149,17 @@
* This call can be used if the application needs to know the value of numFrames before
* the stream is started. This is not normally necessary.
*
- * If a specific size was requested by calling AAudioStreamBuilder_setFramesPerDataCallback()
- * then this will be the same size.
+ * If a specific size was requested by calling
+ * AAudioStreamBuilder_setFramesPerDataCallback() then this will be the same size.
*
* If AAudioStreamBuilder_setFramesPerDataCallback() was not called then this will
- * return the size chosen by AAudio, or AAUDIO_UNSPECIFIED.
+ * return the size chosen by AAudio, or {@link #AAUDIO_UNSPECIFIED}.
*
- * AAUDIO_UNSPECIFIED indicates that the callback buffer size for this stream
+ * {@link #AAUDIO_UNSPECIFIED} indicates that the callback buffer size for this stream
* may vary from one dataProc callback to the next.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return callback buffer size in frames or AAUDIO_UNSPECIFIED
+ * @return callback buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
AAUDIO_API int32_t AAudioStream_getFramesPerDataCallback(AAudioStream* stream) __INTRODUCED_IN(26);
@@ -1202,21 +1269,22 @@
* The session ID can be used to associate a stream with effects processors.
* The effects are controlled using the Android AudioEffect Java API.
*
- * If AAudioStreamBuilder_setSessionId() was called with AAUDIO_SESSION_ID_ALLOCATE
+ * If AAudioStreamBuilder_setSessionId() was
+ * called with {@link #AAUDIO_SESSION_ID_ALLOCATE}
* then a new session ID should be allocated once when the stream is opened.
*
* If AAudioStreamBuilder_setSessionId() was called with a previously allocated
* session ID then that value should be returned.
*
* If AAudioStreamBuilder_setSessionId() was not called then this function should
- * return AAUDIO_SESSION_ID_NONE.
+ * return {@link #AAUDIO_SESSION_ID_NONE}.
*
* The sessionID for a stream should not change once the stream has been opened.
*
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return session ID or AAUDIO_SESSION_ID_NONE
+ * @return session ID or {@link #AAUDIO_SESSION_ID_NONE}
*/
AAUDIO_API aaudio_session_id_t AAudioStream_getSessionId(AAudioStream* stream) __INTRODUCED_IN(28);
@@ -1225,11 +1293,11 @@
* This can be used to synchronize audio with video or MIDI.
* It can also be used to align a recorded stream with a playback stream.
*
- * Timestamps are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
- * AAUDIO_ERROR_INVALID_STATE will be returned if the stream is not started.
+ * Timestamps are only valid when the stream is in {@link #AAUDIO_STREAM_STATE_STARTED}.
+ * {@link #AAUDIO_ERROR_INVALID_STATE} will be returned if the stream is not started.
* Note that because requestStart() is asynchronous, timestamps will not be valid until
* a short time after calling requestStart().
- * So AAUDIO_ERROR_INVALID_STATE should not be considered a fatal error.
+ * So {@link #AAUDIO_ERROR_INVALID_STATE} should not be considered a fatal error.
* Just try calling again later.
*
* If an error occurs, then the position and time will not be modified.
@@ -1240,7 +1308,7 @@
* @param clockid CLOCK_MONOTONIC or CLOCK_BOOTTIME
* @param framePosition pointer to a variable to receive the position
* @param timeNanoseconds pointer to a variable to receive the time
- * @return AAUDIO_OK or a negative error
+ * @return {@link #AAUDIO_OK} or a negative error
*/
AAUDIO_API aaudio_result_t AAudioStream_getTimestamp(AAudioStream* stream,
clockid_t clockid, int64_t *framePosition, int64_t *timeNanoseconds) __INTRODUCED_IN(26);
@@ -1261,7 +1329,7 @@
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return content type, for example AAUDIO_CONTENT_TYPE_MUSIC
+ * @return content type, for example {@link #AAUDIO_CONTENT_TYPE_MUSIC}
*/
AAUDIO_API aaudio_content_type_t AAudioStream_getContentType(AAudioStream* stream)
__INTRODUCED_IN(28);
@@ -1272,11 +1340,23 @@
* Added in API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
- * @return input preset, for example AAUDIO_INPUT_PRESET_CAMCORDER
+ * @return input preset, for example {@link #AAUDIO_INPUT_PRESET_CAMCORDER}
*/
AAUDIO_API aaudio_input_preset_t AAudioStream_getInputPreset(AAudioStream* stream)
__INTRODUCED_IN(28);
+/**
+ * Return the policy that determines whether the audio may or may not be captured
+ * by other apps or the system.
+ *
+ * Added in API level 29.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return the allowed capture policy, for example {@link #AAUDIO_ALLOW_CAPTURE_BY_ALL}
+ */
+AAUDIO_API aaudio_allowed_capture_policy_t AAudioStream_getAllowedCapturePolicy(
+ AAudioStream* stream) __INTRODUCED_IN(29);
+
#ifdef __cplusplus
}
#endif
diff --git a/media/libaaudio/libaaudio.map.txt b/media/libaaudio/libaaudio.map.txt
index cbf5921..a87ede3 100644
--- a/media/libaaudio/libaaudio.map.txt
+++ b/media/libaaudio/libaaudio.map.txt
@@ -20,6 +20,7 @@
AAudioStreamBuilder_setUsage; # introduced=28
AAudioStreamBuilder_setContentType; # introduced=28
AAudioStreamBuilder_setInputPreset; # introduced=28
+ AAudioStreamBuilder_setAllowedCapturePolicy; # introduced=29
AAudioStreamBuilder_setSessionId; # introduced=28
AAudioStreamBuilder_openStream;
AAudioStreamBuilder_delete;
@@ -49,6 +50,7 @@
AAudioStream_getUsage; # introduced=28
AAudioStream_getContentType; # introduced=28
AAudioStream_getInputPreset; # introduced=28
+ AAudioStream_getAllowedCapturePolicy; # introduced=29
AAudioStream_getFramesWritten;
AAudioStream_getFramesRead;
AAudioStream_getSessionId; # introduced=28
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index 3d1bc9b..a987fab 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -62,6 +62,8 @@
if (status != NO_ERROR) goto error;
status = parcel->writeInt32((int32_t) getInputPreset());
if (status != NO_ERROR) goto error;
+ status = parcel->writeInt32((int32_t) getAllowedCapturePolicy());
+ if (status != NO_ERROR) goto error;
status = parcel->writeInt32(getSessionId());
if (status != NO_ERROR) goto error;
return NO_ERROR;
@@ -105,6 +107,9 @@
setInputPreset((aaudio_input_preset_t) value);
status = parcel->readInt32(&value);
if (status != NO_ERROR) goto error;
+ setAllowedCapturePolicy((aaudio_allowed_capture_policy_t) value);
+ status = parcel->readInt32(&value);
+ if (status != NO_ERROR) goto error;
setSessionId(value);
return NO_ERROR;
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 0d71efc..8ae2644 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -204,6 +204,12 @@
streamBuilder->setBufferCapacity(frames);
}
+AAUDIO_API void AAudioStreamBuilder_setAllowedCapturePolicy(
+ AAudioStreamBuilder* builder, aaudio_allowed_capture_policy_t policy) {
+ AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+ streamBuilder->setAllowedCapturePolicy(policy);
+}
+
AAUDIO_API void AAudioStreamBuilder_setSessionId(AAudioStreamBuilder* builder,
aaudio_session_id_t sessionId)
{
@@ -494,6 +500,13 @@
return audioStream->getInputPreset();
}
+AAUDIO_API aaudio_allowed_capture_policy_t AAudioStream_getAllowedCapturePolicy(
+ AAudioStream* stream)
+{
+ AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+ return audioStream->getAllowedCapturePolicy();
+}
+
AAUDIO_API int32_t AAudioStream_getSessionId(AAudioStream* stream)
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index 88da53a..e5bda30 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -35,17 +35,18 @@
AAudioStreamParameters::~AAudioStreamParameters() {}
void AAudioStreamParameters::copyFrom(const AAudioStreamParameters &other) {
- mSamplesPerFrame = other.mSamplesPerFrame;
- mSampleRate = other.mSampleRate;
- mDeviceId = other.mDeviceId;
- mSessionId = other.mSessionId;
- mSharingMode = other.mSharingMode;
- mAudioFormat = other.mAudioFormat;
- mDirection = other.mDirection;
- mBufferCapacity = other.mBufferCapacity;
- mUsage = other.mUsage;
- mContentType = other.mContentType;
- mInputPreset = other.mInputPreset;
+ mSamplesPerFrame = other.mSamplesPerFrame;
+ mSampleRate = other.mSampleRate;
+ mDeviceId = other.mDeviceId;
+ mSessionId = other.mSessionId;
+ mSharingMode = other.mSharingMode;
+ mAudioFormat = other.mAudioFormat;
+ mDirection = other.mDirection;
+ mBufferCapacity = other.mBufferCapacity;
+ mUsage = other.mUsage;
+ mContentType = other.mContentType;
+ mInputPreset = other.mInputPreset;
+ mAllowedCapturePolicy = other.mAllowedCapturePolicy;
}
static aaudio_result_t isFormatValid(audio_format_t format) {
@@ -166,19 +167,32 @@
// break;
}
+ switch (mAllowedCapturePolicy) {
+ case AAUDIO_UNSPECIFIED:
+ case AAUDIO_ALLOW_CAPTURE_BY_ALL:
+ case AAUDIO_ALLOW_CAPTURE_BY_SYSTEM:
+ case AAUDIO_ALLOW_CAPTURE_BY_NONE:
+ break; // valid
+ default:
+ ALOGE("allowed capture policy not valid = %d", mAllowedCapturePolicy);
+ return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ // break;
+ }
+
return AAUDIO_OK;
}
void AAudioStreamParameters::dump() const {
- ALOGD("mDeviceId = %6d", mDeviceId);
- ALOGD("mSessionId = %6d", mSessionId);
- ALOGD("mSampleRate = %6d", mSampleRate);
- ALOGD("mSamplesPerFrame = %6d", mSamplesPerFrame);
- ALOGD("mSharingMode = %6d", (int)mSharingMode);
- ALOGD("mAudioFormat = %6d", (int)mAudioFormat);
- ALOGD("mDirection = %6d", mDirection);
- ALOGD("mBufferCapacity = %6d", mBufferCapacity);
- ALOGD("mUsage = %6d", mUsage);
- ALOGD("mContentType = %6d", mContentType);
- ALOGD("mInputPreset = %6d", mInputPreset);
+ ALOGD("mDeviceId = %6d", mDeviceId);
+ ALOGD("mSessionId = %6d", mSessionId);
+ ALOGD("mSampleRate = %6d", mSampleRate);
+ ALOGD("mSamplesPerFrame = %6d", mSamplesPerFrame);
+ ALOGD("mSharingMode = %6d", (int)mSharingMode);
+ ALOGD("mAudioFormat = %6d", (int)mAudioFormat);
+ ALOGD("mDirection = %6d", mDirection);
+ ALOGD("mBufferCapacity = %6d", mBufferCapacity);
+ ALOGD("mUsage = %6d", mUsage);
+ ALOGD("mContentType = %6d", mContentType);
+ ALOGD("mInputPreset = %6d", mInputPreset);
+ ALOGD("mAllowedCapturePolicy = %6d", mAllowedCapturePolicy);
}
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
index 6beb4b2..2e21a8d 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.h
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h
@@ -112,6 +112,14 @@
mInputPreset = inputPreset;
}
+ aaudio_allowed_capture_policy_t getAllowedCapturePolicy() const {
+ return mAllowedCapturePolicy;
+ }
+
+ void setAllowedCapturePolicy(aaudio_allowed_capture_policy_t policy) {
+ mAllowedCapturePolicy = policy;
+ }
+
aaudio_session_id_t getSessionId() const {
return mSessionId;
}
@@ -138,17 +146,18 @@
void dump() const;
private:
- int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
- int32_t mSampleRate = AAUDIO_UNSPECIFIED;
- int32_t mDeviceId = AAUDIO_UNSPECIFIED;
- aaudio_sharing_mode_t mSharingMode = AAUDIO_SHARING_MODE_SHARED;
- audio_format_t mAudioFormat = AUDIO_FORMAT_DEFAULT;
- aaudio_direction_t mDirection = AAUDIO_DIRECTION_OUTPUT;
- aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
- aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
- aaudio_input_preset_t mInputPreset = AAUDIO_UNSPECIFIED;
- int32_t mBufferCapacity = AAUDIO_UNSPECIFIED;
- aaudio_session_id_t mSessionId = AAUDIO_SESSION_ID_NONE;
+ int32_t mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+ int32_t mSampleRate = AAUDIO_UNSPECIFIED;
+ int32_t mDeviceId = AAUDIO_UNSPECIFIED;
+ aaudio_sharing_mode_t mSharingMode = AAUDIO_SHARING_MODE_SHARED;
+ audio_format_t mAudioFormat = AUDIO_FORMAT_DEFAULT;
+ aaudio_direction_t mDirection = AAUDIO_DIRECTION_OUTPUT;
+ aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
+ aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
+ aaudio_input_preset_t mInputPreset = AAUDIO_UNSPECIFIED;
+ int32_t mBufferCapacity = AAUDIO_UNSPECIFIED;
+ aaudio_allowed_capture_policy_t mAllowedCapturePolicy = AAUDIO_UNSPECIFIED;
+ aaudio_session_id_t mSessionId = AAUDIO_SESSION_ID_NONE;
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index e39a075..732d45c 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -94,6 +94,10 @@
if (mInputPreset == AAUDIO_UNSPECIFIED) {
mInputPreset = AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
}
+ mAllowedCapturePolicy = builder.getAllowedCapturePolicy();
+ if (mAllowedCapturePolicy == AAUDIO_UNSPECIFIED) {
+ mAllowedCapturePolicy = AAUDIO_ALLOW_CAPTURE_BY_ALL;
+ }
// callbacks
mFramesPerDataCallback = builder.getFramesPerDataCallback();
@@ -113,8 +117,8 @@
mPerformanceMode,
(isDataCallbackSet() ? "ON" : "OFF"),
mFramesPerDataCallback);
- ALOGI("open() usage = %d, contentType = %d, inputPreset = %d",
- mUsage, mContentType, mInputPreset);
+ ALOGI("open() usage = %d, contentType = %d, inputPreset = %d, allowedCapturePolicy = %d",
+ mUsage, mContentType, mInputPreset, mAllowedCapturePolicy);
return AAUDIO_OK;
}
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 46951f5..32713b1 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -219,6 +219,10 @@
return mInputPreset;
}
+ aaudio_allowed_capture_policy_t getAllowedCapturePolicy() const {
+ return mAllowedCapturePolicy;
+ }
+
int32_t getSessionId() const {
return mSessionId;
}
@@ -525,6 +529,13 @@
mInputPreset = inputPreset;
}
+ /**
+ * This should not be called after the open() call.
+ */
+ void setAllowedCapturePolicy(aaudio_allowed_capture_policy_t policy) {
+ mAllowedCapturePolicy = policy;
+ }
+
private:
aaudio_result_t safeStop();
@@ -546,6 +557,7 @@
aaudio_usage_t mUsage = AAUDIO_UNSPECIFIED;
aaudio_content_type_t mContentType = AAUDIO_UNSPECIFIED;
aaudio_input_preset_t mInputPreset = AAUDIO_UNSPECIFIED;
+ aaudio_allowed_capture_policy_t mAllowedCapturePolicy = AAUDIO_ALLOW_CAPTURE_BY_ALL;
int32_t mSessionId = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index ff95aed..d628bf7 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -134,12 +134,14 @@
AAudioConvert_contentTypeToInternal(builder.getContentType());
const audio_usage_t usage =
AAudioConvert_usageToInternal(builder.getUsage());
+ const audio_flags_mask_t attributesFlags =
+ AAudioConvert_allowCapturePolicyToAudioFlagsMask(builder.getAllowedCapturePolicy());
const audio_attributes_t attributes = {
.content_type = contentType,
.usage = usage,
.source = AUDIO_SOURCE_DEFAULT, // only used for recording
- .flags = AUDIO_FLAG_NONE, // Different than the AUDIO_OUTPUT_FLAGS
+ .flags = attributesFlags,
.tags = ""
};
@@ -420,6 +422,10 @@
aaudio_result_t AudioStreamTrack::setBufferSize(int32_t requestedFrames)
{
+ // Do not ask for less than one burst.
+ if (requestedFrames < getFramesPerBurst()) {
+ requestedFrames = getFramesPerBurst();
+ }
ssize_t result = mAudioTrack->setBufferSizeInFrames(requestedFrames);
if (result < 0) {
return AAudioConvert_androidToAAudioResult(result);
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 723cbf1..96ed56a 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -217,6 +217,22 @@
return (audio_source_t) preset; // same value
}
+audio_flags_mask_t AAudioConvert_allowCapturePolicyToAudioFlagsMask(
+ aaudio_allowed_capture_policy_t policy) {
+ switch (policy) {
+ case AAUDIO_UNSPECIFIED:
+ case AAUDIO_ALLOW_CAPTURE_BY_ALL:
+ return AUDIO_FLAG_NONE;
+ case AAUDIO_ALLOW_CAPTURE_BY_SYSTEM:
+ return AUDIO_FLAG_NO_MEDIA_PROJECTION;
+ case AAUDIO_ALLOW_CAPTURE_BY_NONE:
+ return AUDIO_FLAG_NO_MEDIA_PROJECTION | AUDIO_FLAG_NO_SYSTEM_CAPTURE;
+ default:
+ ALOGE("%s() 0x%08X unrecognized", __func__, policy);
+ return AUDIO_FLAG_NONE; //
+ }
+}
+
int32_t AAudioConvert_framesToBytes(int32_t numFrames,
int32_t bytesPerFrame,
int32_t *sizeInBytes) {
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index dc2b198..76d0457 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -84,6 +84,14 @@
*/
audio_source_t AAudioConvert_inputPresetToAudioSource(aaudio_input_preset_t preset);
+/**
+ * Note that this function does not validate the passed in value.
+ * That is done somewhere else.
+ * @return internal audio flags mask
+ */
+audio_flags_mask_t AAudioConvert_allowCapturePolicyToAudioFlagsMask(
+ aaudio_allowed_capture_policy_t policy);
+
// Note that this code may be replaced by Settings or by some other system configuration tool.
#define AAUDIO_PROP_MMAP_POLICY "aaudio.mmap_policy"
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index cb243a0..958bb2e 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -196,3 +196,10 @@
"libutils",
],
}
+
+cc_test {
+ name: "test_full_queue",
+ defaults: ["libaaudio_tests_defaults"],
+ srcs: ["test_full_queue.cpp"],
+ shared_libs: ["libaaudio"],
+}
diff --git a/media/libaaudio/tests/test_attributes.cpp b/media/libaaudio/tests/test_attributes.cpp
index dbf8712..32ee2a3 100644
--- a/media/libaaudio/tests/test_attributes.cpp
+++ b/media/libaaudio/tests/test_attributes.cpp
@@ -32,6 +32,7 @@
aaudio_usage_t usage,
aaudio_content_type_t contentType,
aaudio_input_preset_t preset = DONT_SET,
+ aaudio_allowed_capture_policy_t capturePolicy = DONT_SET,
aaudio_direction_t direction = AAUDIO_DIRECTION_OUTPUT) {
float *buffer = new float[kNumFrames * kChannelCount];
@@ -56,6 +57,9 @@
if (preset != DONT_SET) {
AAudioStreamBuilder_setInputPreset(aaudioBuilder, preset);
}
+ if (capturePolicy != DONT_SET) {
+ AAudioStreamBuilder_setAllowedCapturePolicy(aaudioBuilder, capturePolicy);
+ }
// Create an AAudioStream using the Builder.
ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
@@ -80,6 +84,12 @@
: preset;
EXPECT_EQ(expectedPreset, AAudioStream_getInputPreset(aaudioStream));
+ aaudio_allowed_capture_policy_t expectedCapturePolicy =
+ (capturePolicy == DONT_SET || capturePolicy == AAUDIO_UNSPECIFIED)
+ ? AAUDIO_ALLOW_CAPTURE_BY_ALL // default
+ : preset;
+ EXPECT_EQ(expectedCapturePolicy, AAudioStream_getAllowedCapturePolicy(aaudioStream));
+
EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream));
if (direction == AAUDIO_DIRECTION_INPUT) {
@@ -133,13 +143,21 @@
AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE,
};
+static const aaudio_input_preset_t sAllowCapturePolicies[] = {
+ DONT_SET,
+ AAUDIO_UNSPECIFIED,
+ AAUDIO_ALLOW_CAPTURE_BY_ALL,
+ AAUDIO_ALLOW_CAPTURE_BY_SYSTEM,
+ AAUDIO_ALLOW_CAPTURE_BY_NONE,
+};
+
static void checkAttributesUsage(aaudio_performance_mode_t perfMode) {
for (aaudio_usage_t usage : sUsages) {
checkAttributes(perfMode, usage, DONT_SET);
}
}
-static void checkAttributesContentType(aaudio_input_preset_t perfMode) {
+static void checkAttributesContentType(aaudio_performance_mode_t perfMode) {
for (aaudio_content_type_t contentType : sContentypes) {
checkAttributes(perfMode, DONT_SET, contentType);
}
@@ -151,6 +169,18 @@
DONT_SET,
DONT_SET,
inputPreset,
+ DONT_SET,
+ AAUDIO_DIRECTION_INPUT);
+ }
+}
+
+static void checkAttributesAllowedCapturePolicy(aaudio_performance_mode_t perfMode) {
+ for (aaudio_allowed_capture_policy_t policy : sAllowCapturePolicies) {
+ checkAttributes(perfMode,
+ DONT_SET,
+ DONT_SET,
+ DONT_SET,
+ policy,
AAUDIO_DIRECTION_INPUT);
}
}
@@ -167,6 +197,10 @@
checkAttributesInputPreset(AAUDIO_PERFORMANCE_MODE_NONE);
}
+TEST(test_attributes, aaudio_allowed_capture_policy_perfnone) {
+ checkAttributesAllowedCapturePolicy(AAUDIO_PERFORMANCE_MODE_NONE);
+}
+
TEST(test_attributes, aaudio_usage_lowlat) {
checkAttributesUsage(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
}
@@ -178,3 +212,7 @@
TEST(test_attributes, aaudio_input_preset_lowlat) {
checkAttributesInputPreset(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
}
+
+TEST(test_attributes, aaudio_allowed_capture_policy_lowlat) {
+ checkAttributesAllowedCapturePolicy(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
+}
diff --git a/media/libaaudio/tests/test_full_queue.cpp b/media/libaaudio/tests/test_full_queue.cpp
new file mode 100644
index 0000000..12d4fa3
--- /dev/null
+++ b/media/libaaudio/tests/test_full_queue.cpp
@@ -0,0 +1,93 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Test whether a stream dies if it is written to after a delay.
