Snap for 9368510 from ad8ec785d11a366ad833f617f9df6d938e5db881 to mainline-sdkext-release
Change-Id: Id93dc4bbc8cf0d99151d0e8a1fee74814c9f9125
diff --git a/media/codec2/components/avc/C2SoftAvcDec.cpp b/media/codec2/components/avc/C2SoftAvcDec.cpp
index 953afc5..96a4c4a 100644
--- a/media/codec2/components/avc/C2SoftAvcDec.cpp
+++ b/media/codec2/components/avc/C2SoftAvcDec.cpp
@@ -671,6 +671,9 @@
void C2SoftAvcDec::resetPlugin() {
mSignalledOutputEos = false;
mTimeStart = mTimeEnd = systemTime();
+ if (mOutBlock) {
+ mOutBlock.reset();
+ }
}
status_t C2SoftAvcDec::deleteDecoder() {
diff --git a/media/codec2/components/hevc/C2SoftHevcDec.cpp b/media/codec2/components/hevc/C2SoftHevcDec.cpp
index a27c218..15d6dcd 100644
--- a/media/codec2/components/hevc/C2SoftHevcDec.cpp
+++ b/media/codec2/components/hevc/C2SoftHevcDec.cpp
@@ -664,6 +664,9 @@
void C2SoftHevcDec::resetPlugin() {
mSignalledOutputEos = false;
mTimeStart = mTimeEnd = systemTime();
+ if (mOutBlock) {
+ mOutBlock.reset();
+ }
}
status_t C2SoftHevcDec::deleteDecoder() {
diff --git a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
index 9a41910..439323c 100644
--- a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
+++ b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
@@ -732,6 +732,9 @@
void C2SoftMpeg2Dec::resetPlugin() {
mSignalledOutputEos = false;
mTimeStart = mTimeEnd = systemTime();
+ if (mOutBlock) {
+ mOutBlock.reset();
+ }
}
status_t C2SoftMpeg2Dec::deleteDecoder() {
diff --git a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
index 54a1d0e..3bf9c48 100644
--- a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
+++ b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Dec.cpp
@@ -256,7 +256,9 @@
mFramesConfigured = false;
mSignalledOutputEos = false;
mSignalledError = false;
-
+ if (mOutBlock) {
+ mOutBlock.reset();
+ }
return C2_OK;
}
diff --git a/media/codec2/sfplugin/utils/Codec2CommonUtils.cpp b/media/codec2/sfplugin/utils/Codec2CommonUtils.cpp
index ef5800d..332d3ac 100644
--- a/media/codec2/sfplugin/utils/Codec2CommonUtils.cpp
+++ b/media/codec2/sfplugin/utils/Codec2CommonUtils.cpp
@@ -38,7 +38,7 @@
!strcmp(deviceCodeName, "Tiramisu");
}
-bool isVendorApiOrFirstApiAtLeastT() {
+static bool isP010Allowed() {
// The first SDK the device shipped with.
static const int32_t kProductFirstApiLevel =
base::GetIntProperty<int32_t>("ro.product.first_api_level", 0);
@@ -47,6 +47,17 @@
// to signal which VSR requirements they conform to even if the first device SDK was higher.
static const int32_t kBoardFirstApiLevel =
base::GetIntProperty<int32_t>("ro.board.first_api_level", 0);
+
+ // Some devices that launched prior to Android S may not support P010 correctly, even
+ // though they may advertise it as supported.
+ if (kProductFirstApiLevel != 0 && kProductFirstApiLevel < __ANDROID_API_S__) {
+ return false;
+ }
+
+ if (kBoardFirstApiLevel != 0 && kBoardFirstApiLevel < __ANDROID_API_S__) {
+ return false;
+ }
+
static const int32_t kBoardApiLevel =
base::GetIntProperty<int32_t>("ro.board.api_level", 0);
@@ -67,7 +78,7 @@
// API alone. For now limit P010 to devices that launched with Android T or known to conform
// to Android T VSR (as opposed to simply limiting to a T vendor image).
