AudioFlinger: Add clang tidy checks on build
First pass just enables warnings.
Test: compiles
Bug: 252907478
Merged-In: I3d0622ab908b85adb1913d8482d55e1950fdccc0
Change-Id: I3d0622ab908b85adb1913d8482d55e1950fdccc0
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 6d4c3a3..f1797e6 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -19,11 +19,127 @@
],
}
+tidy_errors = [
+ // https://clang.llvm.org/extra/clang-tidy/checks/list.html
+ // For many categories, the checks are too many to specify individually.
+ // Feel free to disable as needed - as warnings are generally ignored,
+ // we treat warnings as errors.
+ "android-*",
+ "bugprone-*",
+ "cert-*",
+ "clang-analyzer-security*",
+ "google-*",
+ "misc-*",
+ //"modernize-*", // explicitly list the modernize as they can be subjective.
+ "modernize-avoid-bind",
+ //"modernize-avoid-c-arrays", // std::array<> can be verbose
+ "modernize-concat-nested-namespaces",
+ //"modernize-deprecated-headers", // C headers still ok even if there is C++ equivalent.
+ "modernize-deprecated-ios-base-aliases",
+ "modernize-loop-convert",
+ "modernize-make-shared",
+ "modernize-make-unique",
+ // "modernize-pass-by-value",
+ "modernize-raw-string-literal",
+ "modernize-redundant-void-arg",
+ "modernize-replace-auto-ptr",
+ "modernize-replace-random-shuffle",
+ "modernize-return-braced-init-list",
+ "modernize-shrink-to-fit",
+ "modernize-unary-static-assert",
+ // "modernize-use-auto", // found in MediaMetricsService.h, debatable - auto can obscure type
+ "modernize-use-bool-literals",
+ "modernize-use-default-member-init",
+ "modernize-use-emplace",
+ "modernize-use-equals-default",
+ "modernize-use-equals-delete",
+ // "modernize-use-nodiscard",
+ "modernize-use-noexcept",
+ "modernize-use-nullptr",
+ "modernize-use-override",
+ //"modernize-use-trailing-return-type", // not necessarily more readable
+ "modernize-use-transparent-functors",
+ "modernize-use-uncaught-exceptions",
+ "modernize-use-using",
+ "performance-*",
+
+ // Remove some pedantic stylistic requirements.
+ "-google-readability-casting", // C++ casts not always necessary and may be verbose
+ "-google-readability-todo", // do not require TODO(info)
+
+ "-bugprone-unhandled-self-assignment",
+ "-bugprone-suspicious-string-compare",
+ "-cert-oop54-cpp", // found in TransactionLog.h
+ "-bugprone-narrowing-conversions", // b/182410845
+
+ // TODO(b/275642749) Reenable these warnings
+ "-bugprone-assignment-in-if-condition",
+ "-bugprone-forward-declaration-namespace",
+ "-bugprone-parent-virtual-call",
+ "-cert-dcl59-cpp",
+ "-cert-err34-c",
+ "-google-build-namespaces",
+ "-google-build-using-namespace",
+ "-google-default-arguments",
+ "-google-runtime-int",
+ "-misc-const-correctness",
+ "-misc-non-private-member-variables-in-classes",
+ "-modernize-concat-nested-namespaces",
+ "-modernize-loop-convert",
+ "-modernize-use-default-member-init",
+ "-modernize-use-equals-default",
+ "-modernize-use-nullptr",
+ "-modernize-use-override",
+ "-modernize-use-using",
+ "-performance-no-int-to-ptr",
+]
+
+// Eventually use common tidy defaults
+cc_defaults {
+ name: "audioflinger_flags_defaults",
+ // https://clang.llvm.org/docs/UsersManual.html#command-line-options
+ // https://clang.llvm.org/docs/DiagnosticsReference.html
+ cflags: [
+ "-Wall",
+ "-Wdeprecated",
+ "-Werror",
+ "-Werror=implicit-fallthrough",
+ "-Werror=sometimes-uninitialized",
+ "-Werror=conditional-uninitialized",
+ "-Wextra",
+
+ // suppress some warning chatter.
+ "-Wno-deprecated-copy-with-dtor",
+ "-Wno-deprecated-copy-with-user-provided-dtor",
+
+ "-Wredundant-decls",
+ "-Wshadow",
+ "-Wstrict-aliasing",
+ "-fstrict-aliasing",
+ "-Wthread-safety",
+ //"-Wthread-safety-negative", // experimental - looks broken in R.