+// Maybe because the message queue from the AAudio service fills up.
+
+#include <stdio.h>
+#include <unistd.h>
+
+#include <aaudio/AAudio.h>
+#include <gtest/gtest.h>
+
+constexpr int64_t kNanosPerSecond = 1000000000;
+constexpr int64_t kTimeoutNanos = kNanosPerSecond / 2;
+constexpr int kNumFrames = 256;
+constexpr int kChannelCount = 2;
+
+static void checkFullQueue(aaudio_performance_mode_t perfMode,
+ int32_t sleepMillis) {
+ std::unique_ptr<float[]> buffer = std::make_unique<float[]>(
+ kNumFrames * kChannelCount);
+
+ AAudioStreamBuilder *aaudioBuilder = nullptr;
+
+ // Use an AAudioStreamBuilder to contain requested parameters.
+ ASSERT_EQ(AAUDIO_OK, AAudio_createStreamBuilder(&aaudioBuilder));
+
+ AAudioStreamBuilder_setChannelCount(aaudioBuilder, kChannelCount);
+
+ // Request stream properties.
+ AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+
+ // Create an AAudioStream using the Builder.
+ AAudioStream *aaudioStream = nullptr;
+ ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder,
+ &aaudioStream));
+ AAudioStreamBuilder_delete(aaudioBuilder);
+
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStart(aaudioStream));
+
+ // Sleep for awhile. This might kill the stream.
+ usleep(sleepMillis * 1000); // 1000 millis in a microsecond
+
+ for (int i = 0; i < 10; i++) {
+ const aaudio_result_t result = AAudioStream_write(aaudioStream,
+ buffer.get(),
+ kNumFrames,
+ kTimeoutNanos);
+ EXPECT_EQ(kNumFrames, result);
+ if (kNumFrames != result) break;
+ }
+
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
+
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
+}
+
+TEST(test_full_queue, aaudio_full_queue_perf_none_50) {
+ checkFullQueue(AAUDIO_PERFORMANCE_MODE_NONE, 50 /* sleepMillis */);
+}
+
+TEST(test_full_queue, aaudio_full_queue_perf_none_200) {
+ checkFullQueue(AAUDIO_PERFORMANCE_MODE_NONE, 200 /* sleepMillis */);
+}
+
+TEST(test_full_queue, aaudio_full_queue_perf_none_1000) {
+ checkFullQueue(AAUDIO_PERFORMANCE_MODE_NONE, 1000 /* sleepMillis */);
+}
+
+TEST(test_full_queue, aaudio_full_queue_low_latency_50) {
+ checkFullQueue(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, 50 /* sleepMillis */);
+}
+
+TEST(test_full_queue, aaudio_full_queue_low_latency_200) {
+ checkFullQueue(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, 200 /* sleepMillis */);
+}
+
+TEST(test_full_queue, aaudio_full_queue_low_latency_1000) {
+ checkFullQueue(AAUDIO_PERFORMANCE_MODE_LOW_LATENCY, 1000 /* sleepMillis */);
+}
diff --git a/media/libaudioclient/AudioPolicy.cpp b/media/libaudioclient/AudioPolicy.cpp
index 3ab38cd..65e797f 100644
--- a/media/libaudioclient/AudioPolicy.cpp
+++ b/media/libaudioclient/AudioPolicy.cpp
@@ -97,6 +97,7 @@
mDeviceType = (audio_devices_t) parcel->readInt32();
mDeviceAddress = parcel->readString8();
mCbFlags = (uint32_t)parcel->readInt32();
+ mAllowPrivilegedPlaybackCapture = parcel->readBool();
size_t size = (size_t)parcel->readInt32();
if (size > MAX_CRITERIA_PER_MIX) {
size = MAX_CRITERIA_PER_MIX;
@@ -120,6 +121,7 @@
parcel->writeInt32(mDeviceType);
parcel->writeString8(mDeviceAddress);
parcel->writeInt32(mCbFlags);
+ parcel->writeBool(mAllowPrivilegedPlaybackCapture);
size_t size = mCriteria.size();
if (size > MAX_CRITERIA_PER_MIX) {
size = MAX_CRITERIA_PER_MIX;
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 0ce2513..f324669 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -1133,6 +1133,12 @@
}
}
+status_t AudioSystem::setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t flags) {
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == nullptr) return PERMISSION_DENIED;
+ return aps->setAllowedCapturePolicy(uid, flags);
+}
+
bool AudioSystem::isOffloadSupported(const audio_offload_info_t& info)
{
ALOGV("isOffloadSupported()");
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 3bac44f..bf98c60 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -101,7 +101,8 @@
LIST_AUDIO_PRODUCT_STRATEGIES,
GET_STRATEGY_FOR_ATTRIBUTES,
LIST_AUDIO_VOLUME_GROUPS,
- GET_VOLUME_GROUP_FOR_ATTRIBUTES
+ GET_VOLUME_GROUP_FOR_ATTRIBUTES,
+ SET_ALLOWED_CAPTURE_POLICY,
};
#define MAX_ITEMS_PER_LIST 1024
@@ -603,6 +604,15 @@
return status;
}
+ status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t flags) override {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ data.writeInt32(uid);
+ data.writeInt32(flags);
+ remote()->transact(SET_ALLOWED_CAPTURE_POLICY, data, &reply);
+ return reply.readInt32();
+ }
+
virtual bool isOffloadSupported(const audio_offload_info_t& info)
{
Parcel data, reply;
@@ -2168,7 +2178,7 @@
CHECK_INTERFACE(IAudioPolicyService, data, reply);
bool isSupported = isHapticPlaybackSupported();
reply->writeBool(isSupported);
- return NO_ERROR;
+ return NO_ERROR;
}
case SET_UID_DEVICE_AFFINITY: {
@@ -2285,6 +2295,15 @@
return NO_ERROR;
}
+ case SET_ALLOWED_CAPTURE_POLICY: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ uid_t uid = data.readInt32();
+ audio_flags_mask_t flags = data.readInt32();
+ status_t status = setAllowedCapturePolicy(uid, flags);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libaudioclient/include/media/AudioPolicy.h b/media/libaudioclient/include/media/AudioPolicy.h
index bf8d627..4b94c12 100644
--- a/media/libaudioclient/include/media/AudioPolicy.h
+++ b/media/libaudioclient/include/media/AudioPolicy.h
@@ -114,6 +114,8 @@
audio_devices_t mDeviceType;
String8 mDeviceAddress;
uint32_t mCbFlags; // flags indicating which callbacks to use, see kCbFlag*
+ /** Ignore the AUDIO_FLAG_NO_MEDIA_PROJECTION */
+ bool mAllowPrivilegedPlaybackCapture = false;
};
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index e64f285..05a1d56 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -299,6 +299,8 @@
static status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory);
+ static status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t flags);
+
// Check if hw offload is possible for given format, stream type, sample rate,
// bit rate, duration, video and streaming or offload property is enabled
static bool isOffloadSupported(const audio_offload_info_t& info);
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index 35540f0..95530ac 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -135,6 +135,7 @@
audio_unique_id_t* id) = 0;
virtual status_t removeSourceDefaultEffect(audio_unique_id_t id) = 0;
virtual status_t removeStreamDefaultEffect(audio_unique_id_t id) = 0;
+ virtual status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t flags) = 0;
// Check if offload is possible for given format, stream type, sample rate,
// bit rate, duration, video and streaming or offload property is enabled
virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 6c8e6a4..c08dddb 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -49,7 +49,7 @@
// Instrument audio signal power logging.
// Note: This assumes channel mask, format, and sample rate do not change after creation.
- if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
+ if (mStream != nullptr /* && mStreamPowerLog.isUserDebugOrEngBuild() */) {
// Obtain audio properties (see StreamHalHidl::getAudioProperties() below).
Return<void> ret = mStream->getAudioProperties(
[&](auto sr, auto m, auto f) {
diff --git a/media/libaudiohal/impl/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
index 7d5ce05..4818fd8 100644
--- a/media/libaudiohal/impl/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -33,7 +33,7 @@
mStream(stream) {
// Instrument audio signal power logging.
// Note: This assumes channel mask, format, and sample rate do not change after creation.
- if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
+ if (mStream != nullptr /* && mStreamPowerLog.isUserDebugOrEngBuild() */) {
mStreamPowerLog.init(mStream->get_sample_rate(mStream),
mStream->get_channels(mStream),
mStream->get_format(mStream));
diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index eeeecce..52936cc 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp
@@ -243,7 +243,7 @@
{
// compute the normalized transition bandwidth
const double tbw = firKaiserTbw(c.mHalfNumCoefs, stopBandAtten);
- const double halfbw = tbw / 2.;
+ const double halfbw = tbw * 0.5;
double fcr; // compute fcr, the 3 dB amplitude cut-off.
if (inSampleRate < outSampleRate) { // upsample
@@ -290,7 +290,7 @@
#if 0
// Keep this debug code in case an app causes resampler design issues.
- const double halfbw = tbw / 2.;
+ const double halfbw = tbw * 0.5;
// print basic filter stats
ALOGD("L:%d hnc:%d stopBandAtten:%lf fcr:%lf atten:%lf tbw:%lf\n",
c.mL, c.mHalfNumCoefs, stopBandAtten, fcr, attenuation, tbw);
@@ -305,7 +305,7 @@
const int32_t passSteps = 1000;
- testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.ML /*stopSteps*/,
+ testFir(coefs, c.mL, c.mHalfNumCoefs, fp, fs, passSteps, passSteps * c.mL /*stopSteps*/,
passMin, passMax, passRipple, stopMax, stopRipple);
ALOGD("passband(%lf, %lf): %.8lf %.8lf %.8lf\n", 0., fp, passMin, passMax, passRipple);
ALOGD("stopband(%lf, %lf): %.8lf %.3lf\n", fs, 0.5, stopMax, stopRipple);
diff --git a/media/libaudioprocessing/AudioResamplerFirGen.h b/media/libaudioprocessing/AudioResamplerFirGen.h
index 39cafeb..100baa3 100644
--- a/media/libaudioprocessing/AudioResamplerFirGen.h
+++ b/media/libaudioprocessing/AudioResamplerFirGen.h
@@ -547,8 +547,8 @@
wstart += wstep;
}
// renormalize - this is needed for integer filter types, use 1 for float or double.
- constexpr int64_t integralShift = std::is_integral<T>::value ? (sizeof(T) * 8 - 1) : 0;
- const double norm = 1. / (L << integralShift);
+ constexpr int integralShift = std::is_integral<T>::value ? (sizeof(T) * CHAR_BIT - 1) : 0;
+ const double norm = 1. / (int64_t{L} << integralShift);
firMin = fmin * norm;
firMax = fmax * norm;
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index 0c8e5bb..d990111 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -5,8 +5,8 @@
header_libs: ["libbase_headers"],
shared_libs: [
- "libaudioutils",
"libaudioprocessing",
+ "libaudioutils",
"libcutils",
"liblog",
"libutils",
diff --git a/media/libaudioprocessing/tests/resampler_tests.cpp b/media/libaudioprocessing/tests/resampler_tests.cpp
index e1623f7..8292291 100644
--- a/media/libaudioprocessing/tests/resampler_tests.cpp
+++ b/media/libaudioprocessing/tests/resampler_tests.cpp
@@ -246,7 +246,8 @@
}
void testFilterResponse(
- size_t channels, unsigned inputFreq, unsigned outputFreq)
+ size_t channels, unsigned inputFreq, unsigned outputFreq,
+ android::AudioResampler::src_quality quality = android::AudioResampler::DYN_HIGH_QUALITY)
{
// create resampler
using ResamplerType = android::AudioResamplerDyn<float, float, float>;
@@ -256,7 +257,7 @@
AUDIO_FORMAT_PCM_FLOAT,
channels,
outputFreq,
- android::AudioResampler::DYN_HIGH_QUALITY)));
+ quality)));
rdyn->setSampleRate(inputFreq);
// get design parameters
@@ -268,17 +269,20 @@
const double attenuation = rdyn->getFilterAttenuation();
const double stopbandDb = rdyn->getStopbandAttenuationDb();
const double passbandDb = rdyn->getPassbandRippleDb();
- const double fp = fcr - tbw / 2;
- const double fs = fcr + tbw / 2;
+ const double fp = fcr - tbw * 0.5;
+ const double fs = fcr + tbw * 0.5;
+ const double idealfs = inputFreq <= outputFreq
+ ? 0.5 // upsample
+ : 0.5 * outputFreq / inputFreq; // downsample
- printf("inputFreq:%d outputFreq:%d design"
+ printf("inputFreq:%d outputFreq:%d design quality %d"
" phases:%d halfLength:%d"
- " fcr:%lf fp:%lf fs:%lf tbw:%lf"
+ " fcr:%lf fp:%lf fs:%lf tbw:%lf fcrp:%lf"
" attenuation:%lf stopRipple:%.lf passRipple:%lf"
"\n",
- inputFreq, outputFreq,
+ inputFreq, outputFreq, quality,
phases, halfLength,
- fcr, fp, fs, tbw,
+ fcr, fp, fs, tbw, fcr * 100. / idealfs,
attenuation, stopbandDb, passbandDb);
// verify design parameters
@@ -541,8 +545,36 @@
}
}
-TEST(audioflinger_resampler, filterresponse) {
- std::vector<int> inSampleRates{
+// Selected downsampling responses for various frequencies relating to hearing aid.
+TEST(audioflinger_resampler, downsamplingresponse) {
+ static constexpr android::AudioResampler::src_quality qualities[] = {
+ android::AudioResampler::DYN_LOW_QUALITY,
+ android::AudioResampler::DYN_MED_QUALITY,
+ android::AudioResampler::DYN_HIGH_QUALITY,
+ };
+ static constexpr int inSampleRates[] = {
+ 32000,
+ 44100,
+ 48000,
+ };
+ static constexpr int outSampleRates[] = {
+ 16000,
+ 24000,
+ };
+
+ for (auto quality : qualities) {
+ for (int outSampleRate : outSampleRates) {
+ for (int inSampleRate : inSampleRates) {
+ testFilterResponse(2 /* channels */, inSampleRate, outSampleRate, quality);
+ }
+ }
+ }
+}
+
+// General responses for typical output device scenarios - 44.1, 48, 96 kHz
+// (48, 96 are part of the same resampler generation family).
+TEST(audioflinger_resampler, generalresponse) {
+ static constexpr int inSampleRates[] = {
8000,
11025,
12000,
@@ -557,7 +589,8 @@
176400,
192000,
};
- std::vector<int> outSampleRates{
+ static constexpr int outSampleRates[] = {
+ 44100,
48000,
96000,
};
diff --git a/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp b/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp
index 0b883f1..c03c6ed 100644
--- a/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp
+++ b/media/libeffects/dynamicsproc/EffectDynamicsProcessing.cpp
@@ -25,6 +25,7 @@
#include <new>
#include <log/log.h>
+#include <sys/param.h>
#include <audio_effects/effect_dynamicsprocessing.h>
#include <dsp/DPBase.h>
@@ -225,10 +226,6 @@
} //switch
}
-static inline bool isPowerOf2(unsigned long n) {
- return (n & (n - 1)) == 0;
-}
-
void DP_configureVariant(DynamicsProcessingContext *pContext, int newVariant) {
ALOGV("DP_configureVariant %d", newVariant);
switch(newVariant) {
@@ -242,7 +239,7 @@
desiredBlock);
if (desiredBlock < minBlockSize) {
currentBlock = minBlockSize;
- } else if (!isPowerOf2(desiredBlock)) {
+ } else if (!powerof2(desiredBlock)) {
//find next highest power of 2.
currentBlock = 1 << (32 - __builtin_clz(desiredBlock));
}
@@ -1297,4 +1294,3 @@
};
}; // extern "C"
-
diff --git a/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp b/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp
index d06fd70..1f53978 100644
--- a/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp
+++ b/media/libeffects/dynamicsproc/dsp/DPFrequency.cpp
@@ -20,6 +20,7 @@
#include <log/log.h>
#include "DPFrequency.h"
#include <algorithm>
+#include <sys/param.h>
namespace dp_fx {
@@ -30,10 +31,6 @@
#define CIRCULAR_BUFFER_UPSAMPLE 4 //4 times buffer size
static constexpr float MIN_ENVELOPE = 1e-6f; //-120 dB
-//helper functionS
-static inline bool isPowerOf2(unsigned long n) {
- return (n & (n - 1)) == 0;
-}
static constexpr float EPSILON = 0.0000001f;
static inline bool isZero(float f) {
@@ -151,7 +148,7 @@
} else if (mBlockSize < MIN_BLOCKSIZE) {
mBlockSize = MIN_BLOCKSIZE;
} else {
- if (!isPowerOf2(blockSize)) {
+ if (!powerof2(blockSize)) {
//find next highest power of 2.
mBlockSize = 1 << (32 - __builtin_clz(blockSize));
}
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
index 1d95342..1b27cb4 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
@@ -793,6 +793,15 @@
{
EQNB_Params.SourceFormat = LVEQNB_STEREO;
}
+#ifdef SUPPORT_MC
+ /* Note: Currently SourceFormat field of EQNB is not been
+ * used by the module.
+ */
+ else if (LocalParams.SourceFormat == LVM_MULTICHANNEL)
+ {
+ EQNB_Params.SourceFormat = LVEQNB_MULTICHANNEL;
+ }
+#endif
else
{
EQNB_Params.SourceFormat = LVEQNB_MONOINSTEREO; /* Force to Mono-in-Stereo mode */
@@ -862,7 +871,16 @@
CS_Params.SpeakerType = LVCS_HEADPHONES;
}
+#ifdef SUPPORT_MC
+ /* Concert sound module processes only the left and right channels
+ * data. So the Source Format is set to LVCS_STEREO for multichannel
+ * input also.