if (format == (AHardwareBuffer_Format)HAL_PIXEL_FORMAT_YCBCR_P010 &&
- !isVendorApiOrFirstApiAtLeastT()) {
+ !isP010Allowed()) {
return false;
}
diff --git a/media/libstagefright/rtsp/AAVCAssembler.cpp b/media/libstagefright/rtsp/AAVCAssembler.cpp
index ddf797c..88f7be7 100644
--- a/media/libstagefright/rtsp/AAVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AAVCAssembler.cpp
@@ -332,6 +332,11 @@
}
bool AAVCAssembler::dropFramesUntilIframe(const sp<ABuffer> &buffer) {
+ if (buffer->size() == 0) {
+ ALOGE("b/230630526 buffer->size() == 0");
+ android_errorWriteLog(0x534e4554, "230630526");
+ return false;
+ }
const uint8_t *data = buffer->data();
unsigned nalType = data[0] & 0x1f;
if (!mFirstIFrameProvided && nalType < 0x5) {
@@ -624,8 +629,7 @@
int32_t firstSeqNo = buffer->int32Data();
// This only works for FU-A type & non-start sequence
- int32_t nalType = buffer->size() >= 1 ? buffer->data()[0] & 0x1f : -1;
- if (nalType != 28 || (buffer->size() >= 2 && buffer->data()[1] & 0x80)) {
+ if (buffer->size() < 2 || (buffer->data()[0] & 0x1f) != 28 || buffer->data()[1] & 0x80) {
return firstSeqNo;
}
diff --git a/media/libstagefright/rtsp/AHEVCAssembler.cpp b/media/libstagefright/rtsp/AHEVCAssembler.cpp
index bb42d1f..72dd981 100644
--- a/media/libstagefright/rtsp/AHEVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AHEVCAssembler.cpp
@@ -629,13 +629,13 @@
int32_t AHEVCAssembler::pickStartSeq(const Queue *queue,
uint32_t first, int64_t play, int64_t jit) {
+ CHECK(!queue->empty());
// pick the first sequence number has the start bit.
sp<ABuffer> buffer = *(queue->begin());
int32_t firstSeqNo = buffer->int32Data();
// This only works for FU-A type & non-start sequence
- unsigned nalType = buffer->data()[0] & 0x1f;
- if (nalType != 28 || buffer->data()[2] & 0x80) {
+ if (buffer->size() < 3 || (buffer->data()[0] & 0x1f) != 28 || buffer->data()[2] & 0x80) {
return firstSeqNo;
}
@@ -645,7 +645,7 @@
if (rtpTime + jit >= play) {
break;
}
- if ((data[2] & 0x80)) {
+ if (it->size() >= 3 && (data[2] & 0x80)) {
const int32_t seqNo = it->int32Data();
ALOGE("finding [HEAD] pkt. \t Seq# (%d ~ )[%d", firstSeqNo, seqNo);
firstSeqNo = seqNo;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f7576f6..23a3a36 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -280,7 +280,6 @@
return opPackageLegacy == package; }) == packages.end()) {
ALOGW("The package name(%s) provided does not correspond to the uid %d",
attributionSource.packageName.value_or("").c_str(), attributionSource.uid);
- checkedAttributionSource.packageName = std::optional<std::string>();
}
}
return checkedAttributionSource;
@@ -579,6 +578,33 @@
audio_io_handle_t io = AUDIO_IO_HANDLE_NONE;
audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
audio_attributes_t localAttr = *attr;
+
+ // TODO b/182392553: refactor or make clearer
+ pid_t clientPid =
+ VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(client.attributionSource.pid));
+ bool updatePid = (clientPid == (pid_t)-1);
+ const uid_t callingUid = IPCThreadState::self()->getCallingUid();
+
+ AttributionSourceState adjAttributionSource = client.attributionSource;
+ if (!isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
+ uid_t clientUid =
+ VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(client.attributionSource.uid));
+ ALOGW_IF(clientUid != callingUid,
+ "%s uid %d tried to pass itself off as %d",