+ "-Wunreachable-code",
+ "-Wunreachable-code-break",
+ "-Wunreachable-code-return",
+ "-Wunused",
+ "-Wused-but-marked-unused",
+ "-D_LIBCPP_ENABLE_THREAD_SAFETY_ANNOTATIONS",
+ ],
+ // https://clang.llvm.org/extra/clang-tidy/
+ tidy: true,
+ tidy_checks: tidy_errors,
+ tidy_checks_as_errors: tidy_errors,
+ tidy_flags: [
+ "-format-style=file",
+ ],
+}
+
cc_library_shared {
name: "libaudioflinger",
defaults: [
"latest_android_media_audio_common_types_cpp_shared",
+ "audioflinger_flags_defaults",
],
srcs: [
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index ba7c6b6..85f6078 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -694,7 +694,7 @@
}
status_t AudioFlinger::addEffectToHal(audio_port_handle_t deviceId,
- audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ audio_module_handle_t hwModuleId, const sp<EffectHalInterface>& effect) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(hwModuleId);
if (audioHwDevice == nullptr) {
@@ -704,7 +704,7 @@
}
status_t AudioFlinger::removeEffectFromHal(audio_port_handle_t deviceId,
- audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ audio_module_handle_t hwModuleId, const sp<EffectHalInterface>& effect) {
AutoMutex lock(mHardwareLock);
AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(hwModuleId);
if (audioHwDevice == nullptr) {
@@ -822,6 +822,7 @@
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
@@ -924,7 +925,6 @@
// to lookup the service if it's not running, as it will block for a second
if (sMediaLogServiceAsBinder != 0) {
dprintf(fd, "\nmedia.log:\n");
- Vector<String16> args;
sMediaLogServiceAsBinder->dump(fd, args);
}
@@ -1421,8 +1421,9 @@
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
- for (size_t i = 0; i < mPlaybackThreads.size(); i++)
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->setMode(mode);
+ }
}
mediametrics::LogItem(mMetricsId)
@@ -1760,7 +1761,7 @@
// forwardAudioHwSyncToDownstreamPatches_l() must be called with AudioFlinger::mLock held
void AudioFlinger::forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
- std::function<bool(const sp<PlaybackThread>&)> useThread)
+ const std::function<bool(const sp<PlaybackThread>&)>& useThread)
{
std::vector<PatchPanel::SoftwarePatch> swPatches;
if (mPatchPanel.getDownstreamSoftwarePatches(upStream, &swPatches) != OK) return;
@@ -1776,7 +1777,7 @@
// Update downstream patches for all playback threads attached to an MSD module
void AudioFlinger::updateDownStreamPatches_l(const struct audio_patch *patch,
- const std::set<audio_io_handle_t> streams)
+ const std::set<audio_io_handle_t>& streams)
{
for (const audio_io_handle_t stream : streams) {
PlaybackThread *playbackThread = checkPlaybackThread_l(stream);
@@ -1982,24 +1983,29 @@
mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
+
std::vector<audio_channel_mask_t> channelMasks = {channelMask};
- if (channelMask != AUDIO_CHANNEL_IN_MONO)
+ if (channelMask != AUDIO_CHANNEL_IN_MONO) {
channelMasks.push_back(AUDIO_CHANNEL_IN_MONO);
- if (channelMask != AUDIO_CHANNEL_IN_STEREO)
+ }
+ if (channelMask != AUDIO_CHANNEL_IN_STEREO) {
channelMasks.push_back(AUDIO_CHANNEL_IN_STEREO);
+ }
std::vector<audio_format_t> formats = {format};
- if (format != AUDIO_FORMAT_PCM_16_BIT)
+ if (format != AUDIO_FORMAT_PCM_16_BIT) {
formats.push_back(AUDIO_FORMAT_PCM_16_BIT);
+ }
std::vector<uint32_t> sampleRates = {sampleRate};
static const uint32_t SR_44100 = 44100;
static const uint32_t SR_48000 = 48000;
-
- if (sampleRate != SR_48000)
+ if (sampleRate != SR_48000) {
sampleRates.push_back(SR_48000);
- if (sampleRate != SR_44100)
+ }
+ if (sampleRate != SR_44100) {
sampleRates.push_back(SR_44100);
+ }
mHardwareStatus = AUDIO_HW_IDLE;
@@ -2468,7 +2474,7 @@
// session and move it to this thread.