+ */
+ if (LocalParams.SourceFormat == LVM_STEREO ||
+ LocalParams.SourceFormat == LVM_MULTICHANNEL)
+#else
if (LocalParams.SourceFormat == LVM_STEREO) /* Mono format not supported */
+#endif
{
CS_Params.SourceFormat = LVCS_STEREO;
}
diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
index 59586e0..fbfdd4d 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
@@ -94,6 +94,7 @@
typedef int32_t LVM_INT32; /* Signed 32-bit word */
typedef uint32_t LVM_UINT32; /* Unsigned 32-bit word */
+typedef int64_t LVM_INT64; /* Signed 64-bit word */
#ifdef BUILD_FLOAT
diff --git a/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h b/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
index 385dbcf..804f1bf 100644
--- a/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
+++ b/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
@@ -184,6 +184,9 @@
{
LVEQNB_STEREO = 0,
LVEQNB_MONOINSTEREO = 1,
+#ifdef SUPPORT_MC
+ LVEQNB_MULTICHANNEL = 2,
+#endif
LVEQNB_SOURCE_MAX = LVM_MAXINT_32
} LVEQNB_SourceFormat_en;
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c
index ea5f74a..61899fe 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c
@@ -22,6 +22,17 @@
#define LVM_MININT_32 0x80000000
+static LVM_INT32 mult32x32in32_shiftr(LVM_INT32 a, LVM_INT32 b, LVM_INT32 c) {
+ LVM_INT64 result = ((LVM_INT64)a * b) >> c;
+
+ if (result >= INT32_MAX) {
+ return INT32_MAX;
+ } else if (result <= INT32_MIN) {
+ return INT32_MIN;
+ } else {
+ return (LVM_INT32)result;
+ }
+}
/************************************************************************************/
/* */
@@ -123,10 +134,10 @@
if(pLVPSA_Inst->pSpectralDataBufferWritePointer != pWrite_Save)
{
- MUL32x32INTO32((AudioTime + (LVM_INT32)((LVM_INT32)pLVPSA_Inst->LocalSamplesCount*1000)),
- (LVM_INT32)LVPSA_SampleRateInvTab[pLVPSA_Inst->CurrentParams.Fs],
- AudioTimeInc,
- LVPSA_FsInvertShift)
+ AudioTimeInc = mult32x32in32_shiftr(
+ (AudioTime + ((LVM_INT32)pLVPSA_Inst->LocalSamplesCount * 1000)),
+ (LVM_INT32)LVPSA_SampleRateInvTab[pLVPSA_Inst->CurrentParams.Fs],
+ LVPSA_FsInvertShift);
pLVPSA_Inst->SpectralDataBufferAudioTime = AudioTime + AudioTimeInc;
}
diff --git a/media/libmedia/IMediaSource.cpp b/media/libmedia/IMediaSource.cpp
index 4dece96..50826c5 100644
--- a/media/libmedia/IMediaSource.cpp
+++ b/media/libmedia/IMediaSource.cpp
@@ -368,13 +368,13 @@
ALOGV("Use shared memory: %zu", length);
transferBuf = buf;
} else {
- ALOGD("Large buffer %zu without IMemory!", length);
+ ALOGV("Large buffer %zu without IMemory!", length);
ret = mGroup->acquire_buffer(
(MediaBufferBase **)&transferBuf, false /* nonBlocking */, length);
if (ret != OK
|| transferBuf == nullptr
|| transferBuf->mMemory == nullptr) {
- ALOGW("Failed to acquire shared memory, size %zu, ret %d",
+ ALOGV("Failed to acquire shared memory, size %zu, ret %d",
length, ret);
if (transferBuf != nullptr) {
transferBuf->release();
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmedia/TypeConverter.cpp
index 469c5b6..5be78d1 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmedia/TypeConverter.cpp
@@ -392,7 +392,8 @@
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_BYPASS_MUTE),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_LOW_LATENCY),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_DEEP_BUFFER),
- MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NO_CAPTURE),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NO_MEDIA_PROJECTION),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NO_SYSTEM_CAPTURE),
TERMINATOR
};
diff --git a/media/libmedia/include/media/IMediaSource.h b/media/libmedia/include/media/IMediaSource.h
index 5ab6e37..381df24 100644
--- a/media/libmedia/include/media/IMediaSource.h
+++ b/media/libmedia/include/media/IMediaSource.h
@@ -124,7 +124,8 @@
return false;
}
- static const size_t kBinderMediaBuffers = 4; // buffers managed by BnMediaSource
+ // align buffer count with video request size in NuMediaExtractor::selectTrack()
+ static const size_t kBinderMediaBuffers = 8; // buffers managed by BnMediaSource
static const size_t kTransferSharedAsSharedThreshold = 4 * 1024; // if >= shared, else inline
static const size_t kTransferInlineAsSharedThreshold = 8 * 1024; // if >= shared, else inline
static const size_t kInlineMaxTransfer = 64 * 1024; // Binder size limited to BINDER_VM_SIZE.
diff --git a/media/libmedia/xsd/vts/Android.bp b/media/libmedia/xsd/vts/Android.bp
new file mode 100644
index 0000000..b590a12
--- /dev/null
+++ b/media/libmedia/xsd/vts/Android.bp
@@ -0,0 +1,33 @@
+//
+// Copyright (C) 2019 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_test {
+ name: "vts_mediaProfiles_validate_test",
+ srcs: [
+ "ValidateMediaProfiles.cpp"
+ ],
+ static_libs: [
+ "android.hardware.audio.common.test.utility",
+ "libxml2",
+ ],
+ shared_libs: [
+ "liblog",
+ ],
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+}
diff --git a/media/libmedia/xsd/vts/Android.mk b/media/libmedia/xsd/vts/Android.mk
new file mode 100644
index 0000000..52c3779
--- /dev/null
+++ b/media/libmedia/xsd/vts/Android.mk
@@ -0,0 +1,22 @@
+#
+# Copyright (C) 2019 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+#
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := VtsValidateMediaProfiles
+include test/vts/tools/build/Android.host_config.mk
diff --git a/media/libmedia/xsd/vts/AndroidTest.xml b/media/libmedia/xsd/vts/AndroidTest.xml
new file mode 100644
index 0000000..e68721b
--- /dev/null
+++ b/media/libmedia/xsd/vts/AndroidTest.xml
@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Config for VTS VtsValidateMediaProfiles.">
+ <option name="config-descriptor:metadata" key="plan" value="vts-treble" />
+ <target_preparer class="com.android.compatibility.common.tradefed.targetprep.VtsFilePusher">
+ <option name="abort-on-push-failure" value="false"/>
+ <option name="push-group" value="HostDrivenTest.push"/>
+ <option name="push" value="DATA/etc/media_profiles.xsd->/data/local/tmp/media_profiles.xsd"/>
+ </target_preparer>
+ <test class="com.android.tradefed.testtype.VtsMultiDeviceTest">
+ <option name="test-module-name" value="VtsValidateMediaProfiles"/>
+ <option name="binary-test-source" value="_32bit::DATA/nativetest/vts_mediaProfiles_validate_test/vts_mediaProfiles_validate_test" />
+ <option name="binary-test-source" value="_64bit::DATA/nativetest64/vts_mediaProfiles_validate_test/vts_mediaProfiles_validate_test" />
+ <option name="binary-test-type" value="gtest"/>
+ <option name="test-timeout" value="30s"/>
+ </test>
+</configuration>
diff --git a/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp b/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp
new file mode 100644
index 0000000..ff9b060
--- /dev/null
+++ b/media/libmedia/xsd/vts/ValidateMediaProfiles.cpp
@@ -0,0 +1,28 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "utility/ValidateXml.h"
+
+TEST(CheckConfig, mediaProfilesValidation) {
+ RecordProperty("description",
+ "Verify that the media profiles file "
+ "is valid according to the schema");
+
+ const char* location = "/vendor/etc";
+
+ EXPECT_ONE_VALID_XML_MULTIPLE_LOCATIONS("media_profiles_V1_0.xml", {location},
+ "/data/local/tmp/media_profiles.xsd");
+}
diff --git a/media/libmediaplayerservice/Android.bp b/media/libmediaplayerservice/Android.bp
index 46a1c24..0776172 100644
--- a/media/libmediaplayerservice/Android.bp
+++ b/media/libmediaplayerservice/Android.bp
@@ -11,6 +11,7 @@
],
shared_libs: [
+ "android.hardware.media.c2@1.0",
"android.hardware.media.omx@1.0",
"libbase",
"libaudioclient",
@@ -21,7 +22,6 @@
"libdl",
"libgui",
"libhidlbase",
- "libhidlmemory",
"liblog",
"libmedia",
"libmedia_omx",
@@ -70,5 +70,3 @@
}
-subdirs = ["*"]
-
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index da95817..5061024 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -34,6 +34,8 @@
#include <utils/misc.h>
+#include <android/hardware/media/omx/1.0/IOmxStore.h>
+#include <android/hardware/media/c2/1.0/IComponentStore.h>
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
#include <binder/MemoryHeapBase.h>
@@ -650,17 +652,17 @@
const sp<MediaPlayerBase>& listener,
int which) {
mService = service;
- mOmx = nullptr;
+ mHService = nullptr;
mListener = listener;
mWhich = which;
}
MediaPlayerService::Client::ServiceDeathNotifier::ServiceDeathNotifier(
- const sp<IOmx>& omx,
+ const sp<android::hidl::base::V1_0::IBase>& hService,
const sp<MediaPlayerBase>& listener,
int which) {
mService = nullptr;
- mOmx = omx;
+ mHService = hService;
mListener = listener;
mWhich = which;
}
@@ -692,9 +694,9 @@
if (mService != nullptr) {
mService->unlinkToDeath(this);
mService = nullptr;
- } else if (mOmx != nullptr) {
- mOmx->unlinkToDeath(this);
- mOmx = nullptr;
+ } else if (mHService != nullptr) {
+ mHService->unlinkToDeath(this);
+ mHService = nullptr;
}
}
@@ -714,10 +716,12 @@
mExtractorDeathListener->unlinkToDeath();
mExtractorDeathListener = nullptr;
}
- if (mCodecDeathListener != nullptr) {
- mCodecDeathListener->unlinkToDeath();
- mCodecDeathListener = nullptr;
+ for (const sp<ServiceDeathNotifier>& codecDeathListener : mCodecDeathListeners) {
+ if (codecDeathListener != nullptr) {
+ codecDeathListener->unlinkToDeath();
+ }
}
+ mCodecDeathListeners.clear();
}
sp<MediaPlayerBase> MediaPlayerService::Client::setDataSource_pre(
@@ -741,20 +745,56 @@
new ServiceDeathNotifier(binder, p, MEDIAEXTRACTOR_PROCESS_DEATH);
binder->linkToDeath(extractorDeathListener);
- sp<IOmx> omx = IOmx::getService();
- if (omx == nullptr) {
- ALOGE("IOmx service is not available");
- return NULL;
+ std::vector<sp<ServiceDeathNotifier>> codecDeathListeners;
+ {
+ using ::android::hidl::base::V1_0::IBase;
+
+ // Listen to OMX's IOmxStore/default
+ {
+ sp<IBase> store = ::android::hardware::media::omx::V1_0::
+ IOmxStore::getService();
+ if (store == nullptr) {
+ ALOGD("OMX service is not available");
+ } else {
+ sp<ServiceDeathNotifier> codecDeathListener =
+ new ServiceDeathNotifier(store, p, MEDIACODEC_PROCESS_DEATH);
+ store->linkToDeath(codecDeathListener, 0);
+ codecDeathListeners.emplace_back(codecDeathListener);
+ }
+ }
+
+ // Listen to Codec2's IComponentStore/software
+ // TODO: Listen to all Codec2 services.
+ {
+ sp<IBase> store = ::android::hardware::media::c2::V1_0::
+ IComponentStore::getService();
+ if (store == nullptr) {
+ ALOGD("Codec2 system service is not available");
+ } else {
+ sp<ServiceDeathNotifier> codecDeathListener =
+ new ServiceDeathNotifier(store, p, MEDIACODEC_PROCESS_DEATH);
+ store->linkToDeath(codecDeathListener, 0);
+ codecDeathListeners.emplace_back(codecDeathListener);
+ }
+
+ store = ::android::hardware::media::c2::V1_0::
+ IComponentStore::getService("software");
+ if (store == nullptr) {
+ ALOGD("Codec2 swcodec service is not available");
+ } else {
+ sp<ServiceDeathNotifier> codecDeathListener =
+ new ServiceDeathNotifier(store, p, MEDIACODEC_PROCESS_DEATH);
+ store->linkToDeath(codecDeathListener, 0);
+ codecDeathListeners.emplace_back(codecDeathListener);
+ }
+ }
}
- sp<ServiceDeathNotifier> codecDeathListener =
- new ServiceDeathNotifier(omx, p, MEDIACODEC_PROCESS_DEATH);
- omx->linkToDeath(codecDeathListener, 0);
Mutex::Autolock lock(mLock);
clearDeathNotifiers_l();
mExtractorDeathListener = extractorDeathListener;
- mCodecDeathListener = codecDeathListener;
+ mCodecDeathListeners.swap(codecDeathListeners);
mAudioDeviceUpdatedListener = new AudioDeviceUpdatedNotifier(p);
if (!p->hardwareOutput()) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index bfb7cc2..26bfa71 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -30,7 +30,7 @@
#include <media/Metadata.h>
#include <media/stagefright/foundation/ABase.h>
-#include <android/hardware/media/omx/1.0/IOmx.h>
+#include <hidl/HidlSupport.h>
#include <system/audio.h>
@@ -42,7 +42,6 @@
class IDataSource;
class IMediaRecorder;
class IMediaMetadataRetriever;
-class IOMX;
class IRemoteDisplay;
class IRemoteDisplayClient;
class MediaRecorderClient;
@@ -70,7 +69,6 @@
class MediaPlayerService : public BnMediaPlayerService
{
class Client;
- typedef ::android::hardware::media::omx::V1_0::IOmx IOmx;
class AudioOutput : public MediaPlayerBase::AudioSink
{
@@ -400,7 +398,7 @@
const sp<MediaPlayerBase>& listener,
int which);
ServiceDeathNotifier(
- const sp<IOmx>& omx,
+ const sp<android::hidl::base::V1_0::IBase>& hService,
const sp<MediaPlayerBase>& listener,
int which);
virtual ~ServiceDeathNotifier();
@@ -413,7 +411,7 @@
private:
int mWhich;
sp<IBinder> mService;
- sp<IOmx> mOmx;
+ sp<android::hidl::base::V1_0::IBase> mHService; // HIDL service
wp<MediaPlayerBase> mListener;
};
@@ -509,7 +507,7 @@
media::Metadata::Filter mMetadataUpdated; // protected by mLock
sp<ServiceDeathNotifier> mExtractorDeathListener;
- sp<ServiceDeathNotifier> mCodecDeathListener;
+ std::vector<sp<ServiceDeathNotifier>> mCodecDeathListeners;
sp<AudioDeviceUpdatedNotifier> mAudioDeviceUpdatedListener;
#if CALLBACK_ANTAGONIZER
Antagonizer* mAntagonizer;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
index 0156ad2..a2cc13e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerCCDecoder.cpp
@@ -328,6 +328,11 @@
const size_t *userData = (size_t *)mpegUserData->data();
for (size_t i = 0; i < mpegUserData->size() / sizeof(size_t); ++i) {
+ if (accessUnit->size() < userData[i]) {
+ ALOGW("b/129068792, skip invalid offset for user data");
+ android_errorWriteLog(0x534e4554, "129068792");
+ continue;
+ }
trackAdded |= parseMPEGUserDataUnit(
timeUs, accessUnit->data() + userData[i], accessUnit->size() - userData[i]);
}
@@ -337,6 +342,12 @@
// returns true if a new CC track is found
bool NuPlayer::CCDecoder::parseMPEGUserDataUnit(int64_t timeUs, const uint8_t *data, size_t size) {
+ if (size < 9) {
+ ALOGW("b/129068792, MPEG user data size too small %zu", size);
+ android_errorWriteLog(0x534e4554, "129068792");
+ return false;
+ }
+
ABitReader br(data + 4, 5);
uint32_t user_identifier = br.getBits(32);
@@ -389,8 +400,14 @@
mDTVCCPacket->setRange(0, mDTVCCPacket->size() + 2);
br.skipBits(16);
} else if (mDTVCCPacket->size() > 0 && cc_type == 2) {
- memcpy(mDTVCCPacket->data() + mDTVCCPacket->size(), br.data(), 2);
- mDTVCCPacket->setRange(0, mDTVCCPacket->size() + 2);
+ if (mDTVCCPacket->capacity() - mDTVCCPacket->size() >= 2) {
+ memcpy(mDTVCCPacket->data() + mDTVCCPacket->size(), br.data(), 2);
+ mDTVCCPacket->setRange(0, mDTVCCPacket->size() + 2);
+ } else {
+ ALOGW("b/129068792, skip CC due to too much data(%zu, %zu)",
+ mDTVCCPacket->capacity(), mDTVCCPacket->size());
+ android_errorWriteLog(0x534e4554, "129068792");
+ }
br.skipBits(16);
} else if (cc_type == 0 || cc_type == 1) {
uint8_t cc_data_1 = br.getBits(8) & 0x7f;
@@ -477,6 +494,11 @@
size_t trackIndex = getTrackIndex(kTrackTypeCEA708, service_number, &trackAdded);
if (mSelectedTrack == (ssize_t)trackIndex) {
sp<ABuffer> ccPacket = new ABuffer(block_size);
+ if (ccPacket->capacity() == 0) {
+ ALOGW("b/129068792, no memory available, %zu", block_size);
+ android_errorWriteLog(0x534e4554, "129068792");
+ return false;
+ }
memcpy(ccPacket->data(), br.data(), block_size);
mCCMap.add(timeUs, ccPacket);
}
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index f00c895..cf1a6f1 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -569,7 +569,6 @@
mFps(-1.0),
mCaptureFps(-1.0),
mCreateInputBuffersSuspended(false),
- mLatency(0),
mTunneled(false),
mDescribeColorAspectsIndex((OMX_INDEXTYPE)0),
mDescribeHDRStaticInfoIndex((OMX_INDEXTYPE)0),
@@ -4425,12 +4424,13 @@
h264type.eProfile == OMX_VIDEO_AVCProfileHigh) {
h264type.nSliceHeaderSpacing = 0;
h264type.bUseHadamard = OMX_TRUE;
- h264type.nRefFrames = 2;
- h264type.nBFrames = mLatency == 0 ? 1 : std::min(1U, mLatency - 1);
-
- // disable B-frames until we have explicit settings for enabling the feature.
- h264type.nRefFrames = 1;
- h264type.nBFrames = 0;
+ int32_t maxBframes = 0;
+ (void)msg->findInt32(KEY_MAX_B_FRAMES, &maxBframes);
+ h264type.nBFrames = uint32_t(maxBframes);
+ if (mLatency && h264type.nBFrames > *mLatency) {
+ h264type.nBFrames = *mLatency;
+ }
+ h264type.nRefFrames = h264type.nBFrames == 0 ? 1 : 2;
h264type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate, h264type.nBFrames);
h264type.nAllowedPictureTypes =
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index dc51b16..b05718c 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -181,6 +181,7 @@
"libcamera_client",
"libcutils",
"libdl",
+ "libdl_android",
"libdrmframework",
"libgui",
"liblog",
@@ -226,6 +227,7 @@
export_shared_lib_headers: [
"libgui",
+ "libhidlmemory",
"libmedia",
"android.hidl.allocator@1.0",
],
@@ -318,21 +320,3 @@
},
}
-subdirs = [
- "codec2",
- "codecs/*",
- "colorconversion",
- "filters",
- "flac/dec",
- "foundation",
- "http",
- "httplive",
- "id3",
- "mpeg2ts",
- "omx",
- "rtsp",
- "tests",
- "timedtext",
- "webm",
- "xmlparser",
-]
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 5194e03..d78d729 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -385,11 +385,11 @@
}
mLastFrameTimestampUs = timeUs;
- size_t numLostBytes = 0;
+ uint64_t numLostBytes = 0; // AudioRecord::getInputFramesLost() returns uint32_t
if (mNumFramesReceived > 0) { // Ignore earlier frame lost
// getInputFramesLost() returns the number of lost frames.
// Convert number of frames lost to number of bytes lost.
- numLostBytes = mRecord->getInputFramesLost() * mRecord->frameSize();
+ numLostBytes = (uint64_t)mRecord->getInputFramesLost() * mRecord->frameSize();
}
CHECK_EQ(numLostBytes & 1, 0u);
@@ -397,11 +397,11 @@
if (numLostBytes > 0) {
// Loss of audio frames should happen rarely; thus the LOGW should
// not cause a logging spam
- ALOGW("Lost audio record data: %zu bytes", numLostBytes);
+ ALOGW("Lost audio record data: %" PRIu64 " bytes", numLostBytes);
}
while (numLostBytes > 0) {
- size_t bufferSize = numLostBytes;
+ uint64_t bufferSize = numLostBytes;
if (numLostBytes > kMaxBufferSize) {
numLostBytes -= kMaxBufferSize;
bufferSize = kMaxBufferSize;
@@ -431,19 +431,17 @@
void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
const size_t bufferSize = buffer->range_length();
const size_t frameSize = mRecord->frameSize();
- const int64_t timestampUs =
- mPrevSampleTimeUs +
- ((1000000LL * (bufferSize / frameSize)) +
- (mSampleRate >> 1)) / mSampleRate;
-
if (mNumFramesReceived == 0) {
buffer->meta_data().setInt64(kKeyAnchorTime, mStartTimeUs);
}
-
+ mNumFramesReceived += bufferSize / frameSize;
+ const int64_t timestampUs =
+ mStartTimeUs +
+ ((1000000LL * mNumFramesReceived) +
+ (mSampleRate >> 1)) / mSampleRate;
buffer->meta_data().setInt64(kKeyTime, mPrevSampleTimeUs);
buffer->meta_data().setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
mPrevSampleTimeUs = timestampUs;
- mNumFramesReceived += bufferSize / frameSize;
mBuffersReceived.push_back(buffer);
mFrameAvailableCondition.signal();
}
diff --git a/media/libstagefright/DataURISource.cpp b/media/libstagefright/DataURISource.cpp
index 3dc345f..b975b38 100644
--- a/media/libstagefright/DataURISource.cpp
+++ b/media/libstagefright/DataURISource.cpp
@@ -13,7 +13,6 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
-
#include <media/stagefright/DataURISource.h>
#include <media/stagefright/foundation/ABuffer.h>
@@ -39,15 +38,27 @@
AString tmp(&uri[5], commaPos - &uri[5]);
if (tmp.endsWith(";base64")) {
- AString encoded(commaPos + 1);
- // Strip CR and LF...
- for (size_t i = encoded.size(); i > 0;) {
- i--;
- if (encoded.c_str()[i] == '\r' || encoded.c_str()[i] == '\n') {
- encoded.erase(i, 1);
+ // strip all CR and LF characters.
+ const char *src = commaPos+1;
+ int len = strlen(src) + 1;
+ char *cleansed = (char *) malloc(len);
+ if (cleansed == NULL) return NULL;
+ char *keeping = cleansed;
+ int left = len;
+ for (int i = 0; i < len ; i++)
+ {
+ const char c = *src++;
+ if (c == '\r' || c == '\n') {
+ continue;
}
+ *keeping++ = c;
+ left--;
}
+ memset(keeping, 0, left);
+
+ AString encoded(cleansed);
+ free(cleansed);
buffer = decodeBase64(encoded);
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index a52da45..2f13dc9 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -71,6 +71,8 @@
static const uint8_t kNalUnitTypePicParamSet = 0x08;
static const int64_t kInitialDelayTimeUs = 700000LL;
static const int64_t kMaxMetadataSize = 0x4000000LL; // 64MB max per-frame metadata size
+static const int64_t kMaxCttsOffsetTimeUs = 30 * 60 * 1000000LL; // 30 minutes
+static const size_t kESDSScratchBufferSize = 10; // kMaxAtomSize in Mpeg4Extractor 64MB
static const char kMetaKey_Version[] = "com.android.version";
static const char kMetaKey_Manufacturer[] = "com.android.manufacturer";
@@ -136,13 +138,6 @@
void resetInternal();
private:
- enum {
- // TODO: need to increase this considering the bug
- // about camera app not sending video frames continuously?