+ __FUNCTION__, callingUid, clientUid);
+ adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
+ updatePid = true;
+ }
+ if (updatePid) {
+ const pid_t callingPid = IPCThreadState::self()->getCallingPid();
+ ALOGW_IF(clientPid != (pid_t)-1 && clientPid != callingPid,
+ "%s uid %d pid %d tried to pass itself off as pid %d",
+ __func__, callingUid, callingPid, clientPid);
+ adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
+ }
+ adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+ adjAttributionSource);
+
if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
audio_config_t fullConfig = AUDIO_CONFIG_INITIALIZER;
fullConfig.sample_rate = config->sample_rate;
@@ -588,7 +614,7 @@
bool isSpatialized;
ret = AudioSystem::getOutputForAttr(&localAttr, &io,
actualSessionId,
- &streamType, client.attributionSource,
+ &streamType, adjAttributionSource,
&fullConfig,
(audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
AUDIO_OUTPUT_FLAG_DIRECT),
@@ -599,7 +625,7 @@
ret = AudioSystem::getInputForAttr(&localAttr, &io,
RECORD_RIID_INVALID,
actualSessionId,
- client.attributionSource,
+ adjAttributionSource,
config,
AUDIO_INPUT_FLAG_MMAP_NOIRQ, deviceId, &portId);
}
@@ -1048,7 +1074,7 @@
audio_attributes_t localAttr = input.attr;
AttributionSourceState adjAttributionSource = input.clientInfo.attributionSource;
- if (!isAudioServerOrMediaServerUid(callingUid)) {
+ if (!isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
ALOGW_IF(clientUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, clientUid);
@@ -1064,6 +1090,8 @@
clientPid = callingPid;
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
}
+ adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+ adjAttributionSource);
audio_session_t sessionId = input.sessionId;
if (sessionId == AUDIO_SESSION_ALLOCATE) {
@@ -2229,7 +2257,7 @@
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
const uid_t currentUid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(
adjAttributionSource.uid));
- if (!isAudioServerOrMediaServerUid(callingUid)) {
+ if (!isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
ALOGW_IF(currentUid != callingUid,
"%s uid %d tried to pass itself off as %d",
__FUNCTION__, callingUid, currentUid);
@@ -2245,7 +2273,8 @@
__func__, callingUid, callingPid, currentPid);
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
}
-
+ adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+ adjAttributionSource);
// we don't yet support anything other than linear PCM
if (!audio_is_valid_format(input.config.format) || !audio_is_linear_pcm(input.config.format)) {
ALOGE("createRecord() invalid format %#x", input.config.format);
@@ -3862,7 +3891,7 @@
const uid_t callingUid = IPCThreadState::self()->getCallingUid();
adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(adjAttributionSource.pid));
- if (currentPid == -1 || !isAudioServerOrMediaServerUid(callingUid)) {
+ if (currentPid == -1 || !isAudioServerOrMediaServerOrSystemServerOrRootUid(callingUid)) {
const pid_t callingPid = IPCThreadState::self()->getCallingPid();
ALOGW_IF(currentPid != -1 && currentPid != callingPid,
"%s uid %d pid %d tried to pass itself off as pid %d",
@@ -3870,6 +3899,7 @@
adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
currentPid = callingPid;
}
+ adjAttributionSource = AudioFlinger::checkAttributionSourcePackage(adjAttributionSource);
ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p",
