Merge "Merge 24Q3 (ab/AP3A.240905.001) to aosp-main-future" into aosp-main-future
diff --git a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
index 4183023..4ab5d10 100644
--- a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
@@ -619,6 +619,13 @@
if (!isValidOMXParam(outParams)) {
return OMX_ErrorBadParameter;
}
+ if (offsetof(DescribeHDR10PlusInfoParams, nValue) + outParams->nParamSize >
+ outParams->nSize) {
+ ALOGE("b/329641908: too large param size; nParamSize=%u nSize=%u",
+ outParams->nParamSize, outParams->nSize);
+ android_errorWriteLog(0x534e4554, "329641908");
+ return OMX_ErrorBadParameter;
+ }
outParams->nParamSizeUsed = info->size();
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index e91e2a3..deb7345 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -294,8 +294,7 @@
virtual status_t startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId,
- uid_t uid,
- bool internal = false) = 0;
+ uid_t uid) = 0;
virtual status_t stopAudioSource(audio_port_handle_t portId) = 0;
virtual status_t setMasterMono(bool mono) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h b/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
index 6167f95..e519766 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
@@ -47,13 +47,17 @@
if (active) {
// On MMAP IOs, the preferred device is selected by the first client (virtual client
- // created when the mmap stream is opened). This client is never active.
+ // created when the mmap stream is opened). This client is never active and we only
+ // consider the Filter criteria, not the active state.
// On non MMAP IOs, the preferred device is honored only if all active clients have
// a preferred device in which case the first client drives the selection.
if (desc->isMmap()) {
- // The client list is never empty on a MMAP IO
- return devices.getDeviceFromId(
- desc->clientsList(false /*activeOnly*/)[0]->preferredDeviceId());
+ auto matchingClients = desc->clientsList(
+ false /*activeOnly*/, filter, false /*preferredDevice*/);
+ if (matchingClients.empty()) {
+ return nullptr;
+ }
+ return devices.getDeviceFromId(matchingClients[0]->preferredDeviceId());
} else {
auto activeClientsWithRoute =
desc->clientsList(true /*activeOnly*/, filter, true /*preferredDevice*/);
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
index a596c43..60da405 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -63,6 +63,8 @@
* HW Audio Source.
*/
virtual bool isInternal() const { return false; }
+ virtual bool isCallRx() const { return false; }
+ virtual bool isCallTx() const { return false; }
audio_port_handle_t portId() const { return mPortId; }
uid_t uid() const { return mUid; }
audio_session_t session() const { return mSessionId; };
@@ -236,7 +238,7 @@
const sp<DeviceDescriptor>& srcDevice,
audio_stream_type_t stream, product_strategy_t strategy,
VolumeSource volumeSource,
- bool isInternal);
+ bool isInternal, bool isCallRx, bool isCallTx);
~SourceClientDescriptor() override = default;
@@ -263,6 +265,8 @@
wp<HwAudioOutputDescriptor> hwOutput() const { return mHwOutput; }
void setHwOutput(const sp<HwAudioOutputDescriptor>& hwOutput);
bool isInternal() const override { return mIsInternal; }
+ bool isCallRx() const override { return mIsCallRx; }
+ bool isCallTx() const override { return mIsCallTx; }
using ClientDescriptor::dump;
void dump(String8 *dst, int spaces) const override;
@@ -294,6 +298,8 @@
* requester to prevent rerouting SwOutput involved in raw patches.