- kMaxCttsOffsetTimeUs = 1000000LL, // 1 second
- kSampleArraySize = 1000,
- };
-
// A helper class to handle faster write box with table entries
template<class TYPE, unsigned ENTRY_SIZE>
// ENTRY_SIZE: # of values in each entry
@@ -2920,14 +2915,18 @@
}
void MPEG4Writer::Track::dumpTimeStamps() {
- ALOGE("Dumping %s track's last 10 frames timestamp and frame type ", getTrackType());
- std::string timeStampString;
- for (std::list<TimestampDebugHelperEntry>::iterator entry = mTimestampDebugHelper.begin();
- entry != mTimestampDebugHelper.end(); ++entry) {
- timeStampString += "(" + std::to_string(entry->pts)+
- "us, " + std::to_string(entry->dts) + "us " + entry->frameType + ") ";
+ if (!mTimestampDebugHelper.empty()) {
+ std::string timeStampString = "Dumping " + std::string(getTrackType()) + " track's last " +
+ std::to_string(mTimestampDebugHelper.size()) +
+ " frames' timestamps(pts, dts) and frame type : ";
+ for (const TimestampDebugHelperEntry& entry : mTimestampDebugHelper) {
+ timeStampString += "\n(" + std::to_string(entry.pts) + "us, " +
+ std::to_string(entry.dts) + "us " + entry.frameType + ") ";
+ }
+ ALOGE("%s", timeStampString.c_str());
+ } else {
+ ALOGE("0 frames to dump timeStamps in %s track ", getTrackType());
}
- ALOGE("%s", timeStampString.c_str());
}
status_t MPEG4Writer::Track::threadEntry() {
@@ -3884,22 +3883,52 @@
mOwner->endBox();
}
+static void generateEsdsSize(size_t dataLength, size_t* sizeGenerated, uint8_t* buffer) {
+ size_t offset = 0, cur = 0;
+ size_t more = 0x00;
+ *sizeGenerated = 0;
+ /* Start with the LSB(7 bits) of dataLength and build the byte sequence upto MSB.
+ * Continuation flag(most significant bit) will be set on the first N-1 bytes.
+ */
+ do {
+ buffer[cur++] = (dataLength & 0x7f) | more;
+ dataLength >>= 7;
+ more = 0x80;
+ ++(*sizeGenerated);
+ } while (dataLength > 0u);
+ --cur;
+ // Reverse the newly formed byte sequence.
+ while (cur > offset) {
+ uint8_t tmp = buffer[cur];
+ buffer[cur--] = buffer[offset];
+ buffer[offset++] = tmp;
+ }
+}
+
void MPEG4Writer::Track::writeMp4aEsdsBox() {
- mOwner->beginBox("esds");
CHECK(mCodecSpecificData);
CHECK_GT(mCodecSpecificDataSize, 0u);
- // Make sure all sizes encode to a single byte.
- CHECK_LT(mCodecSpecificDataSize + 23, 128u);
+ uint8_t sizeESDBuffer[kESDSScratchBufferSize];
+ uint8_t sizeDCDBuffer[kESDSScratchBufferSize];
+ uint8_t sizeDSIBuffer[kESDSScratchBufferSize];
+ size_t sizeESD = 0;
+ size_t sizeDCD = 0;
+ size_t sizeDSI = 0;
+ generateEsdsSize(mCodecSpecificDataSize, &sizeDSI, sizeDSIBuffer);
+ generateEsdsSize(mCodecSpecificDataSize + sizeDSI + 14, &sizeDCD, sizeDCDBuffer);
+ generateEsdsSize(mCodecSpecificDataSize + sizeDSI + sizeDCD + 21, &sizeESD, sizeESDBuffer);
+
+ mOwner->beginBox("esds");
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt8(0x03); // ES_DescrTag
- mOwner->writeInt8(23 + mCodecSpecificDataSize);
+ mOwner->write(sizeESDBuffer, sizeESD);
mOwner->writeInt16(0x0000);// ES_ID
mOwner->writeInt8(0x00);
mOwner->writeInt8(0x04); // DecoderConfigDescrTag
- mOwner->writeInt8(15 + mCodecSpecificDataSize);
+ mOwner->write(sizeDCDBuffer, sizeDCD);
mOwner->writeInt8(0x40); // objectTypeIndication ISO/IEC 14492-2
mOwner->writeInt8(0x15); // streamType AudioStream
@@ -3914,7 +3943,7 @@
mOwner->writeInt32(avgBitrate);
mOwner->writeInt8(0x05); // DecoderSpecificInfoTag
- mOwner->writeInt8(mCodecSpecificDataSize);
+ mOwner->write(sizeDSIBuffer, sizeDSI);
mOwner->write(mCodecSpecificData, mCodecSpecificDataSize);
static const uint8_t kData2[] = {
@@ -3931,20 +3960,27 @@
CHECK(mCodecSpecificData);
CHECK_GT(mCodecSpecificDataSize, 0u);
- // Make sure all sizes encode to a single byte.
- CHECK_LT(23 + mCodecSpecificDataSize, 128u);
+ uint8_t sizeESDBuffer[kESDSScratchBufferSize];
+ uint8_t sizeDCDBuffer[kESDSScratchBufferSize];
+ uint8_t sizeDSIBuffer[kESDSScratchBufferSize];
+ size_t sizeESD = 0;
+ size_t sizeDCD = 0;
+ size_t sizeDSI = 0;
+ generateEsdsSize(mCodecSpecificDataSize, &sizeDSI, sizeDSIBuffer);
+ generateEsdsSize(mCodecSpecificDataSize + sizeDSI + 14, &sizeDCD, sizeDCDBuffer);
+ generateEsdsSize(mCodecSpecificDataSize + sizeDSI + sizeDCD + 21, &sizeESD, sizeESDBuffer);
mOwner->beginBox("esds");
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt8(0x03); // ES_DescrTag
- mOwner->writeInt8(23 + mCodecSpecificDataSize);
+ mOwner->write(sizeESDBuffer, sizeESD);
mOwner->writeInt16(0x0000); // ES_ID
mOwner->writeInt8(0x1f);
mOwner->writeInt8(0x04); // DecoderConfigDescrTag
- mOwner->writeInt8(15 + mCodecSpecificDataSize);
+ mOwner->write(sizeDCDBuffer, sizeDCD);
mOwner->writeInt8(0x20); // objectTypeIndication ISO/IEC 14492-2
mOwner->writeInt8(0x11); // streamType VisualStream
@@ -3962,7 +3998,7 @@
mOwner->writeInt8(0x05); // DecoderSpecificInfoTag
- mOwner->writeInt8(mCodecSpecificDataSize);
+ mOwner->write(sizeDSIBuffer, sizeDSI);
mOwner->write(mCodecSpecificData, mCodecSpecificDataSize);
static const uint8_t kData2[] = {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index d4e4000..b6b7784 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -1052,8 +1052,9 @@
}
// Prevent possible integer overflow in downstream code.
- if ((uint64_t)mVideoWidth * mVideoHeight > (uint64_t)INT32_MAX / 4) {
- ALOGE("buffer size is too big, width=%d, height=%d", mVideoWidth, mVideoHeight);
+ if (mVideoWidth < 0 || mVideoHeight < 0 ||
+ (uint64_t)mVideoWidth * mVideoHeight > (uint64_t)INT32_MAX / 4) {
+ ALOGE("Invalid size(s), width=%d, height=%d", mVideoWidth, mVideoHeight);
return BAD_VALUE;
}
}
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index c3d85ee..50e454c 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -831,7 +831,9 @@
}
void MediaCodecSource::onPause(int64_t pauseStartTimeUs) {
- if ((mFlags & FLAG_USE_SURFACE_INPUT) && (mEncoder != NULL)) {
+ if (mStopping || mOutput.lock()->mEncoderReachedEOS) {
+ // Nothing to do
+ } else if ((mFlags & FLAG_USE_SURFACE_INPUT) && (mEncoder != NULL)) {
sp<AMessage> params = new AMessage;
params->setInt32(PARAMETER_KEY_SUSPEND, true);
params->setInt64(PARAMETER_KEY_SUSPEND_TIME, pauseStartTimeUs);
diff --git a/media/libstagefright/codecs/aacdec/Android.bp b/media/libstagefright/codecs/aacdec/Android.bp
index 25628a2..e0bb5cd 100644
--- a/media/libstagefright/codecs/aacdec/Android.bp
+++ b/media/libstagefright/codecs/aacdec/Android.bp
@@ -29,12 +29,10 @@
static_libs: ["libFraunhoferAAC"],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
"libcutils",
- "liblog",
],
compile_multilib: "32",
}
diff --git a/media/libstagefright/codecs/aacenc/Android.bp b/media/libstagefright/codecs/aacenc/Android.bp
index ec1151b..0d677fe 100644
--- a/media/libstagefright/codecs/aacenc/Android.bp
+++ b/media/libstagefright/codecs/aacenc/Android.bp
@@ -26,11 +26,7 @@
static_libs: ["libFraunhoferAAC"],
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
+
compile_multilib: "32",
}
diff --git a/media/libstagefright/codecs/amrnb/Android.bp b/media/libstagefright/codecs/amrnb/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/amrnb/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/amrnb/dec/Android.bp b/media/libstagefright/codecs/amrnb/dec/Android.bp
index 880f161..f3b272b 100644
--- a/media/libstagefright/codecs/amrnb/dec/Android.bp
+++ b/media/libstagefright/codecs/amrnb/dec/Android.bp
@@ -101,11 +101,9 @@
"libstagefright_amrwbdec",
],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
"libstagefright_amrnb_common",
],
compile_multilib: "32",
diff --git a/media/libstagefright/codecs/amrnb/dec/test/amrnbdec_test.cpp b/media/libstagefright/codecs/amrnb/dec/test/amrnbdec_test.cpp
index 41a9e98..621fda8 100644
--- a/media/libstagefright/codecs/amrnb/dec/test/amrnbdec_test.cpp
+++ b/media/libstagefright/codecs/amrnb/dec/test/amrnbdec_test.cpp
@@ -67,6 +67,7 @@
int bytesRead = fread(header, 1, kFileHeaderSize, fpInput);
if (bytesRead != kFileHeaderSize || memcmp(header, "#!AMR\n", kFileHeaderSize)) {
fprintf(stderr, "Invalid AMR-NB file\n");
+ fclose(fpInput);
return 1;
}
@@ -79,6 +80,7 @@
SNDFILE *handle = sf_open(argv[2], SFM_WRITE, &sfInfo);
if(!handle){
fprintf(stderr, "Could not create %s\n", argv[2]);
+ fclose(fpInput);
return 1;
}
@@ -87,6 +89,8 @@
int err = GSMInitDecode(&amrHandle, (Word8*)"AMRNBDecoder");
if(err != 0){
fprintf(stderr, "Error creating AMR-NB decoder instance\n");
+ fclose(fpInput);
+ sf_close(handle);
return 1;
}
diff --git a/media/libstagefright/codecs/amrnb/enc/Android.bp b/media/libstagefright/codecs/amrnb/enc/Android.bp
index 19fd4a8..1c8b511 100644
--- a/media/libstagefright/codecs/amrnb/enc/Android.bp
+++ b/media/libstagefright/codecs/amrnb/enc/Android.bp
@@ -110,11 +110,9 @@
static_libs: ["libstagefright_amrnbenc"],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
"libstagefright_amrnb_common",
],
compile_multilib: "32",
diff --git a/media/libstagefright/codecs/amrwbenc/Android.bp b/media/libstagefright/codecs/amrwbenc/Android.bp
index b9d45c1..262962f 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.bp
+++ b/media/libstagefright/codecs/amrwbenc/Android.bp
@@ -167,16 +167,11 @@
static_libs: ["libstagefright_amrwbenc"],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
"libstagefright_enc_common",
],
compile_multilib: "32",
}
-//###############################################################################
-
-subdirs = ["SampleCode"]
diff --git a/media/libstagefright/codecs/amrwbenc/SampleCode/AMRWB_E_SAMPLE.c b/media/libstagefright/codecs/amrwbenc/SampleCode/AMRWB_E_SAMPLE.c
index 7c094f3..7282de4 100644
--- a/media/libstagefright/codecs/amrwbenc/SampleCode/AMRWB_E_SAMPLE.c
+++ b/media/libstagefright/codecs/amrwbenc/SampleCode/AMRWB_E_SAMPLE.c
@@ -134,14 +134,16 @@
if(handle == 0)
{
printf("open dll error......");
- return -1;
+ ret = -1;
+ goto safe_exit;
}
pfunc = dlsym(handle, "voGetAMRWBEncAPI");
if(pfunc == 0)
{
printf("open function error......");
- return -1;
+ ret = -1;
+ goto safe_exit;
}
pGetAPI = (VOGETAUDIOENCAPI)pfunc;
@@ -150,7 +152,8 @@
if(returnCode)
{
printf("get APIs error......");
- return -1;
+ ret = -1;
+ goto safe_exit;
}
#else
ret = voGetAMRWBEncAPI(&AudioAPI);
@@ -253,7 +256,8 @@
fclose(fdst);
#ifdef LINUX
- dlclose(handle);
+ if (handle)
+ dlclose(handle);
#endif
return ret;
diff --git a/media/libstagefright/codecs/avcdec/Android.bp b/media/libstagefright/codecs/avcdec/Android.bp
index 8a34845..567bcca 100644
--- a/media/libstagefright/codecs/avcdec/Android.bp
+++ b/media/libstagefright/codecs/avcdec/Android.bp
@@ -22,12 +22,7 @@
"frameworks/native/include/media/openmax",
],
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
sanitize: {
misc_undefined: [
diff --git a/media/libstagefright/codecs/avcenc/Android.bp b/media/libstagefright/codecs/avcenc/Android.bp
index 6371828..0cd39e1 100644
--- a/media/libstagefright/codecs/avcenc/Android.bp
+++ b/media/libstagefright/codecs/avcenc/Android.bp
@@ -16,12 +16,7 @@
"frameworks/native/include/media/openmax",
],
- shared_libs: [
- "libstagefright_foundation",
- "libstagefright_omx",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
sanitize: {
misc_undefined: [
diff --git a/media/libstagefright/codecs/flac/Android.bp b/media/libstagefright/codecs/flac/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/flac/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/flac/dec/Android.bp b/media/libstagefright/codecs/flac/dec/Android.bp
index 3d4a44f..18a3f6b 100644
--- a/media/libstagefright/codecs/flac/dec/Android.bp
+++ b/media/libstagefright/codecs/flac/dec/Android.bp
@@ -28,12 +28,10 @@
cfi: true,
},
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
- "liblog",
"libstagefright_flacdec",
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
],
compile_multilib: "32",
}
diff --git a/media/libstagefright/codecs/flac/enc/Android.bp b/media/libstagefright/codecs/flac/enc/Android.bp
index b32ab08..4149ccd 100644
--- a/media/libstagefright/codecs/flac/enc/Android.bp
+++ b/media/libstagefright/codecs/flac/enc/Android.bp
@@ -19,13 +19,7 @@
],
cfi: true,
},
-
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
header_libs: ["libbase_headers"],
static_libs: [
diff --git a/media/libstagefright/codecs/g711/Android.bp b/media/libstagefright/codecs/g711/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/g711/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/g711/dec/Android.bp b/media/libstagefright/codecs/g711/dec/Android.bp
index 7097688..c273179 100644
--- a/media/libstagefright/codecs/g711/dec/Android.bp
+++ b/media/libstagefright/codecs/g711/dec/Android.bp
@@ -12,12 +12,7 @@
"frameworks/native/include/media/openmax",
],
- shared_libs: [
- "libstagefright_foundation",
- "libstagefright_omx",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
cflags: ["-Werror"],
diff --git a/media/libstagefright/codecs/gsm/Android.bp b/media/libstagefright/codecs/gsm/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/gsm/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/gsm/dec/Android.bp b/media/libstagefright/codecs/gsm/dec/Android.bp
index a973f70..3c5ebfe 100644
--- a/media/libstagefright/codecs/gsm/dec/Android.bp
+++ b/media/libstagefright/codecs/gsm/dec/Android.bp
@@ -25,12 +25,7 @@
cfi: true,
},
- shared_libs: [
- "libstagefright_foundation",
- "libstagefright_omx",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
static_libs: ["libgsm"],
compile_multilib: "32",
diff --git a/media/libstagefright/codecs/hevcdec/Android.bp b/media/libstagefright/codecs/hevcdec/Android.bp
index 60fc446..cc91d53 100644
--- a/media/libstagefright/codecs/hevcdec/Android.bp
+++ b/media/libstagefright/codecs/hevcdec/Android.bp
@@ -30,12 +30,7 @@
cfi: true,
},
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
// We need this because the current asm generates the following link error:
// requires unsupported dynamic reloc R_ARM_REL32; recompile with -fPIC
diff --git a/media/libstagefright/codecs/m4v_h263/Android.bp b/media/libstagefright/codecs/m4v_h263/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/m4v_h263/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/m4v_h263/dec/Android.bp b/media/libstagefright/codecs/m4v_h263/dec/Android.bp
index 41141b1..0523143 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/Android.bp
+++ b/media/libstagefright/codecs/m4v_h263/dec/Android.bp
@@ -91,12 +91,7 @@
static_libs: ["libstagefright_m4vh263dec"],
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
sanitize: {
misc_undefined: [
diff --git a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
index a8fcdd1..60750d9 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/SoftMPEG4.cpp
@@ -24,7 +24,6 @@
#include <media/stagefright/foundation/AUtils.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
-#include <media/IOMX.h>
#include "mp4dec_api.h"
@@ -118,9 +117,14 @@
outHeader->nFlags = OMX_BUFFERFLAG_EOS;
List<BufferInfo *>::iterator it = outQueue.begin();
- while ((*it)->mHeader != outHeader) {
+ while (it != outQueue.end() && (*it)->mHeader != outHeader) {
++it;
}
+ if (it == outQueue.end()) {
+ ALOGE("couldn't find port buffer %d in outQueue: b/109891727", mNumSamplesOutput & 1);
+ android_errorWriteLog(0x534e4554, "109891727");
+ return;
+ }
BufferInfo *outInfo = *it;
outInfo->mOwnedByUs = false;
diff --git a/media/libstagefright/codecs/m4v_h263/enc/Android.bp b/media/libstagefright/codecs/m4v_h263/enc/Android.bp
index d4f7d50..d38f4b1 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/Android.bp
+++ b/media/libstagefright/codecs/m4v_h263/enc/Android.bp
@@ -77,12 +77,7 @@
static_libs: ["libstagefright_m4vh263enc"],
- shared_libs: [
- "libstagefright_foundation",
- "libstagefright_omx",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
sanitize: {
misc_undefined: [
diff --git a/media/libstagefright/codecs/mp3dec/Android.bp b/media/libstagefright/codecs/mp3dec/Android.bp
index 2154f84..9173ed6 100644
--- a/media/libstagefright/codecs/mp3dec/Android.bp
+++ b/media/libstagefright/codecs/mp3dec/Android.bp
@@ -105,12 +105,7 @@
cfi: true,
},
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
static_libs: ["libstagefright_mp3dec"],
compile_multilib: "32",
diff --git a/media/libstagefright/codecs/mpeg2dec/Android.bp b/media/libstagefright/codecs/mpeg2dec/Android.bp
index c655544..26e786e 100644
--- a/media/libstagefright/codecs/mpeg2dec/Android.bp
+++ b/media/libstagefright/codecs/mpeg2dec/Android.bp
@@ -20,12 +20,7 @@
"frameworks/native/include/media/openmax",
],
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
ldflags: ["-Wl,-Bsymbolic"],
diff --git a/media/libstagefright/codecs/on2/Android.bp b/media/libstagefright/codecs/on2/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/on2/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/on2/dec/Android.bp b/media/libstagefright/codecs/on2/dec/Android.bp
index 174f183..abd21d7 100644