adjAttributionSource.pid, effectClient.get(), priority, sessionId, io,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 07e82a8..683e320 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -8165,8 +8165,6 @@
audio_input_flags_t inputFlags = mInput->flags;
audio_input_flags_t requestedFlags = *flags;
uint32_t sampleRate;
- AttributionSourceState checkedAttributionSource = AudioFlinger::checkAttributionSourcePackage(
- attributionSource);
lStatus = initCheck();
if (lStatus != NO_ERROR) {
@@ -8181,7 +8179,7 @@
}
if (maxSharedAudioHistoryMs != 0) {
- if (!captureHotwordAllowed(checkedAttributionSource)) {
+ if (!captureHotwordAllowed(attributionSource)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -8302,16 +8300,16 @@
Mutex::Autolock _l(mLock);
int32_t startFrames = -1;
if (!mSharedAudioPackageName.empty()
- && mSharedAudioPackageName == checkedAttributionSource.packageName
+ && mSharedAudioPackageName == attributionSource.packageName
&& mSharedAudioSessionId == sessionId
- && captureHotwordAllowed(checkedAttributionSource)) {
+ && captureHotwordAllowed(attributionSource)) {
startFrames = mSharedAudioStartFrames;
}
track = new RecordTrack(this, client, attr, sampleRate,
format, channelMask, frameCount,
nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid,
- checkedAttributionSource, *flags, TrackBase::TYPE_DEFAULT, portId,
+ attributionSource, *flags, TrackBase::TYPE_DEFAULT, portId,
startFrames);
lStatus = track->initCheck();
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6135020..83a8bb0 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -529,10 +529,7 @@
id, attr.flags);
return nullptr;
}
-
- AttributionSourceState checkedAttributionSource = AudioFlinger::checkAttributionSourcePackage(
- attributionSource);
- return new OpPlayAudioMonitor(checkedAttributionSource, attr.usage, id);
+ return new OpPlayAudioMonitor(attributionSource, attr.usage, id);
}
AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor(
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index df49bba..49224c5 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -352,31 +352,20 @@
ALOGV("%s()", __func__);
Mutex::Autolock _l(mLock);
- // TODO b/182392553: refactor or remove
- AttributionSourceState adjAttributionSource = attributionSource;
- const uid_t callingUid = IPCThreadState::self()->getCallingUid();
- if (!isAudioServerOrMediaServerUid(callingUid) || attributionSource.uid == -1) {
- int32_t callingUidAidl = VALUE_OR_RETURN_BINDER_STATUS(
- legacy2aidl_uid_t_int32_t(callingUid));
- ALOGW_IF(attributionSource.uid != -1 && attributionSource.uid != callingUidAidl,
- "%s uid %d tried to pass itself off as %d", __func__,
- callingUidAidl, attributionSource.uid);
- adjAttributionSource.uid = callingUidAidl;
- }
if (!mPackageManager.allowPlaybackCapture(VALUE_OR_RETURN_BINDER_STATUS(
- aidl2legacy_int32_t_uid_t(adjAttributionSource.uid)))) {
+ aidl2legacy_int32_t_uid_t(attributionSource.uid)))) {
attr.flags = static_cast<audio_flags_mask_t>(attr.flags | AUDIO_FLAG_NO_MEDIA_PROJECTION);
}
if (((attr.flags & (AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE)) != 0)
- && !bypassInterruptionPolicyAllowed(adjAttributionSource)) {
+ && !bypassInterruptionPolicyAllowed(attributionSource)) {
attr.flags = static_cast<audio_flags_mask_t>(
attr.flags & ~(AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE));
}
if (attr.content_type == AUDIO_CONTENT_TYPE_ULTRASOUND) {