*/
bool mIsInternal = false;
+ bool mIsCallRx = false;
+ bool mIsCallTx = false;
};
class SourceClientCollection :
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index 667c189..ad6977b 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -96,12 +96,14 @@
SourceClientDescriptor::SourceClientDescriptor(audio_port_handle_t portId, uid_t uid,
audio_attributes_t attributes, const struct audio_port_config &config,
const sp<DeviceDescriptor>& srcDevice, audio_stream_type_t stream,
- product_strategy_t strategy, VolumeSource volumeSource, bool isInternal) :
+ product_strategy_t strategy, VolumeSource volumeSource,
+ bool isInternal, bool isCallRx, bool isCallTx) :
TrackClientDescriptor::TrackClientDescriptor(portId, uid, AUDIO_SESSION_NONE, attributes,
{config.sample_rate, config.channel_mask, config.format}, AUDIO_PORT_HANDLE_NONE,
stream, strategy, volumeSource, AUDIO_OUTPUT_FLAG_NONE, false,
{} /* Sources do not support secondary outputs*/, nullptr),
- mSrcDevice(srcDevice), mIsInternal(isInternal)
+ mSrcDevice(srcDevice), mIsInternal(isInternal),
+ mIsCallRx(isCallRx), mIsCallTx(isCallTx)
{
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 2517300..41bf68e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -768,7 +768,7 @@
}
muteWaitMs = setOutputDevices(__func__, mPrimaryOutput, rxDevices, true, delayMs);
} else { // create RX path audio patch
- connectTelephonyRxAudioSource();
+ connectTelephonyRxAudioSource(delayMs);
// If the TX device is on the primary HW module but RX device is
// on other HW module, SinkMetaData of telephony input should handle it
// assuming the device uses audio HAL V5.0 and above
@@ -803,7 +803,7 @@
return false;
}
-void AudioPolicyManager::connectTelephonyRxAudioSource()
+void AudioPolicyManager::connectTelephonyRxAudioSource(uint32_t delayMs)
{
disconnectTelephonyAudioSource(mCallRxSourceClient);
const struct audio_port_config source = {
@@ -813,7 +813,8 @@
const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL);
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
- status_t status = startAudioSource(&source, &aa, &portId, 0 /*uid*/, true /*internal*/);
+ status_t status = startAudioSourceInternal(&source, &aa, &portId, 0 /*uid*/,
+ true /*internal*/, true /*isCallRx*/, delayMs);
ALOGE_IF(status != OK, "%s: failed to start audio source (%d)", __func__, status);
mCallRxSourceClient = mAudioSources.valueFor(portId);
ALOGE_IF(mCallRxSourceClient == nullptr,
@@ -850,7 +851,8 @@
srcDevice->toAudioPortConfig(&source);
mCallTxSourceClient = new SourceClientDescriptor(
callTxSourceClientPortId, mUidCached, aa, source, srcDevice, AUDIO_STREAM_PATCH,
- mCommunnicationStrategy, toVolumeSource(aa), true);
+ mCommunnicationStrategy, toVolumeSource(aa), true,
+ false /*isCallRx*/, true /*isCallTx*/);
mCallTxSourceClient->setPreferredDeviceId(sinkDevice->getId());
audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE;
@@ -5108,7 +5110,7 @@
new SourceClientDescriptor(
portId, uid, attributes, *source, srcDevice, AUDIO_STREAM_PATCH,
mEngine->getProductStrategyForAttributes(attributes), toVolumeSource(attributes),
- true);
+ true, false /*isCallRx*/, false /*isCallTx*/);
sourceDesc->setPreferredDeviceId(sinkDevice->getId());
status_t status =
@@ -5440,7 +5442,7 @@
outputDesc->toAudioPortConfig(&srcMixPortConfig, nullptr);
// for volume control, we may need a valid stream
srcMixPortConfig.ext.mix.usecase.stream =
- (!sourceDesc->isInternal() || isCallTxAudioSource(sourceDesc)) ?
+ (!sourceDesc->isInternal() || sourceDesc->isCallTx()) ?
mEngine->getStreamTypeForAttributes(sourceDesc->attributes()) :
AUDIO_STREAM_PATCH;
patchBuilder.addSource(srcMixPortConfig);
@@ -5778,7 +5780,16 @@
status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId,
- uid_t uid, bool internal)
+ uid_t uid) {
+ return startAudioSourceInternal(source, attributes, portId, uid,
+ false /*internal*/, false /*isCallRx*/, 0 /*delayMs*/);
+}
+
+status_t AudioPolicyManager::startAudioSourceInternal(const struct audio_port_config *source,
+ const audio_attributes_t *attributes,
+ audio_port_handle_t *portId,
+ uid_t uid, bool internal, bool isCallRx,
+ uint32_t delayMs)
{
ALOGV("%s", __FUNCTION__);
*portId = AUDIO_PORT_HANDLE_NONE;
@@ -5811,16 +5822,17 @@
new SourceClientDescriptor(*portId, uid, *attributes, *source, srcDevice,
mEngine->getStreamTypeForAttributes(*attributes),
mEngine->getProductStrategyForAttributes(*attributes),
- toVolumeSource(*attributes), internal);
+ toVolumeSource(*attributes), internal, isCallRx, false);