--- a/media/libstagefright/codecs/on2/dec/Android.bp
+++ b/media/libstagefright/codecs/on2/dec/Android.bp
@@ -14,12 +14,7 @@
static_libs: ["libvpx"],
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
cflags: ["-Werror"],
diff --git a/media/libstagefright/codecs/on2/enc/Android.bp b/media/libstagefright/codecs/on2/enc/Android.bp
index 891a771..ea46bad 100644
--- a/media/libstagefright/codecs/on2/enc/Android.bp
+++ b/media/libstagefright/codecs/on2/enc/Android.bp
@@ -30,11 +30,7 @@
static_libs: ["libvpx"],
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
+
compile_multilib: "32",
}
diff --git a/media/libstagefright/codecs/on2/enc/SoftVP8Encoder.h b/media/libstagefright/codecs/on2/enc/SoftVP8Encoder.h
index b4904bf..c5c2abf 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVP8Encoder.h
+++ b/media/libstagefright/codecs/on2/enc/SoftVP8Encoder.h
@@ -23,8 +23,6 @@
#include <OMX_VideoExt.h>
#include <OMX_IndexExt.h>
-#include <hardware/gralloc.h>
-
#include "vpx/vpx_encoder.h"
#include "vpx/vpx_codec.h"
#include "vpx/vp8cx.h"
diff --git a/media/libstagefright/codecs/on2/enc/SoftVP9Encoder.h b/media/libstagefright/codecs/on2/enc/SoftVP9Encoder.h
index 85df69a..308a9ac 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVP9Encoder.h
+++ b/media/libstagefright/codecs/on2/enc/SoftVP9Encoder.h
@@ -23,8 +23,6 @@
#include <OMX_VideoExt.h>
#include <OMX_IndexExt.h>
-#include <hardware/gralloc.h>
-
#include "vpx/vpx_encoder.h"
#include "vpx/vpx_codec.h"
#include "vpx/vp8cx.h"
diff --git a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h
index 263d134..7208d69 100644
--- a/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h
+++ b/media/libstagefright/codecs/on2/enc/SoftVPXEncoder.h
@@ -23,8 +23,6 @@
#include <OMX_VideoExt.h>
#include <OMX_IndexExt.h>
-#include <hardware/gralloc.h>
-
#include "vpx/vpx_encoder.h"
#include "vpx/vpx_codec.h"
#include "vpx/vp8cx.h"
diff --git a/media/libstagefright/codecs/opus/Android.bp b/media/libstagefright/codecs/opus/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/opus/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/opus/dec/Android.bp b/media/libstagefright/codecs/opus/dec/Android.bp
index afe459d..bfcae07 100644
--- a/media/libstagefright/codecs/opus/dec/Android.bp
+++ b/media/libstagefright/codecs/opus/dec/Android.bp
@@ -12,12 +12,10 @@
"frameworks/native/include/media/openmax",
],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
"libopus",
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
],
cflags: ["-Werror"],
diff --git a/media/libstagefright/codecs/raw/Android.bp b/media/libstagefright/codecs/raw/Android.bp
index f822445..1c23bad 100644
--- a/media/libstagefright/codecs/raw/Android.bp
+++ b/media/libstagefright/codecs/raw/Android.bp
@@ -24,11 +24,7 @@
cfi: true,
},
- shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
- ],
+ defaults: ["omx_soft_libs"],
+
compile_multilib: "32",
}
diff --git a/media/libstagefright/codecs/vorbis/Android.bp b/media/libstagefright/codecs/vorbis/Android.bp
deleted file mode 100644
index b44c296..0000000
--- a/media/libstagefright/codecs/vorbis/Android.bp
+++ /dev/null
@@ -1 +0,0 @@
-subdirs = ["*"]
diff --git a/media/libstagefright/codecs/vorbis/dec/Android.bp b/media/libstagefright/codecs/vorbis/dec/Android.bp
index a9265cb..2d1a922 100644
--- a/media/libstagefright/codecs/vorbis/dec/Android.bp
+++ b/media/libstagefright/codecs/vorbis/dec/Android.bp
@@ -12,12 +12,10 @@
"frameworks/native/include/media/openmax",
],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
"libvorbisidec",
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
- "liblog",
],
cflags: ["-Werror"],
diff --git a/media/libstagefright/codecs/xaacdec/Android.bp b/media/libstagefright/codecs/xaacdec/Android.bp
index 7392f1e..e49eb8f 100644
--- a/media/libstagefright/codecs/xaacdec/Android.bp
+++ b/media/libstagefright/codecs/xaacdec/Android.bp
@@ -24,12 +24,10 @@
static_libs: ["libxaacdec"],
+ defaults: ["omx_soft_libs"],
+
shared_libs: [
- "libstagefright_omx",
- "libstagefright_foundation",
- "libutils",
"libcutils",
- "liblog",
],
compile_multilib: "32",
diff --git a/media/libstagefright/data/media_codecs_google_audio.xml b/media/libstagefright/data/media_codecs_google_audio.xml
index 632088a..8899adc 100644
--- a/media/libstagefright/data/media_codecs_google_audio.xml
+++ b/media/libstagefright/data/media_codecs_google_audio.xml
@@ -58,7 +58,7 @@
</MediaCodec>
<MediaCodec name="OMX.google.raw.decoder" type="audio/raw">
<Limit name="channel-count" max="8" />
- <Limit name="sample-rate" ranges="8000-96000" />
+ <Limit name="sample-rate" ranges="8000-192000" />
<Limit name="bitrate" range="1-10000000" />
</MediaCodec>
<MediaCodec name="OMX.google.flac.decoder" type="audio/flac">
diff --git a/media/libstagefright/data/media_codecs_google_c2_audio.xml b/media/libstagefright/data/media_codecs_google_c2_audio.xml
index 47a9715..be2404d 100644
--- a/media/libstagefright/data/media_codecs_google_c2_audio.xml
+++ b/media/libstagefright/data/media_codecs_google_c2_audio.xml
@@ -67,7 +67,7 @@
<MediaCodec name="c2.android.raw.decoder" type="audio/raw">
<Alias name="OMX.google.raw.decoder" />
<Limit name="channel-count" max="8" />
- <Limit name="sample-rate" ranges="8000-96000" />
+ <Limit name="sample-rate" ranges="8000-192000" />
<Limit name="bitrate" range="1-10000000" />
</MediaCodec>
<MediaCodec name="c2.android.flac.decoder" type="audio/flac">
diff --git a/media/libstagefright/include/media/stagefright/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 9d46d2d..784fd36 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -286,7 +286,7 @@
double mFps;
double mCaptureFps;
bool mCreateInputBuffersSuspended;
- uint32_t mLatency;
+ std::optional<uint32_t> mLatency;
bool mTunneled;
diff --git a/media/libstagefright/include/media/stagefright/MPEG4Writer.h b/media/libstagefright/include/media/stagefright/MPEG4Writer.h
index 803155d..6f19023 100644
--- a/media/libstagefright/include/media/stagefright/MPEG4Writer.h
+++ b/media/libstagefright/include/media/stagefright/MPEG4Writer.h
@@ -82,10 +82,6 @@
kWhatSwitch = 'swch',
};
- enum {
- kMaxCttsOffsetTimeUs = 1000000LL, // 1 second
- };
-
int mFd;
int mNextFd;
sp<MetaData> mStartMeta;
diff --git a/media/libstagefright/include/media/stagefright/MediaCodecConstants.h b/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
index 2dca5c3..8b6944b 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodecConstants.h
@@ -772,6 +772,7 @@
constexpr char KEY_LANGUAGE[] = "language";
constexpr char KEY_LATENCY[] = "latency";
constexpr char KEY_LEVEL[] = "level";
+constexpr char KEY_MAX_B_FRAMES[] = "max-bframes";
constexpr char KEY_MAX_BIT_RATE[] = "max-bitrate";
constexpr char KEY_MAX_FPS_TO_ENCODER[] = "max-fps-to-encoder";
constexpr char KEY_MAX_HEIGHT[] = "max-height";
diff --git a/media/libstagefright/omx/Android.bp b/media/libstagefright/omx/Android.bp
index 4383004..c06aca5 100644
--- a/media/libstagefright/omx/Android.bp
+++ b/media/libstagefright/omx/Android.bp
@@ -11,11 +11,6 @@
"OMXNodeInstance.cpp",
"OMXUtils.cpp",
"OmxGraphicBufferSource.cpp",
- "SimpleSoftOMXComponent.cpp",
- "SoftOMXComponent.cpp",
- "SoftOMXPlugin.cpp",
- "SoftVideoDecoderOMXComponent.cpp",
- "SoftVideoEncoderOMXComponent.cpp",
"1.0/Omx.cpp",
"1.0/OmxStore.cpp",
"1.0/WGraphicBufferSource.cpp",
@@ -56,6 +51,7 @@
"libvndksupport",
"android.hardware.media.omx@1.0",
"android.hardware.graphics.bufferqueue@1.0",
+ "libstagefright_omx_soft",
],
export_shared_lib_headers: [
@@ -81,6 +77,64 @@
},
}
+cc_defaults {
+ name: "omx_soft_libs",
+ shared_libs: [
+ "libutils",
+ "liblog",
+ "libstagefright_foundation",
+ "libstagefright_omx_soft",
+ ],
+}
+
+cc_library_shared {
+ name: "libstagefright_omx_soft",
+ vendor_available: true,
+ vndk: {
+ enabled: true,
+ },
+
+ srcs: [
+ "SimpleSoftOMXComponent.cpp",
+ "SoftOMXComponent.cpp",
+ "SoftOMXPlugin.cpp",
+ "SoftVideoDecoderOMXComponent.cpp",
+ "SoftVideoEncoderOMXComponent.cpp",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ "libui",
+ "libstagefright_foundation",
+ ],
+
+ export_shared_lib_headers: [
+ "libstagefright_foundation",
+ "libutils",
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ "-Wno-unused-parameter",
+ "-Wno-documentation",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "signed-integer-overflow",
+ "unsigned-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
+
cc_library_shared {
name: "libstagefright_omx_utils",
vendor_available: true,
@@ -116,6 +170,3 @@
cflags: ["-Wall", "-Werror"],
}
-//###############################################################################
-
-subdirs = ["tests"]
diff --git a/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp
index 2fbbb44..d75acda 100644
--- a/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftVideoEncoderOMXComponent.cpp
@@ -559,7 +559,7 @@
if (nativeMeta.nFenceFd >= 0) {
sp<Fence> fence = new Fence(nativeMeta.nFenceFd);
nativeMeta.nFenceFd = -1;
- status_t err = fence->wait(IOMX::kFenceTimeoutMs);
+ status_t err = fence->wait(kFenceTimeoutMs);
if (err != OK) {
ALOGE("Timed out waiting on input fence");
return NULL;
diff --git a/media/libstagefright/omx/include/media/stagefright/omx/SoftOMXComponent.h b/media/libstagefright/omx/include/media/stagefright/omx/SoftOMXComponent.h
index 3ab6f88..79f0c77 100644
--- a/media/libstagefright/omx/include/media/stagefright/omx/SoftOMXComponent.h
+++ b/media/libstagefright/omx/include/media/stagefright/omx/SoftOMXComponent.h
@@ -21,12 +21,15 @@
#include <media/stagefright/foundation/ABase.h>
#include <media/stagefright/foundation/AString.h>
#include <utils/RefBase.h>
-
+#include <utils/Log.h>
#include <OMX_Component.h>
namespace android {
struct SoftOMXComponent : public RefBase {
+ enum {
+ kFenceTimeoutMs = 1000
+ };
SoftOMXComponent(
const char *name,
const OMX_CALLBACKTYPE *callbacks,
diff --git a/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoDecoderOMXComponent.h b/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoDecoderOMXComponent.h
index 3b381ce..d7c1658 100644
--- a/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoDecoderOMXComponent.h
+++ b/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoDecoderOMXComponent.h
@@ -23,7 +23,10 @@
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/AHandlerReflector.h>
#include <media/stagefright/foundation/ColorUtils.h>
-#include <media/IOMX.h>
+#include <media/openmax/OMX_Core.h>
+#include <media/openmax/OMX_Video.h>
+#include <media/openmax/OMX_VideoExt.h>
+
#include <media/hardware/HardwareAPI.h>
#include <utils/RefBase.h>
diff --git a/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h b/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h
index 2d6f31b..9cb72dd 100644
--- a/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h
+++ b/media/libstagefright/omx/include/media/stagefright/omx/SoftVideoEncoderOMXComponent.h
@@ -18,7 +18,9 @@
#define SOFT_VIDEO_ENCODER_OMX_COMPONENT_H_
-#include <media/IOMX.h>
+#include <media/openmax/OMX_Core.h>
+#include <media/openmax/OMX_Video.h>
+#include <media/openmax/OMX_VideoExt.h>
#include "SimpleSoftOMXComponent.h"
diff --git a/media/libstagefright/xmlparser/vts/Android.bp b/media/libstagefright/xmlparser/vts/Android.bp
new file mode 100644
index 0000000..3f93e9e
--- /dev/null
+++ b/media/libstagefright/xmlparser/vts/Android.bp
@@ -0,0 +1,33 @@
+//
+// Copyright (C) 2019 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_test {
+ name: "vts_mediaCodecs_validate_test",
+ srcs: [
+ "ValidateMediaCodecs.cpp"
+ ],
+ static_libs: [
+ "android.hardware.audio.common.test.utility",
+ "libxml2",
+ ],
+ shared_libs: [
+ "liblog",
+ ],
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+}
diff --git a/media/libstagefright/xmlparser/vts/Android.mk b/media/libstagefright/xmlparser/vts/Android.mk
new file mode 100644
index 0000000..d5290ba
--- /dev/null
+++ b/media/libstagefright/xmlparser/vts/Android.mk
@@ -0,0 +1,22 @@
+#
+# Copyright (C) 2019 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+#
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := VtsValidateMediaCodecs
+include test/vts/tools/build/Android.host_config.mk
diff --git a/media/libstagefright/xmlparser/vts/AndroidTest.xml b/media/libstagefright/xmlparser/vts/AndroidTest.xml
new file mode 100644
index 0000000..97ee107
--- /dev/null
+++ b/media/libstagefright/xmlparser/vts/AndroidTest.xml
@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Config for VTS VtsValidateMediaCodecs.">
+ <option name="config-descriptor:metadata" key="plan" value="vts-treble" />
+ <target_preparer class="com.android.compatibility.common.tradefed.targetprep.VtsFilePusher">
+ <option name="abort-on-push-failure" value="false"/>
+ <option name="push-group" value="HostDrivenTest.push"/>
+ <option name="push" value="DATA/etc/media_codecs.xsd->/data/local/tmp/media_codecs.xsd"/>
+ </target_preparer>
+ <test class="com.android.tradefed.testtype.VtsMultiDeviceTest">
+ <option name="test-module-name" value="VtsValidateMediaCodecs"/>
+ <option name="binary-test-source" value="_32bit::DATA/nativetest/vts_mediaCodecs_validate_test/vts_mediaCodecs_validate_test" />
+ <option name="binary-test-source" value="_64bit::DATA/nativetest64/vts_mediaCodecs_validate_test/vts_mediaCodecs_validate_test" />
+ <option name="binary-test-type" value="gtest"/>
+ <option name="test-timeout" value="30s"/>
+ </test>
+</configuration>
diff --git a/media/libstagefright/xmlparser/vts/ValidateMediaCodecs.cpp b/media/libstagefright/xmlparser/vts/ValidateMediaCodecs.cpp
new file mode 100644
index 0000000..b07833e
--- /dev/null
+++ b/media/libstagefright/xmlparser/vts/ValidateMediaCodecs.cpp
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <dirent.h>
+#include <regex>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/stat.h>
+#include <unistd.h>
+#include <string>
+
+#include "utility/ValidateXml.h"
+
+static void get_files_in_dirs(const char* dir_path, std::vector<std::string>& files) {
+ DIR* d;
+ struct dirent* de;
+
+ d = opendir(dir_path);
+ if (d == nullptr) {
+ return;
+ }
+
+ while ((de = readdir(d))) {
+ if (de->d_type != DT_REG) {
+ continue;
+ }
+ if (std::regex_match(de->d_name, std::regex("(media_codecs)(.*)(.xml)"))) {
+ files.push_back(de->d_name);
+ }
+ }
+ closedir(d);
+}
+
+TEST(CheckConfig, mediaCodecsValidation) {
+ RecordProperty("description",
+ "Verify that the media codecs file "
+ "is valid according to the schema");
+
+ const char* location = "/vendor/etc";
+
+ std::vector<std::string> files;
+ get_files_in_dirs(location, files);
+
+ for (std::string file_name : files) {
+ EXPECT_ONE_VALID_XML_MULTIPLE_LOCATIONS(file_name.c_str(), {location},
+ "/data/local/tmp/media_codecs.xsd");
+ }
+}
diff --git a/media/mediaserver/mediaserver.rc b/media/mediaserver/mediaserver.rc
index 8cfcd79..f6c325c 100644
--- a/media/mediaserver/mediaserver.rc
+++ b/media/mediaserver/mediaserver.rc
@@ -2,7 +2,5 @@
class main
user media
group audio camera inet net_bt net_bt_admin net_bw_acct drmrpc mediadrm
- # TODO(b/123275379): Remove updatable when http://aosp/878198 has landed
- updatable
ioprio rt 4
writepid /dev/cpuset/foreground/tasks /dev/stune/foreground/tasks
diff --git a/media/mtp/Android.bp b/media/mtp/Android.bp
index 2cf9b82..66a3139 100644
--- a/media/mtp/Android.bp
+++ b/media/mtp/Android.bp
@@ -54,6 +54,3 @@
],
}
-subdirs = [
- "tests",
-]
diff --git a/media/ndk/Android.bp b/media/ndk/Android.bp
index f4cc704..a4f5730 100644
--- a/media/ndk/Android.bp
+++ b/media/ndk/Android.bp
@@ -73,6 +73,7 @@
shared_libs: [
"android.hardware.graphics.bufferqueue@1.0",
"android.hidl.token@1.0-utils",
+ "libandroid_runtime_lazy",
"libbinder",
"libmedia",
"libmedia_omx",
@@ -93,12 +94,6 @@
"libmediandk_utils",
],
- required: [
- // libmediandk may be used by Java and non-Java things. When lower-level things use it,
- // they shouldn't have to take on the cost of loading libandroid_runtime.
- "libandroid_runtime",
- ],
-
export_include_dirs: ["include"],
export_shared_lib_headers: [
diff --git a/media/ndk/NdkMediaDataSource.cpp b/media/ndk/NdkMediaDataSource.cpp
index 0891f2a..7979c2f 100644
--- a/media/ndk/NdkMediaDataSource.cpp
+++ b/media/ndk/NdkMediaDataSource.cpp
@@ -23,7 +23,8 @@
#include <jni.h>
#include <unistd.h>
-#include <binder/IBinder.h>
+#include <android_runtime/AndroidRuntime.h>
+#include <android_util_Binder.h>
#include <cutils/properties.h>
#include <utils/Log.h>
#include <utils/StrongPointer.h>
@@ -39,67 +40,9 @@
#include "../../libstagefright/include/NuCachedSource2.h"
#include "NdkMediaDataSourceCallbacksPriv.h"
-#include <mutex> // std::call_once,once_flag
-#include <dlfcn.h> // dlopen
using namespace android;
-// load libandroid_runtime.so lazily.
-// A vendor process may use libmediandk but should not depend on libandroid_runtime.