- if (!accessUltrasoundAllowed(adjAttributionSource)) {
+ if (!accessUltrasoundAllowed(attributionSource)) {
ALOGE("%s: permission denied: ultrasound not allowed for uid %d pid %d",
- __func__, adjAttributionSource.uid, adjAttributionSource.pid);
+ __func__, attributionSource.uid, attributionSource.pid);
return binderStatusFromStatusT(PERMISSION_DENIED);
}
}
@@ -386,7 +375,7 @@
bool isSpatialized = false;
status_t result = mAudioPolicyManager->getOutputForAttr(&attr, &output, session,
&stream,
- adjAttributionSource,
+ attributionSource,
&config,
&flags, &selectedDeviceId, &portId,
&secondaryOutputs,
@@ -401,20 +390,20 @@
break;
case AudioPolicyInterface::API_OUTPUT_TELEPHONY_TX:
if (((attr.flags & AUDIO_FLAG_CALL_REDIRECTION) != 0)
- && !callAudioInterceptionAllowed(adjAttributionSource)) {
+ && !callAudioInterceptionAllowed(attributionSource)) {
ALOGE("%s() permission denied: call redirection not allowed for uid %d",
- __func__, adjAttributionSource.uid);
+ __func__, attributionSource.uid);
result = PERMISSION_DENIED;
- } else if (!modifyPhoneStateAllowed(adjAttributionSource)) {
+ } else if (!modifyPhoneStateAllowed(attributionSource)) {
ALOGE("%s() permission denied: modify phone state not allowed for uid %d",
- __func__, adjAttributionSource.uid);
+ __func__, attributionSource.uid);
result = PERMISSION_DENIED;
}
break;
case AudioPolicyInterface::API_OUT_MIX_PLAYBACK:
- if (!modifyAudioRoutingAllowed(adjAttributionSource)) {
+ if (!modifyAudioRoutingAllowed(attributionSource)) {
ALOGE("%s() permission denied: modify audio routing not allowed for uid %d",
- __func__, adjAttributionSource.uid);
+ __func__, attributionSource.uid);
result = PERMISSION_DENIED;
}
break;
@@ -427,7 +416,7 @@
if (result == NO_ERROR) {
sp<AudioPlaybackClient> client =
- new AudioPlaybackClient(attr, output, adjAttributionSource, session,
+ new AudioPlaybackClient(attr, output, attributionSource, session,
portId, selectedDeviceId, stream, isSpatialized);
mAudioPlaybackClients.add(portId, client);
@@ -613,33 +602,8 @@
return binderStatusFromStatusT(BAD_VALUE);
}
- // Make sure attribution source represents the current caller
- AttributionSourceState adjAttributionSource = attributionSource;
- // TODO b/182392553: refactor or remove
- bool updatePid = (attributionSource.pid == -1);
- const uid_t callingUid =IPCThreadState::self()->getCallingUid();
- const uid_t currentUid = VALUE_OR_RETURN_BINDER_STATUS(aidl2legacy_int32_t_uid_t(
- attributionSource.uid));
- if (!isAudioServerOrMediaServerUid(callingUid)) {
- ALOGW_IF(currentUid != (uid_t)-1 && currentUid != callingUid,
- "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid,
- currentUid);
- adjAttributionSource.uid = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_uid_t_int32_t(
- callingUid));
- updatePid = true;
- }
-
- if (updatePid) {
- const int32_t callingPid = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_pid_t_int32_t(
- IPCThreadState::self()->getCallingPid()));
- ALOGW_IF(attributionSource.pid != -1 && attributionSource.pid != callingPid,
- "%s uid %d pid %d tried to pass itself off as pid %d",
- __func__, adjAttributionSource.uid, callingPid, attributionSource.pid);
- adjAttributionSource.pid = callingPid;
- }
-
RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(validateUsage(attr,
- adjAttributionSource)));
+ attributionSource)));
// check calling permissions.
