[automerge] audio: fix missing package name in attribution source 2p: 410d0567be
Original change: https://googleplex-android-review.googlesource.com/c/platform/frameworks/av/+/20493625
Bug: 243376549
Bug: 258021433
Change-Id: I27f15e54c7d086fa289c1aa5c7c59aca20cefefd
Merged-In: I40040b8ace382f145dcfc8d04d81dcf6a259dfeb
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 65a163f..63c791b 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -185,7 +185,6 @@
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;
@@ -423,6 +422,33 @@
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;
@@ -432,7 +458,7 @@
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),
@@ -443,7 +469,7 @@
ret = AudioSystem::getInputForAttr(&localAttr, &io,
RECORD_RIID_INVALID,
actualSessionId,
- client.attributionSource,
+ adjAttributionSource,
config,
AUDIO_INPUT_FLAG_MMAP_NOIRQ, deviceId, &portId);
}
@@ -859,7 +885,7 @@
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);
@@ -875,6 +901,8 @@
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) {
@@ -2038,7 +2066,7 @@
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);
@@ -2054,7 +2082,8 @@
__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);
@@ -3630,7 +3659,7 @@
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",
@@ -3638,6 +3667,7 @@
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 a9a8765..1d22e19 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7822,8 +7822,6 @@
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) {
@@ -7838,7 +7836,7 @@
}
if (maxSharedAudioHistoryMs != 0) {
- if (!captureHotwordAllowed(checkedAttributionSource)) {
+ if (!captureHotwordAllowed(attributionSource)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -7960,16 +7958,16 @@
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 e0c5fa5..34cf53f 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 77223b6..9856b0b 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -313,23 +313,12 @@
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));
}
@@ -337,7 +326,7 @@
AudioPolicyInterface::output_type_t outputType;
status_t result = mAudioPolicyManager->getOutputForAttr(&attr, &output, session,
&stream,
- adjAttributionSource,
+ attributionSource,
&config,
&flags, &selectedDeviceId, &portId,
&secondaryOutputs,
@@ -350,16 +339,16 @@
case AudioPolicyInterface::API_OUTPUT_LEGACY:
break;
case AudioPolicyInterface::API_OUTPUT_TELEPHONY_TX:
- if (!modifyPhoneStateAllowed(adjAttributionSource)) {
+ 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;
@@ -372,7 +361,7 @@
if (result == NO_ERROR) {
sp<AudioPlaybackClient> client =
- new AudioPlaybackClient(attr, output, adjAttributionSource, session,
+ new AudioPlaybackClient(attr, output, attributionSource, session,
portId, selectedDeviceId, stream);
mAudioPlaybackClients.add(portId, client);
@@ -552,46 +541,21 @@
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.usage,
- adjAttributionSource)));
+ attributionSource)));
// check calling permissions.
// Capturing from FM_TUNER source is controlled by captureTunerAudioInputAllowed() and
// captureAudioOutputAllowed() (deprecated) as this does not affect users privacy
// as does capturing from an actual microphone.
- if (!(recordingAllowed(adjAttributionSource, attr.source)
+ if (!(recordingAllowed(attributionSource, attr.source)
|| attr.source == AUDIO_SOURCE_FM_TUNER)) {
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 canCaptureOutput = captureAudioOutputAllowed(attributionSource);
if ((inputSource == AUDIO_SOURCE_VOICE_UPLINK ||
inputSource == AUDIO_SOURCE_VOICE_DOWNLINK ||
inputSource == AUDIO_SOURCE_VOICE_CALL ||
@@ -601,12 +565,12 @@
}
if (inputSource == AUDIO_SOURCE_FM_TUNER
- && !captureTunerAudioInputAllowed(adjAttributionSource)
+ && !captureTunerAudioInputAllowed(attributionSource)
&& !canCaptureOutput) {
return binderStatusFromStatusT(PERMISSION_DENIED);
}
- bool canCaptureHotword = captureHotwordAllowed(adjAttributionSource);
+ bool canCaptureHotword = captureHotwordAllowed(attributionSource);
if ((inputSource == AUDIO_SOURCE_HOTWORD) && !canCaptureHotword) {
return binderStatusFromStatusT(PERMISSION_DENIED);
}
@@ -614,7 +578,7 @@
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);
}
@@ -628,7 +592,7 @@
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);
@@ -652,7 +616,7 @@
}
break;
case AudioPolicyInterface::API_INPUT_MIX_EXT_POLICY_REROUTE:
- if (!modifyAudioRoutingAllowed(adjAttributionSource)) {
+ if (!modifyAudioRoutingAllowed(attributionSource)) {
ALOGE("getInputForAttr() permission denied: modify audio routing not allowed");
status = PERMISSION_DENIED;
}
@@ -673,7 +637,7 @@
}
sp<AudioRecordClient> client = new AudioRecordClient(attr, input, session, portId,
- selectedDeviceId, adjAttributionSource,
+ selectedDeviceId, attributionSource,
canCaptureOutput, canCaptureHotword,
mOutputCommandThread);
mAudioRecordClients.add(portId, client);