-// TODO(jooyung): remove duplicate (b/125550121)
-// frameworks/native/libs/binder/ndk/ibinder_jni.cpp
-namespace {
-
-typedef JNIEnv* (*getJNIEnv_t)();
-typedef sp<IBinder> (*ibinderForJavaObject_t)(JNIEnv* env, jobject obj);
-
-getJNIEnv_t getJNIEnv_;
-ibinderForJavaObject_t ibinderForJavaObject_;
-
-std::once_flag mLoadFlag;
-
-void load() {
- std::call_once(mLoadFlag, []() {
- void* handle = dlopen("libandroid_runtime.so", RTLD_LAZY);
- if (handle == nullptr) {
- ALOGE("Could not open libandroid_runtime.");
- return;
- }
-
- getJNIEnv_ = reinterpret_cast<getJNIEnv_t>(
- dlsym(handle, "_ZN7android14AndroidRuntime9getJNIEnvEv"));
- if (getJNIEnv_ == nullptr) {
- ALOGE("Could not find AndroidRuntime::getJNIEnv.");
- // no return
- }
-
- ibinderForJavaObject_ = reinterpret_cast<ibinderForJavaObject_t>(
- dlsym(handle, "_ZN7android20ibinderForJavaObjectEP7_JNIEnvP8_jobject"));
- if (ibinderForJavaObject_ == nullptr) {
- ALOGE("Could not find ibinderForJavaObject.");
- // no return
- }
- });
-}
-
-JNIEnv* getJNIEnv() {
- load();
- if (getJNIEnv_ == nullptr) {
- return nullptr;
- }
- return (getJNIEnv_)();
-}
-
-sp<IBinder> ibinderForJavaObject(JNIEnv* env, jobject obj) {
- load();
- if (ibinderForJavaObject_ == nullptr) {
- return nullptr;
- }
- return (ibinderForJavaObject_)(env, obj);
-}
-
-} // namespace
-
struct AMediaDataSource {
void *userdata;
AMediaDataSourceReadAt readAt;
@@ -181,14 +124,9 @@
if (obj == NULL) {
return NULL;
}
- sp<IBinder> binder;
switch (version) {
case 1:
- binder = ibinderForJavaObject(env, obj);
- if (binder == NULL) {
- return NULL;
- }
- return interface_cast<IMediaHTTPService>(binder);
+ return interface_cast<IMediaHTTPService>(ibinderForJavaObject(env, obj));
case 2:
return new JMedia2HTTPService(env, obj);
default:
@@ -241,7 +179,7 @@
switch (version) {
case 1:
- env = getJNIEnv();
+ env = AndroidRuntime::getJNIEnv();
clazz = "android/media/MediaHTTPService";
method = "createHttpServiceBinderIfNecessary";
signature = "(Ljava/lang/String;)Landroid/os/IBinder;";
diff --git a/media/utils/ServiceUtilities.cpp b/media/utils/ServiceUtilities.cpp
index 2fb24f5..768cd1e 100644
--- a/media/utils/ServiceUtilities.cpp
+++ b/media/utils/ServiceUtilities.cpp
@@ -130,14 +130,22 @@
return ok;
}
-bool captureHotwordAllowed(pid_t pid, uid_t uid) {
+bool captureMediaOutputAllowed(pid_t pid, uid_t uid) {
+ if (isAudioServerOrRootUid(uid)) return true;
+ static const String16 sCaptureMediaOutput("android.permission.CAPTURE_MEDIA_OUTPUT");
+ bool ok = PermissionCache::checkPermission(sCaptureMediaOutput, pid, uid);
+ if (!ok) ALOGE("Request requires android.permission.CAPTURE_MEDIA_OUTPUT");
+ return ok;
+}
+
+bool captureHotwordAllowed(const String16& opPackageName, pid_t pid, uid_t uid) {
// CAPTURE_AUDIO_HOTWORD permission implies RECORD_AUDIO permission
- bool ok = recordingAllowed(String16(""), pid, uid);
+ bool ok = recordingAllowed(opPackageName, pid, uid);
if (ok) {
static const String16 sCaptureHotwordAllowed("android.permission.CAPTURE_AUDIO_HOTWORD");
// IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
- ok = PermissionCache::checkCallingPermission(sCaptureHotwordAllowed);
+ ok = PermissionCache::checkPermission(sCaptureHotwordAllowed, pid, uid);
}
if (!ok) ALOGE("android.permission.CAPTURE_AUDIO_HOTWORD");
return ok;
diff --git a/media/utils/include/mediautils/ServiceUtilities.h b/media/utils/include/mediautils/ServiceUtilities.h
index 94370ee..c5fe05f 100644
--- a/media/utils/include/mediautils/ServiceUtilities.h
+++ b/media/utils/include/mediautils/ServiceUtilities.h
@@ -74,7 +74,8 @@
bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid);
void finishRecording(const String16& opPackageName, uid_t uid);
bool captureAudioOutputAllowed(pid_t pid, uid_t uid);
-bool captureHotwordAllowed(pid_t pid, uid_t uid);
+bool captureMediaOutputAllowed(pid_t pid, uid_t uid);
+bool captureHotwordAllowed(const String16& opPackageName, pid_t pid, uid_t uid);
bool settingsAllowed();
bool modifyAudioRoutingAllowed();
bool modifyDefaultAudioEffectsAllowed();
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
new file mode 100644
index 0000000..96ad54b
--- /dev/null
+++ b/services/audioflinger/Android.bp
@@ -0,0 +1,73 @@
+
+
+cc_library_shared {
+ name: "libaudioflinger",
+
+ srcs: [
+ "AudioFlinger.cpp",
+ "AudioHwDevice.cpp",
+ "AudioStreamOut.cpp",
+ "AudioWatchdog.cpp",
+ "BufLog.cpp",
+ "Effects.cpp",
+ "FastCapture.cpp",
+ "FastCaptureDumpState.cpp",
+ "FastCaptureState.cpp",
+ "FastMixer.cpp",
+ "FastMixerDumpState.cpp",
+ "FastMixerState.cpp",
+ "FastThread.cpp",
+ "FastThreadDumpState.cpp",
+ "FastThreadState.cpp",
+ "NBAIO_Tee.cpp",
+ "PatchPanel.cpp",
+ "SpdifStreamOut.cpp",
+ "StateQueue.cpp",
+ "Threads.cpp",
+ "Tracks.cpp",
+ "TypedLogger.cpp",
+ ],
+
+ include_dirs: [
+ "frameworks/av/services/audiopolicy",
+ "frameworks/av/services/medialog",
+ ],
+
+ shared_libs: [
+ "libaudiohal",
+ "libaudioprocessing",
+ "libaudiospdif",
+ "libaudioutils",
+ "libcutils",
+ "libutils",
+ "liblog",
+ "libbinder",
+ "libaudioclient",
+ "libmedialogservice",
+ "libmediametrics",
+ "libmediautils",
+ "libnbaio",
+ "libnblog",
+ "libpowermanager",
+ "libmediautils",
+ "libmemunreachable",
+ "libmedia_helper",
+ "libvibrator",
+ ],
+
+ static_libs: [
+ "libcpustats",
+ "libsndfile",
+ ],
+
+ cflags: [
+ "-DSTATE_QUEUE_INSTANTIATIONS=\"StateQueueInstantiations.cpp\"",
+ "-fvisibility=hidden",
+ "-Werror",
+ "-Wall",
+ ],
+ sanitize: {
+ integer_overflow: true,
+ },
+
+}
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
deleted file mode 100644
index 40980a6..0000000
--- a/services/audioflinger/Android.mk
+++ /dev/null
@@ -1,74 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- AudioFlinger.cpp \
- Threads.cpp \
- Tracks.cpp \
- AudioHwDevice.cpp \
- AudioStreamOut.cpp \
- SpdifStreamOut.cpp \
- Effects.cpp \
- PatchPanel.cpp \
- StateQueue.cpp \
- BufLog.cpp \
- TypedLogger.cpp \
- NBAIO_Tee.cpp \
-
-LOCAL_C_INCLUDES := \
- frameworks/av/services/audiopolicy \
- frameworks/av/services/medialog \
- $(call include-path-for, audio-utils)
-
-LOCAL_SHARED_LIBRARIES := \
- libaudiohal \
- libaudioprocessing \
- libaudiospdif \
- libaudioutils \
- libcutils \
- libutils \
- liblog \
- libbinder \
- libaudioclient \
- libmedialogservice \
- libmediametrics \
- libmediautils \
- libnbaio \
- libnblog \
- libpowermanager \
- libmediautils \
- libmemunreachable \
- libmedia_helper \
- libvibrator
-
-LOCAL_STATIC_LIBRARIES := \
- libcpustats \
- libsndfile \
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_MODULE:= libaudioflinger
-
-LOCAL_SRC_FILES += \
- AudioWatchdog.cpp \
- FastCapture.cpp \
- FastCaptureDumpState.cpp \
- FastCaptureState.cpp \
- FastMixer.cpp \
- FastMixerDumpState.cpp \
- FastMixerState.cpp \
- FastThread.cpp \
- FastThreadDumpState.cpp \
- FastThreadState.cpp
-
-LOCAL_CFLAGS += -DSTATE_QUEUE_INSTANTIATIONS='"StateQueueInstantiations.cpp"'
-
-LOCAL_CFLAGS += -fvisibility=hidden
-
-LOCAL_CFLAGS += -Werror -Wall
-LOCAL_SANITIZE := integer_overflow
-
-include $(BUILD_SHARED_LIBRARY)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 43260c2..0825cb4 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -466,15 +466,8 @@
bool AudioFlinger::dumpTryLock(Mutex& mutex)
{
- bool locked = false;
- for (int i = 0; i < kDumpLockRetries; ++i) {
- if (mutex.tryLock() == NO_ERROR) {
- locked = true;
- break;
- }
- usleep(kDumpLockSleepUs);
- }
- return locked;
+ status_t err = mutex.timedLock(kDumpLockTimeoutNs);
+ return err == NO_ERROR;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 3804418..9960f0e 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -433,8 +433,7 @@
static uint32_t mScreenState;
// Internal dump utilities.
- static const int kDumpLockRetries = 50;
- static const int kDumpLockSleepUs = 20000;
+ static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND;
static bool dumpTryLock(Mutex& mutex);
void dumpPermissionDenial(int fd, const Vector<String16>& args);
void dumpClients(int fd, const Vector<String16>& args);
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index ecaeb52..2b34267 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -2335,13 +2335,10 @@
void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
{
- const size_t SIZE = 256;
- char buffer[SIZE];
String8 result;
- size_t numEffects = mEffects.size();
- snprintf(buffer, SIZE, " %zu effects for session %d\n", numEffects, mSessionId);
- result.append(buffer);
+ const size_t numEffects = mEffects.size();
+ result.appendFormat(" %zu effects for session %d\n", numEffects, mSessionId);
if (numEffects) {
bool locked = AudioFlinger::dumpTryLock(mLock);
@@ -2369,6 +2366,8 @@
if (locked) {
mLock.unlock();
}
+ } else {
+ write(fd, result.string(), result.size());
}
}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 0e1e97f..04d62fa 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -786,12 +786,8 @@
}
}
-void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::ThreadBase::dump(int fd, const Vector<String16>& args)
{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
this, mThreadName, getTid(), type(), threadTypeToString(type()));
@@ -800,6 +796,21 @@
dprintf(fd, " Thread may be deadlocked\n");
}
+ dumpBase_l(fd, args);
+ dumpInternals_l(fd, args);
+ dumpTracks_l(fd, args);
+ dumpEffectChains_l(fd, args);
+
+ if (locked) {
+ mLock.unlock();
+ }
+
+ dprintf(fd, " Local log:\n");
+ mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
+}
+
+void AudioFlinger::ThreadBase::dumpBase_l(int fd, const Vector<String16>& args __unused)
+{
dprintf(fd, " I/O handle: %d\n", mId);
dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no");
dprintf(fd, " Sample rate: %u Hz\n", mSampleRate);
@@ -814,6 +825,8 @@
dprintf(fd, " Pending config events:");
size_t numConfig = mConfigEvents.size();
if (numConfig) {
+ const size_t SIZE = 256;
+ char buffer[SIZE];
for (size_t i = 0; i < numConfig; i++) {
mConfigEvents[i]->dump(buffer, SIZE);
dprintf(fd, "\n %s", buffer);
@@ -858,17 +871,12 @@
isOutput() ? "write" : "read",
mLatencyMs.toString().c_str());
}
-
- if (locked) {
- mLock.unlock();
- }
}
-void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args)
+void AudioFlinger::ThreadBase::dumpEffectChains_l(int fd, const Vector<String16>& args)
{
const size_t SIZE = 256;
char buffer[SIZE];
- String8 result;
size_t numEffectChains = mEffectChains.size();
snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains);
@@ -1819,16 +1827,24 @@
free(mEffectBuffer);
}
-void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args)
+// Thread virtuals
+
+void AudioFlinger::PlaybackThread::onFirstRef()
{
- dumpInternals(fd, args);
- dumpTracks(fd, args);
- dumpEffectChains(fd, args);
- dprintf(fd, " Local log:\n");
- mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
+ run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
}
-void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
+// ThreadBase virtuals
+void AudioFlinger::PlaybackThread::preExit()
+{
+ ALOGV(" preExit()");
+ // FIXME this is using hard-coded strings but in the future, this functionality will be
+ // converted to use audio HAL extensions required to support tunneling
+ status_t result = mOutput->stream->setParameters(String8("exiting=1"));
+ ALOGE_IF(result != OK, "Error when setting parameters on exit: %d", result);
+}
+
+void AudioFlinger::PlaybackThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
{
String8 result;
@@ -1893,10 +1909,8 @@
write(fd, result.string(), result.size());
}
-void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
{
- dumpBase(fd, args);
-
dprintf(fd, " Master mute: %s\n", mMasterMute ? "on" : "off");
if (mHapticChannelMask != AUDIO_CHANNEL_NONE) {
dprintf(fd, " Haptic channel mask: %#x (%s)\n", mHapticChannelMask,
@@ -1927,23 +1941,6 @@
}
}
-// Thread virtuals
-
-void AudioFlinger::PlaybackThread::onFirstRef()
-{
- run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO);
-}
-
-// ThreadBase virtuals
-void AudioFlinger::PlaybackThread::preExit()
-{
- ALOGV(" preExit()");
- // FIXME this is using hard-coded strings but in the future, this functionality will be
- // converted to use audio HAL extensions required to support tunneling
- status_t result = mOutput->stream->setParameters(String8("exiting=1"));
- ALOGE_IF(result != OK, "Error when setting parameters on exit: %d", result);
-}
-
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
@@ -5356,9 +5353,9 @@
}
-void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::MixerThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
- PlaybackThread::dumpInternals(fd, args);
+ PlaybackThread::dumpInternals_l(fd, args);
dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
dprintf(fd, " AudioMixer tracks: %s\n", mAudioMixer->trackNames().c_str());
dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off");
@@ -5432,9 +5429,9 @@
{
}
-void AudioFlinger::DirectOutputThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::DirectOutputThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
- PlaybackThread::dumpInternals(fd, args);
+ PlaybackThread::dumpInternals_l(fd, args);
dprintf(fd, " Master balance: %f Left: %f Right: %f\n",
mMasterBalance.load(), mMasterBalanceLeft, mMasterBalanceRight);
}
@@ -6449,9 +6446,9 @@
}
}
-void AudioFlinger::DuplicatingThread::dumpInternals(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
{
- MixerThread::dumpInternals(fd, args);
+ MixerThread::dumpInternals_l(fd, args);
std::stringstream ss;
const size_t numTracks = mOutputTracks.size();
@@ -7781,19 +7778,8 @@
}
}
-void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::RecordThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
{
- dumpInternals(fd, args);
- dumpTracks(fd, args);
- dumpEffectChains(fd, args);
- dprintf(fd, " Local log:\n");
- mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
-}
-
-void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args)
-{
- dumpBase(fd, args);
-
AudioStreamIn *input = mInput;
audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE;
dprintf(fd, " AudioStreamIn: %p flags %#x (%s)\n",
@@ -7820,7 +7806,7 @@
copy->dump(fd);
}
-void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::RecordThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
{
String8 result;
size_t numtracks = mTracks.size();
@@ -9085,19 +9071,8 @@
}
}
-void AudioFlinger::MmapThread::dump(int fd, const Vector<String16>& args)
+void AudioFlinger::MmapThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
{
- dumpInternals(fd, args);
- dumpTracks(fd, args);
- dumpEffectChains(fd, args);
- dprintf(fd, " Local log:\n");
- mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */);
-}
-
-void AudioFlinger::MmapThread::dumpInternals(int fd, const Vector<String16>& args)
-{
- dumpBase(fd, args);
-
dprintf(fd, " Attributes: content type %d usage %d source %d\n",
mAttr.content_type, mAttr.usage, mAttr.source);
dprintf(fd, " Session: %d port Id: %d\n", mSessionId, mPortId);
@@ -9106,7 +9081,7 @@
}
}
-void AudioFlinger::MmapThread::dumpTracks(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::MmapThread::dumpTracks_l(int fd, const Vector<String16>& args __unused)
{
String8 result;
size_t numtracks = mActiveTracks.size();
@@ -9329,9 +9304,9 @@
}
}
-void AudioFlinger::MmapPlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
+void AudioFlinger::MmapPlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
- MmapThread::dumpInternals(fd, args);
+ MmapThread::dumpInternals_l(fd, args);
dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n",
mStreamType, mStreamVolume, mHalVolFloat, mStreamMute);
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index e5abce7..18cb361 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -43,9 +43,6 @@
virtual status_t readyToRun();
- void dumpBase(int fd, const Vector<String16>& args);
- void dumpEffectChains(int fd, const Vector<String16>& args);
-
void clearPowerManager();
// base for record and playback
@@ -418,7 +415,7 @@
bool isMsdDevice() const { return mIsMsdDevice; }
- virtual void dump(int fd, const Vector<String16>& args) = 0;
+ void dump(int fd, const Vector<String16>& args);
// deliver stats to mediametrics.
void sendStatistics(bool force);
@@ -470,6 +467,11 @@
return INVALID_OPERATION;
}
+ virtual void dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
+ { }
+ virtual void dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
+
+
friend class AudioFlinger; // for mEffectChains
const type_t mType;
@@ -657,6 +659,10 @@
};
SimpleLog mLocalLog;
+
+private:
+ void dumpBase_l(int fd, const Vector<String16>& args);
+ void dumpEffectChains_l(int fd, const Vector<String16>& args);
};
class VolumeInterface {
@@ -709,8 +715,6 @@
audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady);
virtual ~PlaybackThread();
- void dump(int fd, const Vector<String16>& args) override;
-
// Thread virtuals
virtual bool threadLoop();
@@ -760,6 +764,9 @@
mActiveTracks.updatePowerState(this, true /* force */);
}
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
+ void dumpTracks_l(int fd, const Vector<String16>& args) override;
+
public:
virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
@@ -1009,9 +1016,6 @@
void updateMetadata_l() final;
virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata);
- virtual void dumpInternals(int fd, const Vector<String16>& args);
- void dumpTracks(int fd, const Vector<String16>& args);
-
// The Tracks class manages tracks added and removed from the Thread.
template <typename T>
class Tracks {
@@ -1166,7 +1170,6 @@
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
- virtual void dumpInternals(int fd, const Vector<String16>& args);
virtual bool isTrackAllowed_l(
audio_channel_mask_t channelMask, audio_format_t format,
@@ -1185,6 +1188,8 @@
}
}
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
+
// threadLoop snippets
virtual ssize_t threadLoop_write();
virtual void threadLoop_standby();
@@ -1266,8 +1271,6 @@
virtual bool checkForNewParameter_l(const String8& keyValuePair,
status_t& status);
- void dumpInternals(int fd, const Vector<String16>& args) override;
-
virtual void flushHw_l();
void setMasterBalance(float balance) override;
@@ -1278,6 +1281,8 @@
virtual uint32_t suspendSleepTimeUs() const;
virtual void cacheParameters_l();
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
+
// threadLoop snippets
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove);
virtual void threadLoop_mix();
@@ -1397,8 +1402,6 @@
virtual ~DuplicatingThread();
// Thread virtuals
- virtual void dumpInternals(int fd, const Vector<String16>& args) override;
-
void addOutputTrack(MixerThread* thread);
void removeOutputTrack(MixerThread* thread);
uint32_t waitTimeMs() const { return mWaitTimeMs; }
@@ -1407,6 +1410,7 @@
const StreamOutHalInterface::SourceMetadata& metadata) override;
protected:
virtual uint32_t activeSleepTimeUs() const;
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
private:
bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
@@ -1512,9 +1516,6 @@
void destroyTrack_l(const sp<RecordTrack>& track);
void removeTrack_l(const sp<RecordTrack>& track);
- void dumpInternals(int fd, const Vector<String16>& args);
- void dumpTracks(int fd, const Vector<String16>& args);
-
// Thread virtuals
virtual bool threadLoop();
virtual void preExit();
@@ -1551,7 +1552,6 @@
// return true if the caller should then do it's part of the stopping process
bool stop(RecordTrack* recordTrack);
- void dump(int fd, const Vector<String16>& args) override;
AudioStreamIn* clearInput();
virtual sp<StreamHalInterface> stream() const;
@@ -1619,6 +1619,11 @@
return audio_is_input_device(
mInDevice & mTimestampCorrectedDevices);
}
+
+protected:
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
+ void dumpTracks_l(int fd, const Vector<String16>& args) override;
+
private:
// Enter standby if not already in standby, and set mStandby flag
void standbyIfNotAlreadyInStandby();
@@ -1768,11 +1773,9 @@
// Sets the UID records silence
virtual void setRecordSilenced(uid_t uid __unused, bool silenced __unused) {}
- void dump(int fd, const Vector<String16>& args) override;
- virtual void dumpInternals(int fd, const Vector<String16>& args);
- void dumpTracks(int fd, const Vector<String16>& args);
-
protected:
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
+ void dumpTracks_l(int fd, const Vector<String16>& args) override;
audio_attributes_t mAttr;
audio_session_t mSessionId;
@@ -1822,8 +1825,6 @@
virtual void checkSilentMode_l();
void processVolume_l() override;
- virtual void dumpInternals(int fd, const Vector<String16>& args);
-
virtual bool isOutput() const override { return true; }
void updateMetadata_l() override;
@@ -1831,6 +1832,7 @@
virtual void toAudioPortConfig(struct audio_port_config *config);
protected:
+ void dumpInternals_l(int fd, const Vector<String16>& args) override;
audio_stream_type_t mStreamType;
float mMasterVolume;
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index acbfc9e..a2cf7aa 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -198,6 +198,7 @@
//dump state
virtual status_t dump(int fd) = 0;
+ virtual status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t flags) = 0;
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo) = 0;
virtual bool isDirectOutputSupported(const audio_config_base_t& config,
const audio_attributes_t& attributes) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index cc43fe6..33e506f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -135,11 +135,18 @@
/**
* @brief filter the devices supported by this collection against another collection
* @param devices to filter against
- * @return
+ * @return a filtered DeviceVector
*/
DeviceVector filter(const DeviceVector &devices) const;
/**
+ * @brief filter the devices supported by this collection before sending
+ * then to the Engine via AudioPolicyManagerObserver interface
+ * @return a filtered DeviceVector
+ */
+ DeviceVector filterForEngine() const;
+
+ /**
* @brief merge two vectors. As SortedVector Implementation is buggy (it does not check the size
* of the destination vector, only of the source, it provides a safe implementation
* @param devices source device vector to merge with
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index f7289ca..1b812c0 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -47,32 +47,29 @@
int indexCriterion = 0;
for (const auto &criterion : mCriteria) {
- dst->appendFormat("%*s- Criterion %d:\n", spaces + 2, "", indexCriterion++);
+ dst->appendFormat("%*s- Criterion %d: ", spaces + 2, "", indexCriterion++);
- std::string usageLiteral;
- if (!UsageTypeConverter::toString(criterion.mValue.mUsage, usageLiteral)) {
- ALOGE("%s: failed to convert usage %d", __FUNCTION__, criterion.mValue.mUsage);
- return;
+ std::string ruleType, ruleValue;
+ bool unknownRule = !RuleTypeConverter::toString(criterion.mRule, ruleType);
+ switch (criterion.mRule & ~RULE_EXCLUSION_MASK) { // no need to match RULE_EXCLUDE_...
+ case RULE_MATCH_ATTRIBUTE_USAGE:
+ UsageTypeConverter::toString(criterion.mValue.mUsage, ruleValue);
+ break;
+ case RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET:
+ SourceTypeConverter::toString(criterion.mValue.mSource, ruleValue);
+ break;
+ case RULE_MATCH_UID:
+ ruleValue = std::to_string(criterion.mValue.mUid);
+ break;
+ default:
+ unknownRule = true;
}
- dst->appendFormat("%*s- Usage:%s\n", spaces + 4, "", usageLiteral.c_str());
- if (mMixType == MIX_TYPE_RECORDERS) {
- std::string sourceLiteral;
- if (!SourceTypeConverter::toString(criterion.mValue.mSource, sourceLiteral)) {
- ALOGE("%s: failed to convert source %d", __FUNCTION__, criterion.mValue.mSource);
- return;
- }
- dst->appendFormat("%*s- Source:%s\n", spaces + 4, "", sourceLiteral.c_str());
-
+ if (!unknownRule) {
+ dst->appendFormat("%s %s\n", ruleType.c_str(), ruleValue.c_str());
+ } else {
+ dst->appendFormat("Unknown rule type value 0x%x\n", criterion.mRule);
}
- dst->appendFormat("%*s- Uid:%d\n", spaces + 4, "", criterion.mValue.mUid);
-
- std::string ruleLiteral;
- if (!RuleTypeConverter::toString(criterion.mRule, ruleLiteral)) {
- ALOGE("%s: failed to convert source %d", __FUNCTION__,criterion.mRule);
- return;
- }
- dst->appendFormat("%*s- Rule:%s\n", spaces + 4, "", ruleLiteral.c_str());
}
}
@@ -150,7 +147,10 @@
}
switch (mixMatch(policyMix.get(), i, attributes, uid)) {
- case MixMatchStatus::INVALID_MIX: return BAD_VALUE; // TODO: Do we really want to abort?