// Capturing from the following sources does not require permission RECORD_AUDIO
@@ -650,17 +614,17 @@
// type is API_INPUT_MIX_EXT_POLICY_REROUTE and by AudioService if a media projection
// is used and input type is API_INPUT_MIX_PUBLIC_CAPTURE_PLAYBACK
// - ECHO_REFERENCE source is controlled by captureAudioOutputAllowed()
- if (!(recordingAllowed(adjAttributionSource, inputSource)
+ if (!(recordingAllowed(attributionSource, inputSource)
|| inputSource == AUDIO_SOURCE_FM_TUNER
|| inputSource == AUDIO_SOURCE_REMOTE_SUBMIX
|| inputSource == AUDIO_SOURCE_ECHO_REFERENCE)) {
ALOGE("%s permission denied: recording not allowed for %s",
- __func__, adjAttributionSource.toString().c_str());
+ __func__, attributionSource.toString().c_str());
return binderStatusFromStatusT(PERMISSION_DENIED);
}
- bool canCaptureOutput = captureAudioOutputAllowed(adjAttributionSource);
- bool canInterceptCallAudio = callAudioInterceptionAllowed(adjAttributionSource);
+ bool canCaptureOutput = captureAudioOutputAllowed(attributionSource);
+ bool canInterceptCallAudio = callAudioInterceptionAllowed(attributionSource);
bool isCallAudioSource = inputSource == AUDIO_SOURCE_VOICE_UPLINK
|| inputSource == AUDIO_SOURCE_VOICE_DOWNLINK
|| inputSource == AUDIO_SOURCE_VOICE_CALL;
@@ -674,11 +638,11 @@
}
if (inputSource == AUDIO_SOURCE_FM_TUNER
&& !canCaptureOutput
- && !captureTunerAudioInputAllowed(adjAttributionSource)) {
+ && !captureTunerAudioInputAllowed(attributionSource)) {
return binderStatusFromStatusT(PERMISSION_DENIED);
}
- bool canCaptureHotword = captureHotwordAllowed(adjAttributionSource);
+ bool canCaptureHotword = captureHotwordAllowed(attributionSource);
if ((inputSource == AUDIO_SOURCE_HOTWORD) && !canCaptureHotword) {
return binderStatusFromStatusT(PERMISSION_DENIED);
}
@@ -686,14 +650,14 @@
if (((flags & AUDIO_INPUT_FLAG_HW_HOTWORD) != 0)
&& !canCaptureHotword) {
ALOGE("%s: permission denied: hotword mode not allowed"
- " for uid %d pid %d", __func__, adjAttributionSource.uid, adjAttributionSource.pid);
+ " for uid %d pid %d", __func__, attributionSource.uid, attributionSource.pid);
return binderStatusFromStatusT(PERMISSION_DENIED);
}
if (attr.source == AUDIO_SOURCE_ULTRASOUND) {
- if (!accessUltrasoundAllowed(adjAttributionSource)) {
+ if (!accessUltrasoundAllowed(attributionSource)) {
ALOGE("%s: permission denied: ultrasound not allowed for uid %d pid %d",
- __func__, adjAttributionSource.uid, adjAttributionSource.pid);
+ __func__, attributionSource.uid, attributionSource.pid);
return binderStatusFromStatusT(PERMISSION_DENIED);
}
}
@@ -708,7 +672,7 @@
AutoCallerClear acc;
// the audio_in_acoustics_t parameter is ignored by get_input()
status = mAudioPolicyManager->getInputForAttr(&attr, &input, riid, session,
- adjAttributionSource, &config,
+ attributionSource, &config,
flags, &selectedDeviceId,
&inputType, &portId);
@@ -737,7 +701,7 @@
}
break;
case AudioPolicyInterface::API_INPUT_MIX_EXT_POLICY_REROUTE:
- if (!(modifyAudioRoutingAllowed(adjAttributionSource)
+ if (!(modifyAudioRoutingAllowed(attributionSource)
|| ((attr.flags & AUDIO_FLAG_CALL_REDIRECTION) != 0
&& canInterceptCallAudio))) {
ALOGE("%s permission denied for remote submix capture", __func__);
@@ -760,7 +724,7 @@
}
sp<AudioRecordClient> client = new AudioRecordClient(attr, input, session, portId,
- selectedDeviceId, adjAttributionSource,
+ selectedDeviceId, attributionSource,
canCaptureOutput, canCaptureHotword,
mOutputCommandThread);
mAudioRecordClients.add(portId, client);
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index e7d945f..bfce4ba 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -1803,12 +1803,14 @@
void AudioPolicyService::SensorPrivacyPolicy::registerSelf() {
SensorPrivacyManager spm;
mSensorPrivacyEnabled = spm.isSensorPrivacyEnabled();
+ (void)spm.addToggleSensorPrivacyListener(this);
spm.addSensorPrivacyListener(this);
}
void AudioPolicyService::SensorPrivacyPolicy::unregisterSelf() {
SensorPrivacyManager spm;
spm.removeSensorPrivacyListener(this);
+ spm.removeToggleSensorPrivacyListener(this);
}
bool AudioPolicyService::SensorPrivacyPolicy::isSensorPrivacyEnabled() {