- status_t status = connectAudioSource(sourceDesc);
+ status_t status = connectAudioSource(sourceDesc, delayMs);
if (status == NO_ERROR) {
mAudioSources.add(*portId, sourceDesc);
}
return status;
}
-status_t AudioPolicyManager::connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
+status_t AudioPolicyManager::connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc,
+ uint32_t delayMs)
{
ALOGV("%s handle %d", __FUNCTION__, sourceDesc->portId());
@@ -5846,7 +5858,7 @@
audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
return connectAudioSourceToSink(
- sourceDesc, sinkDevice, patchBuilder.patch(), handle, mUidCached, 0 /*delayMs*/);
+ sourceDesc, sinkDevice, patchBuilder.patch(), handle, mUidCached, delayMs);
}
status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId)
@@ -7220,8 +7232,8 @@
sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
if (sourceDesc != nullptr && followsSameRouting(attr, sourceDesc->attributes())
&& sourceDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE
- && !isCallRxAudioSource(sourceDesc) && !sourceDesc->isInternal()) {
- connectAudioSource(sourceDesc);
+ && !sourceDesc->isCallRx() && !sourceDesc->isInternal()) {
+ connectAudioSource(sourceDesc, 0 /*delayMs*/);
}
}
}
@@ -7327,8 +7339,8 @@
}
}
sp<SourceClientDescriptor> source = getSourceForAttributesOnOutput(srcOut, attr);
- if (source != nullptr && !isCallRxAudioSource(source) && !source->isInternal()) {
- connectAudioSource(source);
+ if (source != nullptr && !source->isCallRx() && !source->isInternal()) {
+ connectAudioSource(source, 0 /*delayMs*/);
}
}
@@ -8500,7 +8512,7 @@
sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
if (sourceDesc->isConnected() && (sourceDesc->srcDevice()->equals(deviceDesc) ||
sourceDesc->sinkDevice()->equals(deviceDesc))
- && !isCallRxAudioSource(sourceDesc)) {
+ && !sourceDesc->isCallRx()) {
disconnectAudioSource(sourceDesc);
}
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 011e867..1b4f33a 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -345,8 +345,7 @@
virtual status_t startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId,
- uid_t uid,
- bool internal = false);
+ uid_t uid);
virtual status_t stopAudioSource(audio_port_handle_t portId);
virtual status_t setMasterMono(bool mono);
@@ -705,15 +704,7 @@
void updateCallAndOutputRouting(bool forceVolumeReeval = true, uint32_t delayMs = 0,
bool skipDelays = false);
- bool isCallRxAudioSource(const sp<SourceClientDescriptor> &source) {
- return mCallRxSourceClient != nullptr && source == mCallRxSourceClient;
- }
-
- bool isCallTxAudioSource(const sp<SourceClientDescriptor> &source) {
- return mCallTxSourceClient != nullptr && source == mCallTxSourceClient;
- }
-
- void connectTelephonyRxAudioSource();
+ void connectTelephonyRxAudioSource(uint32_t delayMs);
void disconnectTelephonyAudioSource(sp<SourceClientDescriptor> &clientDesc);
@@ -938,7 +929,8 @@
status_t hasPrimaryOutput() const { return mPrimaryOutput != 0; }
- status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
+ status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc,
+ uint32_t delayMs);
status_t disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
status_t connectAudioSourceToSink(const sp<SourceClientDescriptor>& sourceDesc,
@@ -976,6 +968,13 @@
void checkLeBroadcastRoutes(bool wasUnicastActive,
sp<SwAudioOutputDescriptor> ignoredOutput, uint32_t delayMs);
+ status_t startAudioSourceInternal(const struct audio_port_config *source,
+ const audio_attributes_t *attributes,
+ audio_port_handle_t *portId,
+ uid_t uid,
+ bool internal,
+ bool isCallRx,
+ uint32_t delayMs);
const uid_t mUidCached; // AID_AUDIOSERVER
sp<const AudioPolicyConfig> mConfig;
EngineInstance mEngine; // Audio Policy Engine instance
diff --git a/services/camera/libcameraservice/utils/CameraServiceProxyWrapper.cpp b/services/camera/libcameraservice/utils/CameraServiceProxyWrapper.cpp
index d5e3790..85bca6f 100644
--- a/services/camera/libcameraservice/utils/CameraServiceProxyWrapper.cpp
+++ b/services/camera/libcameraservice/utils/CameraServiceProxyWrapper.cpp
@@ -268,6 +268,12 @@
const auto& gbps = config.getGraphicBufferProducers();
int32_t width = 0, height = 0;
if (gbps.size() > 0) {
+ if (gbps[0] == nullptr) {
+ ALOGE("%s: Failed to query size due to abandoned surface.",
+ __FUNCTION__);
+ return CameraFeatureCombinationStats::CAMERA_FEATURE_UNKNOWN;
+ }
+
sp<Surface> surface = new Surface(gbps[0], /*useAsync*/false);
ANativeWindow *anw = surface.get();