+ case MixMatchStatus::INVALID_MIX:
+ // The mix has contradictory rules, ignore it
+ // TODO: reject invalid mix at registration
+ continue;
case MixMatchStatus::NO_MATCH:
ALOGV("%s: Mix %zu: does not match", __func__, i);
continue; // skip the mix
@@ -180,7 +180,12 @@
// Loopback render mixes are created from a public API and thus restricted
// to non sensible audio that have not opted out.
if (is_mix_loopback_render(mix->mRouteFlags)) {
- if ((attributes.flags & AUDIO_FLAG_NO_CAPTURE) == AUDIO_FLAG_NO_CAPTURE) {
+ auto hasFlag = [](auto flags, auto flag) { return (flags & flag) == flag; };
+ if (hasFlag(attributes.flags, AUDIO_FLAG_NO_SYSTEM_CAPTURE)) {
+ return MixMatchStatus::NO_MATCH;
+ }
+ if (!mix->mAllowPrivilegedPlaybackCapture &&
+ hasFlag(attributes.flags, AUDIO_FLAG_NO_MEDIA_PROJECTION)) {
return MixMatchStatus::NO_MATCH;
}
if (!(attributes.usage == AUDIO_USAGE_UNKNOWN ||
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 91961d0..ecd5b34 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -40,7 +40,7 @@
mTagName(tagName), mDeviceType(type), mEncodedFormats(encodedFormats)
{
mCurrentEncodedFormat = AUDIO_FORMAT_DEFAULT;
- if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+ if (audio_is_remote_submix_device(type)) {
mAddress = String8("0");
}
/* If framework runs against a pre 5.0 Audio HAL, encoded formats are absent from the config.
@@ -399,6 +399,18 @@
return filter(devices).size() == devices.size();
}
+DeviceVector DeviceVector::filterForEngine() const
+{
+ DeviceVector filteredDevices;
+ for (const auto &device : *this) {
+ if (audio_is_remote_submix_device(device->type()) && device->address() != "0") {
+ continue;
+ }
+ filteredDevices.add(device);
+ }
+ return filteredDevices;
+}
+
void DeviceDescriptor::log() const
{
ALOGI("Device id:%d type:0x%08X:%s, addr:%s", mId, mDeviceType,
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index ec7ff57..96a8337 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -398,6 +398,13 @@
if (!moduleDevices.contains(device)) {
continue;
}
+
+ // removal of remote submix devices associated with a dynamic policy is
+ // handled by removeOutputProfile() and removeInputProfile()
+ if (audio_is_remote_submix_device(device->type()) && device->address() != "0") {
+ continue;
+ }
+
device->detach();
// Only remove from dynamic list, not from declared list!!!
if (!hwModule->getDynamicDevices().contains(device)) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp b/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp
index 7c76d8a..2b5455e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/TypeConverter.cpp
@@ -52,7 +52,6 @@
template <>
const RuleTypeConverter::Table RuleTypeConverter::mTable[] = {
- MAKE_STRING_FROM_ENUM(RULE_EXCLUSION_MASK),
MAKE_STRING_FROM_ENUM(RULE_MATCH_ATTRIBUTE_USAGE),
MAKE_STRING_FROM_ENUM(RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET),
MAKE_STRING_FROM_ENUM(RULE_MATCH_UID),
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
index 43ba625..ebd82a7 100644
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
@@ -46,9 +46,9 @@
virtual const AudioInputCollection &getInputs() const = 0;
- virtual const DeviceVector &getAvailableOutputDevices() const = 0;
+ virtual const DeviceVector getAvailableOutputDevices() const = 0;
- virtual const DeviceVector &getAvailableInputDevices() const = 0;
+ virtual const DeviceVector getAvailableInputDevices() const = 0;
virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const = 0;
diff --git a/services/audiopolicy/engineconfigurable/src/Engine.cpp b/services/audiopolicy/engineconfigurable/src/Engine.cpp
index 89aaa84..e59d983 100644
--- a/services/audiopolicy/engineconfigurable/src/Engine.cpp
+++ b/services/audiopolicy/engineconfigurable/src/Engine.cpp
@@ -207,7 +207,7 @@
ALOGE("%s: Trying to get device on invalid strategy %d", __FUNCTION__, ps);
return {};
}
- const DeviceVector &availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
+ const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
uint32_t availableOutputDevicesType = availableOutputDevices.types();
@@ -272,7 +272,7 @@
return DeviceVector(preferredDevice);
}
product_strategy_t strategy = getProductStrategyForAttributes(attributes);
- const DeviceVector &availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
+ const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
//
// @TODO: what is the priority of explicit routing? Shall it be considered first as it used to
@@ -298,7 +298,7 @@
sp<AudioPolicyMix> *mix) const
{
const auto &policyMixes = getApmObserver()->getAudioPolicyMixCollection();
- const auto &availableInputDevices = getApmObserver()->getAvailableInputDevices();
+ const auto availableInputDevices = getApmObserver()->getAvailableInputDevices();
const auto &inputs = getApmObserver()->getInputs();
std::string address;
//
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 66a6965..592a0b9 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -502,8 +502,8 @@
audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) const
{
- const DeviceVector &availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
- const DeviceVector &availableInputDevices = getApmObserver()->getAvailableInputDevices();
+ const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
+ const DeviceVector availableInputDevices = getApmObserver()->getAvailableInputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
@@ -704,7 +704,7 @@
return DeviceVector(preferredDevice);
}
product_strategy_t strategy = getProductStrategyForAttributes(attributes);
- const DeviceVector &availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
+ const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
//
// @TODO: what is the priority of explicit routing? Shall it be considered first as it used to
@@ -730,7 +730,7 @@
sp<AudioPolicyMix> *mix) const
{
const auto &policyMixes = getApmObserver()->getAudioPolicyMixCollection();
- const auto &availableInputDevices = getApmObserver()->getAvailableInputDevices();
+ const auto availableInputDevices = getApmObserver()->getAvailableInputDevices();
const auto &inputs = getApmObserver()->getInputs();
std::string address;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 651017d..61c47a1 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -169,8 +169,6 @@
broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
return INVALID_OPERATION;
}
- // Propagate device availability to Engine
- mEngine->setDeviceConnectionState(device, state);
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
@@ -200,8 +198,6 @@
// Reset active device codec
device->setEncodedFormat(AUDIO_FORMAT_DEFAULT);
- // Propagate device availability to Engine
- mEngine->setDeviceConnectionState(device, state);
} break;
default:
@@ -209,6 +205,9 @@
return BAD_VALUE;
}
+ // Propagate device availability to Engine
+ setEngineDeviceConnectionState(device, state);
+
// No need to evaluate playback routing when connecting a remote submix
// output device used by a dynamic policy of type recorder as no
// playback use case is affected.
@@ -318,9 +317,6 @@
if (mAvailableInputDevices.add(device) < 0) {
return NO_MEMORY;
}
-
- // Propagate device availability to Engine
- mEngine->setDeviceConnectionState(device, state);
} break;
// handle input device disconnection
@@ -337,9 +333,6 @@
checkInputsForDevice(device, state, inputs);
mAvailableInputDevices.remove(device);
-
- // Propagate device availability to Engine
- mEngine->setDeviceConnectionState(device, state);
} break;
default:
@@ -347,6 +340,9 @@
return BAD_VALUE;
}
+ // Propagate device availability to Engine
+ setEngineDeviceConnectionState(device, state);
+
closeAllInputs();
// As the input device list can impact the output device selection, update
// getDeviceForStrategy() cache
@@ -369,6 +365,17 @@
return BAD_VALUE;
}
+void AudioPolicyManager::setEngineDeviceConnectionState(const sp<DeviceDescriptor> device,
+ audio_policy_dev_state_t state) {
+
+ // the Engine does not have to know about remote submix devices used by dynamic audio policies
+ if (audio_is_remote_submix_device(device->type()) && device->address() != "0") {
+ return;
+ }
+ mEngine->setDeviceConnectionState(device, state);
+}
+
+
audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device,
const char *device_address)
{
@@ -416,8 +423,7 @@
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
// Check if the device is currently connected
- DeviceVector availableDevices = getAvailableOutputDevices();
- DeviceVector deviceList = availableDevices.getDevicesFromTypeMask(device);
+ DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromTypeMask(device);
if (deviceList.empty()) {
// Nothing to do: device is not connected
return NO_ERROR;
@@ -945,6 +951,9 @@
if (status != NO_ERROR) {
return status;
}
+ if (auto it = mAllowedCapturePolicies.find(uid); it != end(mAllowedCapturePolicies)) {
+ resultAttr->flags |= it->second;
+ }
*stream = mEngine->getStreamTypeForAttributes(*resultAttr);
ALOGV("%s() attributes=%s stream=%s session %d selectedDeviceId %d", __func__,
@@ -3022,6 +3031,11 @@
mPolicyMixes.dump(dst);
mAudioSources.dump(dst);
+ dst->appendFormat(" AllowedCapturePolicies:\n");
+ for (auto& policy : mAllowedCapturePolicies) {
+ dst->appendFormat(" - uid=%d flag_mask=%#x\n", policy.first, policy.second);
+ }
+
dst->appendFormat("\nPolicy Engine dump:\n");
mEngine->dump(dst);
}
@@ -3034,6 +3048,12 @@
return NO_ERROR;
}
+status_t AudioPolicyManager::setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy)
+{
+ mAllowedCapturePolicies[uid] = capturePolicy;
+ return NO_ERROR;
+}
+
// This function checks for the parameters which can be offloaded.
// This can be enhanced depending on the capability of the DSP and policy
// of the system.
@@ -4385,7 +4405,7 @@
continue;
}
// Device is now validated and can be appended to the available devices of the engine
- mEngine->setDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+ setEngineDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
i++;
}
};
@@ -4399,7 +4419,7 @@
status = NO_INIT;
}
// If microphones address is empty, set it according to device type
- for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
+ for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
if (mAvailableInputDevices[i]->address().isEmpty()) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BUILTIN_MIC) {
mAvailableInputDevices[i]->setAddress(String8(AUDIO_BOTTOM_MICROPHONE_ADDRESS));
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 1c98684..a700868 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -218,6 +218,7 @@
status_t dump(int fd) override;
+ status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy) override;
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
virtual bool isDirectOutputSupported(const audio_config_base_t& config,
@@ -341,13 +342,13 @@
{
return mInputs;
}
- virtual const DeviceVector &getAvailableOutputDevices() const
+ virtual const DeviceVector getAvailableOutputDevices() const
{
- return mAvailableOutputDevices;
+ return mAvailableOutputDevices.filterForEngine();
}
- virtual const DeviceVector &getAvailableInputDevices() const
+ virtual const DeviceVector getAvailableInputDevices() const
{
- return mAvailableInputDevices;
+ return mAvailableInputDevices.filterForEngine();
}
virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const
{
@@ -754,6 +755,8 @@
// Surround formats that are enabled manually. Taken into account when
// "encoded surround" is forced into "manual" mode.
std::unordered_set<audio_format_t> mManualSurroundFormats;
+
+ std::unordered_map<uid_t, audio_flags_mask_t> mAllowedCapturePolicies;
private:
// Add or remove AC3 DTS encodings based on user preferences.
void modifySurroundFormats(const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr);
@@ -844,6 +847,10 @@
const char *device_address,
const char *device_name,
audio_format_t encodedFormat);
+
+ void setEngineDeviceConnectionState(const sp<DeviceDescriptor> device,
+ audio_policy_dev_state_t state);
+
void updateMono(audio_io_handle_t output) {
AudioParameter param;
param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 93e3c44..b8036bb 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -192,7 +192,7 @@
}
audio_attributes_t attr = *originalAttr;
if (!mPackageManager.allowPlaybackCapture(uid)) {
- attr.flags |= AUDIO_FLAG_NO_CAPTURE;
+ attr.flags |= AUDIO_FLAG_NO_MEDIA_PROJECTION;
}
audio_output_flags_t originalFlags = flags;
AutoCallerClear acc;
@@ -322,7 +322,7 @@
return;
}
sp<AudioPlaybackClient> client = mAudioPlaybackClients.valueAt(index);
- mAudioRecordClients.removeItem(portId);
+ mAudioPlaybackClients.removeItem(portId);
// called from internal thread: no need to clear caller identity
mAudioPolicyManager->releaseOutput(portId);
@@ -385,7 +385,7 @@
return PERMISSION_DENIED;
}
- bool canCaptureHotword = captureHotwordAllowed(pid, uid);
+ bool canCaptureHotword = captureHotwordAllowed(opPackageName, pid, uid);
if ((attr->source == AUDIO_SOURCE_HOTWORD) && !canCaptureHotword) {
return BAD_VALUE;
}
@@ -948,6 +948,20 @@
return audioPolicyEffects->removeStreamDefaultEffect(id);
}
+status_t AudioPolicyService::setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy) {
+ Mutex::Autolock _l(mLock);
+ if (mAudioPolicyManager == NULL) {
+ ALOGV("%s() mAudioPolicyManager == NULL", __func__);
+ return NO_INIT;
+ }
+ uint_t callingUid = IPCThreadState::self()->getCallingUid();
+ if (uid != callingUid) {
+ ALOGD("%s() uid invalid %d != %d", __func__, uid, callingUid);
+ return PERMISSION_DENIED;
+ }
+ return mAudioPolicyManager->setAllowedCapturePolicy(uid, capturePolicy);
+}
+
bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
{
if (mAudioPolicyManager == NULL) {
@@ -1083,6 +1097,14 @@
return PERMISSION_DENIED;
}
+ bool needCaptureMediaOutput = std::any_of(mixes.begin(), mixes.end(), [](auto& mix) {
+ return mix.mAllowPrivilegedPlaybackCapture; });
+ const uid_t callingUid = IPCThreadState::self()->getCallingUid();
+ const pid_t callingPid = IPCThreadState::self()->getCallingPid();
+ if (needCaptureMediaOutput && !captureMediaOutputAllowed(callingPid, callingUid)) {
+ return PERMISSION_DENIED;
+ }
+
if (mAudioPolicyManager == NULL) {
return NO_INIT;
}
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index e858e8d..f63fa81 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -22,8 +22,9 @@
#define __STDINT_LIMITS
#define __STDC_LIMIT_MACROS
#include <stdint.h>
-
#include <sys/time.h>
+
+#include <audio_utils/clock.h>
#include <binder/IServiceManager.h>
#include <utils/Log.h>
#include <cutils/properties.h>
@@ -48,8 +49,7 @@
static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n";
static const char kCmdDeadlockedString[] = "AudioPolicyService command thread may be deadlocked\n";
-static const int kDumpLockRetries = 50;
-static const int kDumpLockSleepUs = 20000;
+static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND;
static const nsecs_t kAudioCommandTimeoutNs = seconds(3); // 3 seconds
@@ -376,17 +376,10 @@
IPCThreadState::self()->getCallingPid());
}
-static bool tryLock(Mutex& mutex)
+static bool dumpTryLock(Mutex& mutex)
{
- bool locked = false;
- for (int i = 0; i < kDumpLockRetries; ++i) {
- if (mutex.tryLock() == NO_ERROR) {
- locked = true;
- break;
- }
- usleep(kDumpLockSleepUs);
- }
- return locked;
+ status_t err = mutex.timedLock(kDumpLockTimeoutNs);
+ return err == NO_ERROR;
}
status_t AudioPolicyService::dumpInternals(int fd)
@@ -627,7 +620,7 @@
if (!dumpAllowed()) {
dumpPermissionDenial(fd);
} else {
- bool locked = tryLock(mLock);
+ bool locked = dumpTryLock(mLock);
if (!locked) {
String8 result(kDeadlockedString);
write(fd, result.string(), result.size());
@@ -1260,7 +1253,7 @@
result.append(buffer);
write(fd, result.string(), result.size());
- bool locked = tryLock(mLock);
+ bool locked = dumpTryLock(mLock);
if (!locked) {
String8 result2(kCmdDeadlockedString);
write(fd, result2.string(), result2.size());
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 160f70f..efdba56 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -182,6 +182,7 @@
audio_io_handle_t output,
int delayMs = 0);
virtual status_t setVoiceVolume(float volume, int delayMs = 0);
+ status_t setAllowedCapturePolicy(uint_t uid, audio_flags_mask_t capturePolicy) override;
virtual bool isOffloadSupported(const audio_offload_info_t &config);
virtual bool isDirectOutputSupported(const audio_config_base_t& config,
const audio_attributes_t& attributes);
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 51d0682..8113c3f 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -972,8 +972,9 @@
userid_t clientUserId = multiuser_get_user_id(clientUid);
// Only allow clients who are being used by the current foreground device user, unless calling
- // from our own process.
- if (callingPid != getpid() && (mAllowedUsers.find(clientUserId) == mAllowedUsers.end())) {
+ // from our own process OR the caller is using the cameraserver's HIDL interface.
+ if (!hardware::IPCThreadState::self()->isServingCall() && callingPid != getpid() &&
+ (mAllowedUsers.find(clientUserId) == mAllowedUsers.end())) {
ALOGE("CameraService::connect X (PID %d) rejected (cannot connect from "
"device user %d, currently allowed device users: %s)", callingPid, clientUserId,
toString(mAllowedUsers).string());
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 22e09e4..bc0dafe 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -1062,14 +1062,18 @@
nsecs_t waitDuration = kBaseGetBufferWait + getExpectedInFlightDuration();
status_t res = outputStream->getBuffer(&sb, waitDuration);
if (res != OK) {
- ALOGE("%s: Can't get output buffer for stream %d: %s (%d)",
- __FUNCTION__, streamId, strerror(-res), res);
if (res == NO_INIT || res == DEAD_OBJECT) {
+ ALOGV("%s: Can't get output buffer for stream %d: %s (%d)",
+ __FUNCTION__, streamId, strerror(-res), res);
bufRet.val.error(StreamBufferRequestError::STREAM_DISCONNECTED);
- } else if (res == TIMED_OUT || res == NO_MEMORY) {
- bufRet.val.error(StreamBufferRequestError::NO_BUFFER_AVAILABLE);
} else {
- bufRet.val.error(StreamBufferRequestError::UNKNOWN_ERROR);
+ ALOGE("%s: Can't get output buffer for stream %d: %s (%d)",
+ __FUNCTION__, streamId, strerror(-res), res);
+ if (res == TIMED_OUT || res == NO_MEMORY) {
+ bufRet.val.error(StreamBufferRequestError::NO_BUFFER_AVAILABLE);
+ } else {
+ bufRet.val.error(StreamBufferRequestError::UNKNOWN_ERROR);
+ }
}
currentReqSucceeds = false;
break;
@@ -3154,9 +3158,10 @@
// Note: stream may be deallocated at this point, if this buffer was
// the last reference to it.
- if (res != OK) {
- ALOGE("Can't return buffer to its stream: %s (%d)",
- strerror(-res), res);
+ if (res == NO_INIT || res == DEAD_OBJECT) {
+ ALOGV("Can't return buffer to its stream: %s (%d)", strerror(-res), res);
+ } else if (res != OK) {
+ ALOGE("Can't return buffer to its stream: %s (%d)", strerror(-res), res);
}
// Long processing consumers can cause returnBuffer timeout for shared stream
@@ -3777,10 +3782,12 @@
hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_INVALID_ERROR;
int streamId = 0;
+ String16 physicalCameraId;
if (msg.error_stream != NULL) {
Camera3Stream *stream =
Camera3Stream::cast(msg.error_stream);
streamId = stream->getId();
+ physicalCameraId = String16(stream->physicalCameraId());
}
ALOGV("Camera %s: %s: HAL error, frame %d, stream %d: %d",
mId.string(), __FUNCTION__, msg.frame_number,
@@ -3802,13 +3809,29 @@
InFlightRequest &r = mInFlightMap.editValueAt(idx);
r.requestStatus = msg.error_code;
resultExtras = r.resultExtras;
- if (hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_RESULT == errorCode
+ bool logicalDeviceResultError = false;
+ if (hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_RESULT ==
+ errorCode) {
+ if (physicalCameraId.size() > 0) {
+ String8 cameraId(physicalCameraId);
+ if (r.physicalCameraIds.find(cameraId) == r.physicalCameraIds.end()) {
+ ALOGE("%s: Reported result failure for physical camera device: %s "
+ " which is not part of the respective request!",
+ __FUNCTION__, cameraId.string());
+ break;
+ }
+ resultExtras.errorPhysicalCameraId = physicalCameraId;
+ } else {
+ logicalDeviceResultError = true;
+ }
+ }
+
+ if (logicalDeviceResultError
|| hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_REQUEST ==
errorCode) {
r.skipResultMetadata = true;
}
- if (hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_RESULT ==
- errorCode) {
+ if (logicalDeviceResultError) {
// In case of missing result check whether the buffers
// returned. If they returned, then remove inflight
// request.
@@ -5580,7 +5603,7 @@
if (mUseHalBufManager) {
if (outputStream->isAbandoned()) {
- ALOGE("%s: stream %d is abandoned.", __FUNCTION__, streamId);
+ ALOGV("%s: stream %d is abandoned, skipping request", __FUNCTION__, streamId);
return TIMED_OUT;
}
// HAL will request buffer through requestStreamBuffer API
@@ -5598,7 +5621,7 @@
// Can't get output buffer from gralloc queue - this could be due to
// abandoned queue or other consumer misbehavior, so not a fatal
// error
- ALOGE("RequestThread: Can't get output buffer, skipping request:"
+ ALOGV("RequestThread: Can't get output buffer, skipping request:"
" %s (%d)", strerror(-res), res);
return TIMED_OUT;
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index baba856..1c77581 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -233,6 +233,7 @@
* queueBuffer
*/
sp<ANativeWindow> currentConsumer = mConsumer;
+ StreamState state = mState;
mLock.unlock();
ANativeWindowBuffer *anwBuffer = container_of(buffer.buffer, ANativeWindowBuffer, handle);
@@ -244,7 +245,7 @@
if (mDropBuffers) {
ALOGV("%s: Dropping a frame for stream %d.", __FUNCTION__, mId);
} else if (buffer.status == CAMERA3_BUFFER_STATUS_ERROR) {
- ALOGW("%s: A frame is dropped for stream %d due to buffer error.", __FUNCTION__, mId);
+ ALOGV("%s: A frame is dropped for stream %d due to buffer error.", __FUNCTION__, mId);
} else {
ALOGE("%s: Stream %d: timestamp shouldn't be 0", __FUNCTION__, mId);
}
@@ -252,7 +253,7 @@
res = currentConsumer->cancelBuffer(currentConsumer.get(),
anwBuffer,
anwReleaseFence);
- if (res != OK) {
+ if (shouldLogError(res, state)) {
ALOGE("%s: Stream %d: Error cancelling buffer to native window:"
" %s (%d)", __FUNCTION__, mId, strerror(-res), res);
}
@@ -284,9 +285,9 @@
}
res = queueBufferToConsumer(currentConsumer, anwBuffer, anwReleaseFence, surface_ids);
- if (res != OK) {
- ALOGE("%s: Stream %d: Error queueing buffer to native window: "
- "%s (%d)", __FUNCTION__, mId, strerror(-res), res);
+ if (shouldLogError(res, state)) {
+ ALOGE("%s: Stream %d: Error queueing buffer to native window:"
+ " %s (%d)", __FUNCTION__, mId, strerror(-res), res);
}
}
mLock.lock();
@@ -534,10 +535,11 @@
// successful return.
*anb = gb.get();
res = mConsumer->attachBuffer(*anb);
- if (res != OK) {
+ if (shouldLogError(res, mState)) {
ALOGE("%s: Stream %d: Can't attach the output buffer to this surface: %s (%d)",
__FUNCTION__, mId, strerror(-res), res);
-
+ }
+ if (res != OK) {
checkRetAndSetAbandonedLocked(res);
return res;
}
@@ -592,9 +594,10 @@
ALOGV("Stream %d: Attached new buffer", getId());
if (res != OK) {
- ALOGE("%s: Stream %d: Can't attach the output buffer to this surface: %s (%d)",
- __FUNCTION__, mId, strerror(-res), res);
-
+ if (shouldLogError(res, mState)) {
+ ALOGE("%s: Stream %d: Can't attach the output buffer to this surface:"
+ " %s (%d)", __FUNCTION__, mId, strerror(-res), res);
+ }
checkRetAndSetAbandonedLocked(res);
return res;
}
@@ -604,9 +607,10 @@
return res;
}
} else if (res != OK) {
- ALOGE("%s: Stream %d: Can't dequeue next output buffer: %s (%d)",
- __FUNCTION__, mId, strerror(-res), res);
-
+ if (shouldLogError(res, mState)) {
+ ALOGE("%s: Stream %d: Can't dequeue next output buffer: %s (%d)",
+ __FUNCTION__, mId, strerror(-res), res);
+ }
checkRetAndSetAbandonedLocked(res);
return res;
}
@@ -639,6 +643,16 @@
}
}
+bool Camera3OutputStream::shouldLogError(status_t res, StreamState state) {
+ if (res == OK) {
+ return false;
+ }
+ if ((res == DEAD_OBJECT || res == NO_INIT) && state == STATE_ABANDONED) {
+ return false;
+ }
+ return true;
+}
+
status_t Camera3OutputStream::disconnectLocked() {
status_t res;
@@ -838,7 +852,9 @@
ALOGW("%s: the released buffer has already been freed by the buffer queue!", __FUNCTION__);
} else if (res != OK) {
// Treat other errors as abandonment
- ALOGE("%s: detach next buffer failed: %s (%d).", __FUNCTION__, strerror(-res), res);
+ if (shouldLogError(res, mState)) {
+ ALOGE("%s: detach next buffer failed: %s (%d).", __FUNCTION__, strerror(-res), res);
+ }
mState = STATE_ABANDONED;
return res;
}
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.h b/services/camera/libcameraservice/device3/Camera3OutputStream.h
index 30fc2f7..729c655 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.h
@@ -317,6 +317,10 @@
// Check return status of IGBP calls and set abandoned state accordingly
void checkRetAndSetAbandonedLocked(status_t res);
+ // If the status indicates abandonded stream, only log when state hasn't been updated to
+ // STATE_ABANDONED
+ static bool shouldLogError(status_t res, StreamState state);
+
static const int32_t kDequeueLatencyBinSize = 5; // in ms
CameraLatencyHistogram mDequeueBufferLatency;
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index 5eb6a23..3d21029 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -458,7 +458,7 @@
// Zero for formats with fixed buffer size for given dimensions.
const size_t mMaxSize;
- enum {
+ enum StreamState {
STATE_ERROR,
STATE_CONSTRUCTED,
STATE_IN_CONFIG,
diff --git a/services/camera/libcameraservice/hidl/Convert.cpp b/services/camera/libcameraservice/hidl/Convert.cpp
index c2ed23a..866c3b5 100644
--- a/services/camera/libcameraservice/hidl/Convert.cpp
+++ b/services/camera/libcameraservice/hidl/Convert.cpp
@@ -157,6 +157,8 @@
hCaptureResultExtras.frameNumber = captureResultExtras.frameNumber;
hCaptureResultExtras.partialResultCount = captureResultExtras.partialResultCount;
hCaptureResultExtras.errorStreamId = captureResultExtras.errorStreamId;
+ hCaptureResultExtras.errorPhysicalCameraId = hidl_string(String8(
+ captureResultExtras.errorPhysicalCameraId).string());
return hCaptureResultExtras;
}
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy b/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
index 78ecaf5..4faf8b2 100644
--- a/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
@@ -28,7 +28,6 @@
fstatfs: 1
memfd_create: 1
ftruncate: 1
-ftruncate64: 1
# mremap: Ensure |flags| are (MREMAP_MAYMOVE | MREMAP_FIXED) TODO: Once minijail
# parser support for '<' is in this needs to be modified to also prevent
@@ -62,3 +61,21 @@
getrandom: 1
madvise: 1
+# crash dump policy additions
+clock_gettime: 1
+getpid: 1
+gettid: 1
+pipe2: 1
+recvmsg: 1
+process_vm_readv: 1
+tgkill: 1
+rt_sigaction: 1
+rt_tgsigqueueinfo: 1
+#mprotect: arg2 in 0x1|0x2
+munmap: 1
+#mmap: arg2 in 0x1|0x2
+geteuid: 1
+getgid: 1
+getegid: 1
+getgroups: 1
+
diff --git a/services/mediaextractor/mediaextractor.rc b/services/mediaextractor/mediaextractor.rc
index 6b2d0a5..5fc2941 100644
--- a/services/mediaextractor/mediaextractor.rc
+++ b/services/mediaextractor/mediaextractor.rc
@@ -2,7 +2,5 @@
class main
user mediaex
group drmrpc mediadrm
- # TODO(b/123275379): Remove updatable when http://aosp/878198 has landed
- updatable
ioprio rt 4
writepid /dev/cpuset/foreground/tasks
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 6c28083..fbf7d10 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -93,12 +93,14 @@
const audio_source_t source = (direction == AAUDIO_DIRECTION_INPUT)
? AAudioConvert_inputPresetToAudioSource(getInputPreset())
: AUDIO_SOURCE_DEFAULT;
+ const audio_flags_mask_t flags = AUDIO_FLAG_LOW_LATENCY |
+ AAudioConvert_allowCapturePolicyToAudioFlagsMask(getAllowedCapturePolicy());
const audio_attributes_t attributes = {
.content_type = contentType,
.usage = usage,
.source = source,
- .flags = AUDIO_FLAG_LOW_LATENCY,
+ .flags = flags,
.tags = ""
};
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index b16b5dc..880a3d7 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -343,6 +343,20 @@
return writeUpMessageQueue(&command);
}
+bool AAudioServiceStreamBase::isUpMessageQueueBusy() {
+ std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
+ if (mUpMessageQueue == nullptr) {
+ ALOGE("%s(): mUpMessageQueue null! - stream not open", __func__);
+ return true;
+ }
+ int32_t framesAvailable = mUpMessageQueue->getFifoBuffer()
+ ->getFullFramesAvailable();
+ int32_t capacity = mUpMessageQueue->getFifoBuffer()
+ ->getBufferCapacityInFrames();
+ // Is it half full or more
+ return framesAvailable >= (capacity / 2);
+}
+
aaudio_result_t AAudioServiceStreamBase::writeUpMessageQueue(AAudioServiceMessage *command) {
std::lock_guard<std::mutex> lock(mUpMessageQueueLock);
if (mUpMessageQueue == nullptr) {
@@ -366,6 +380,13 @@
aaudio_result_t AAudioServiceStreamBase::sendCurrentTimestamp() {
AAudioServiceMessage command;
+ // It is not worth filling up the queue with timestamps.
+ // That can cause the stream to get suspended.
+ // So just drop the timestamp if the queue is getting full.
+ if (isUpMessageQueueBusy()) {
+ return AAUDIO_OK;
+ }
+
// Send a timestamp for the clock model.
aaudio_result_t result = getFreeRunningPosition(&command.timestamp.position,
&command.timestamp.timestamp);
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index ffc768b..097bc64 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -312,6 +312,12 @@
android::wp<AAudioServiceEndpoint> mServiceEndpointWeak;
private:
+
+ /**
+ * @return true if the queue is getting full.
+ */
+ bool isUpMessageQueueBusy();
+
aaudio_handle_t mHandle = -1;
bool mFlowing = false;
diff --git a/services/soundtrigger/Android.bp b/services/soundtrigger/Android.bp
index 1f2283a..3f02f48 100644
--- a/services/soundtrigger/Android.bp
+++ b/services/soundtrigger/Android.bp
@@ -28,6 +28,7 @@
"libhardware",
"libsoundtrigger",
"libaudioclient",
+ "libaudioutils",
"libmediautils",
"libhwbinder",
diff --git a/services/soundtrigger/SoundTriggerHwService.cpp b/services/soundtrigger/SoundTriggerHwService.cpp
index fe2ccf2..377d30b 100644
--- a/services/soundtrigger/SoundTriggerHwService.cpp
+++ b/services/soundtrigger/SoundTriggerHwService.cpp
@@ -22,6 +22,7 @@
#include <sys/types.h>
#include <pthread.h>
+#include <audio_utils/clock.h>
#include <system/sound_trigger.h>
#include <cutils/atomic.h>
#include <cutils/properties.h>
@@ -81,11 +82,13 @@
}
}
-status_t SoundTriggerHwService::listModules(struct sound_trigger_module_descriptor *modules,
+status_t SoundTriggerHwService::listModules(const String16& opPackageName,
+ struct sound_trigger_module_descriptor *modules,
uint32_t *numModules)
{
ALOGV("listModules");
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(opPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
@@ -102,12 +105,14 @@
return NO_ERROR;
}
-status_t SoundTriggerHwService::attach(const sound_trigger_module_handle_t handle,
+status_t SoundTriggerHwService::attach(const String16& opPackageName,
+ const sound_trigger_module_handle_t handle,
const sp<ISoundTriggerClient>& client,
sp<ISoundTrigger>& moduleInterface)
{
ALOGV("attach module %d", handle);
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(opPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
@@ -123,7 +128,7 @@
}
sp<Module> module = mModules.valueAt(index);
- sp<ModuleClient> moduleClient = module->addClient(client);
+ sp<ModuleClient> moduleClient = module->addClient(client, opPackageName);
if (moduleClient == 0) {
return NO_INIT;
}
@@ -146,20 +151,12 @@
}
-static const int kDumpLockRetries = 50;
-static const int kDumpLockSleep = 60000;
+static const int kDumpLockTimeoutNs = 1 * NANOS_PER_SECOND;
-static bool tryLock(Mutex& mutex)
+static bool dumpTryLock(Mutex& mutex)
{
- bool locked = false;
- for (int i = 0; i < kDumpLockRetries; ++i) {
- if (mutex.tryLock() == NO_ERROR) {
- locked = true;
- break;
- }
- usleep(kDumpLockSleep);
- }
- return locked;
+ status_t err = mutex.timedLock(kDumpLockTimeoutNs);
+ return err == NO_ERROR;
}
status_t SoundTriggerHwService::dump(int fd, const Vector<String16>& args __unused) {
@@ -168,7 +165,7 @@
result.appendFormat("Permission Denial: can't dump SoundTriggerHwService");
write(fd, result.string(), result.size());
} else {
- bool locked = tryLock(mServiceLock);
+ bool locked = dumpTryLock(mServiceLock);
// failed to lock - SoundTriggerHwService is probably deadlocked
if (!locked) {
result.append("SoundTriggerHwService may be deadlocked\n");
@@ -486,7 +483,8 @@
}
sp<SoundTriggerHwService::ModuleClient>
-SoundTriggerHwService::Module::addClient(const sp<ISoundTriggerClient>& client)
+SoundTriggerHwService::Module::addClient(const sp<ISoundTriggerClient>& client,
+ const String16& opPackageName)
{
AutoMutex lock(mLock);
sp<ModuleClient> moduleClient;
@@ -497,7 +495,7 @@
return moduleClient;
}
}
- moduleClient = new ModuleClient(this, client);
+ moduleClient = new ModuleClient(this, client, opPackageName);
ALOGV("addClient() client %p", moduleClient.get());
mModuleClients.add(moduleClient);
@@ -920,8 +918,9 @@
#define LOG_TAG "SoundTriggerHwService::ModuleClient"
SoundTriggerHwService::ModuleClient::ModuleClient(const sp<Module>& module,
- const sp<ISoundTriggerClient>& client)
- : mModule(module), mClient(client)
+ const sp<ISoundTriggerClient>& client,
+ const String16& opPackageName)
+ : mModule(module), mClient(client), mOpPackageName(opPackageName)
{
}
@@ -945,7 +944,8 @@
void SoundTriggerHwService::ModuleClient::detach() {
ALOGV("detach()");
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(mOpPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return;
}
@@ -969,7 +969,8 @@
sound_model_handle_t *handle)
{
ALOGV("loadSoundModel() handle");
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(mOpPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
@@ -987,7 +988,8 @@
status_t SoundTriggerHwService::ModuleClient::unloadSoundModel(sound_model_handle_t handle)
{
ALOGV("unloadSoundModel() model handle %d", handle);
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(mOpPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
@@ -1003,7 +1005,8 @@
const sp<IMemory>& dataMemory)
{
ALOGV("startRecognition() model handle %d", handle);
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(mOpPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
@@ -1021,7 +1024,8 @@
status_t SoundTriggerHwService::ModuleClient::stopRecognition(sound_model_handle_t handle)
{
ALOGV("stopRecognition() model handle %d", handle);
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(mOpPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
@@ -1036,7 +1040,8 @@
status_t SoundTriggerHwService::ModuleClient::getModelState(sound_model_handle_t handle)
{
ALOGV("getModelState() model handle %d", handle);
- if (!captureHotwordAllowed(IPCThreadState::self()->getCallingPid(),
+ if (!captureHotwordAllowed(mOpPackageName,
+ IPCThreadState::self()->getCallingPid(),
IPCThreadState::self()->getCallingUid())) {
return PERMISSION_DENIED;
}
diff --git a/services/soundtrigger/SoundTriggerHwService.h b/services/soundtrigger/SoundTriggerHwService.h
index 4258ec0..43ad611 100644
--- a/services/soundtrigger/SoundTriggerHwService.h
+++ b/services/soundtrigger/SoundTriggerHwService.h
@@ -47,10 +47,12 @@
virtual ~SoundTriggerHwService();
// ISoundTriggerHwService
- virtual status_t listModules(struct sound_trigger_module_descriptor *modules,
+ virtual status_t listModules(const String16& opPackageName,
+ struct sound_trigger_module_descriptor *modules,
uint32_t *numModules);
- virtual status_t attach(const sound_trigger_module_handle_t handle,
+ virtual status_t attach(const String16& opPackageName,
+ const sound_trigger_module_handle_t handle,
const sp<ISoundTriggerClient>& client,
sp<ISoundTrigger>& module);
@@ -133,7 +135,8 @@
void setCaptureState_l(bool active);
- sp<ModuleClient> addClient(const sp<ISoundTriggerClient>& client);
+ sp<ModuleClient> addClient(const sp<ISoundTriggerClient>& client,
+ const String16& opPackageName);
void detach(const sp<ModuleClient>& moduleClient);
@@ -156,7 +159,8 @@
public:
ModuleClient(const sp<Module>& module,
- const sp<ISoundTriggerClient>& client);
+ const sp<ISoundTriggerClient>& client,
+ const String16& opPackageName);
virtual ~ModuleClient();
@@ -190,6 +194,7 @@
mutable Mutex mLock;
wp<Module> mModule;
sp<ISoundTriggerClient> mClient;
+ const String16 mOpPackageName;
}; // class ModuleClient
class CallbackThread : public Thread {
diff --git a/soundtrigger/ISoundTriggerHwService.cpp b/soundtrigger/ISoundTriggerHwService.cpp
index d44f5cb..bd107b4 100644
--- a/soundtrigger/ISoundTriggerHwService.cpp
+++ b/soundtrigger/ISoundTriggerHwService.cpp
@@ -50,7 +50,8 @@
{
}
- virtual status_t listModules(struct sound_trigger_module_descriptor *modules,
+ virtual status_t listModules(const String16& opPackageName,
+ struct sound_trigger_module_descriptor *modules,
uint32_t *numModules)
{
if (numModules == NULL || (*numModules != 0 && modules == NULL)) {
@@ -58,6 +59,7 @@
}
Parcel data, reply;
data.writeInterfaceToken(ISoundTriggerHwService::getInterfaceDescriptor());
+ data.writeString16(opPackageName);
unsigned int numModulesReq = (modules == NULL) ? 0 : *numModules;
data.writeInt32(numModulesReq);
status_t status = remote()->transact(LIST_MODULES, data, &reply);
@@ -77,12 +79,14 @@
return status;
}
- virtual status_t attach(const sound_trigger_module_handle_t handle,
+ virtual status_t attach(const String16& opPackageName,
+ const sound_trigger_module_handle_t handle,
const sp<ISoundTriggerClient>& client,
sp<ISoundTrigger>& module)
{
Parcel data, reply;
data.writeInterfaceToken(ISoundTriggerHwService::getInterfaceDescriptor());
+ data.writeString16(opPackageName);
data.write(&handle, sizeof(sound_trigger_module_handle_t));
data.writeStrongBinder(IInterface::asBinder(client));
status_t status = remote()->transact(ATTACH, data, &reply);
@@ -120,6 +124,11 @@
switch(code) {
case LIST_MODULES: {
CHECK_INTERFACE(ISoundTriggerHwService, data, reply);
+ String16 opPackageName;
+ status_t status = data.readString16(&opPackageName);
+ if (status != NO_ERROR) {
+ return status;
+ }
unsigned int numModulesReq = data.readInt32();
if (numModulesReq > MAX_ITEMS_PER_LIST) {
numModulesReq = MAX_ITEMS_PER_LIST;
@@ -133,7 +142,7 @@
reply->writeInt32(0);
return NO_ERROR;
}
- status_t status = listModules(modules, &numModules);
+ status = listModules(opPackageName, modules, &numModules);
reply->writeInt32(status);
reply->writeInt32(numModules);
ALOGV("LIST_MODULES status %d got numModules %d", status, numModules);
@@ -151,12 +160,17 @@
case ATTACH: {
CHECK_INTERFACE(ISoundTriggerHwService, data, reply);
+ String16 opPackageName;
+ status_t status = data.readString16(&opPackageName);
+ if (status != NO_ERROR) {
+ return status;
+ }
sound_trigger_module_handle_t handle;
data.read(&handle, sizeof(sound_trigger_module_handle_t));
sp<ISoundTriggerClient> client =
interface_cast<ISoundTriggerClient>(data.readStrongBinder());
sp<ISoundTrigger> module;
- status_t status = attach(handle, client, module);
+ status = attach(opPackageName, handle, client, module);
reply->writeInt32(status);
if (module != 0) {
reply->writeInt32(1);
diff --git a/soundtrigger/SoundTrigger.cpp b/soundtrigger/SoundTrigger.cpp
index d1eb367..9708ea7 100644
--- a/soundtrigger/SoundTrigger.cpp
+++ b/soundtrigger/SoundTrigger.cpp
@@ -80,19 +80,21 @@
}
// Static methods
-status_t SoundTrigger::listModules(struct sound_trigger_module_descriptor *modules,
- uint32_t *numModules)
+status_t SoundTrigger::listModules(const String16& opPackageName,
+ struct sound_trigger_module_descriptor *modules,
+ uint32_t *numModules)
{
ALOGV("listModules()");
const sp<ISoundTriggerHwService> service = getSoundTriggerHwService();
if (service == 0) {
return NO_INIT;
}
- return service->listModules(modules, numModules);
+ return service->listModules(opPackageName, modules, numModules);
}
-sp<SoundTrigger> SoundTrigger::attach(const sound_trigger_module_handle_t module,
- const sp<SoundTriggerCallback>& callback)
+sp<SoundTrigger> SoundTrigger::attach(const String16& opPackageName,
+ const sound_trigger_module_handle_t module,
+ const sp<SoundTriggerCallback>& callback)
{
ALOGV("attach()");
sp<SoundTrigger> soundTrigger;
@@ -101,7 +103,8 @@
return soundTrigger;
}
soundTrigger = new SoundTrigger(module, callback);
- status_t status = service->attach(module, soundTrigger, soundTrigger->mISoundTrigger);
+ status_t status = service->attach(opPackageName, module, soundTrigger,
+ soundTrigger->mISoundTrigger);
if (status == NO_ERROR && soundTrigger->mISoundTrigger != 0) {
IInterface::asBinder(soundTrigger->mISoundTrigger)->linkToDeath(soundTrigger);