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
- Mutex::Autolock _l(thread->mLock);
+ Mutex::Autolock _l2(thread->mLock);
thread->addEffectChain_l(chain);
}
break;
@@ -2848,7 +2854,7 @@
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *halConfig,
- audio_config_base_t *mixerConfig __unused,
+ audio_config_base_t *mixerConfig,
audio_devices_t deviceType,
const String8& address,
audio_output_flags_t flags)
@@ -2970,7 +2976,6 @@
aidl2legacy_int32_t_audio_output_flags_t_mask(request.flags));
audio_io_handle_t output;
- uint32_t latencyMs;
ALOGI("openOutput() this %p, module %d Device %s, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
@@ -2993,6 +2998,7 @@
sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
&mixerConfig, deviceType, address, flags);
if (thread != 0) {
+ uint32_t latencyMs = 0;
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
latencyMs = playbackThread->latency();
@@ -3341,7 +3347,7 @@
continue;
}
if (t->hasAudioSession(chain->sessionId()) != 0) {
- Mutex::Autolock _l(t->mLock);
+ Mutex::Autolock _l2(t->mLock);
ALOGV("closeInput() found thread %d for effect session %d",
t->id(), chain->sessionId());
t->addEffectChain_l(chain);
@@ -3546,7 +3552,8 @@
}
for (size_t i = 0; i < chains.size(); i++) {
- sp<EffectChain> ec = chains[i];
+ // clang-tidy suggests const ref
+ sp<EffectChain> ec = chains[i]; // NOLINT(performance-unnecessary-copy-initialization)
int sessionid = ec->sessionId();
sp<ThreadBase> t = ec->thread().promote();
if (t == 0) {
@@ -3715,7 +3722,7 @@
PlaybackThread *thread = primaryPlaybackThread_l();
if (thread == NULL) {
- return DeviceTypeSet();
+ return {};
}
return thread->outDeviceTypes();
@@ -4227,7 +4234,7 @@
// session and used it instead of creating a new one.
sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
if (chain != 0) {
- Mutex::Autolock _l(thread->mLock);
+ Mutex::Autolock _l2(thread->mLock);
thread->addEffectChain_l(chain);
}
}
@@ -4345,6 +4352,7 @@
status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId,
AudioFlinger::PlaybackThread *srcThread,
AudioFlinger::PlaybackThread *dstThread)
+NO_THREAD_SAFETY_ANALYSIS // requires srcThread and dstThread locks
{
ALOGV("moveEffectChain_l() session %d from thread %p to thread %p",
sessionId, srcThread, dstThread);
@@ -4374,11 +4382,12 @@
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
sp<EffectChain> dstChain;
- sp<EffectModule> effect = chain->getEffectFromId_l(0);
Vector< sp<EffectModule> > removed;
status_t status = NO_ERROR;
std::string errorString;
- while (effect != nullptr) {
+ // process effects one by one.
+ for (sp<EffectModule> effect = chain->getEffectFromId_l(0); effect != nullptr;
+ effect = chain->getEffectFromId_l(0)) {
srcThread->removeEffect_l(effect);
removed.add(effect);
status = dstThread->addEffect_l(effect);
@@ -4399,7 +4408,6 @@
break;
}
}
- effect = chain->getEffectFromId_l(0);
}
size_t restored = 0;
@@ -4503,6 +4511,7 @@
}
bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l()
+NO_THREAD_SAFETY_ANALYSIS // thread lock for getEffectChain_l.
{
if (mGlobalEffectEnableTime != 0 &&
((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) {
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index ebfe32c..364b033 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -328,12 +328,12 @@
static void onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration);
status_t addEffectToHal(audio_port_handle_t deviceId,
- audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect);
+ audio_module_handle_t hwModuleId, const sp<EffectHalInterface>& effect);
status_t removeEffectFromHal(audio_port_handle_t deviceId,
- audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect);
+ audio_module_handle_t hwModuleId, const sp<EffectHalInterface>& effect);
void updateDownStreamPatches_l(const struct audio_patch *patch,
- const std::set<audio_io_handle_t> streams);
+ const std::set<audio_io_handle_t>& streams);
std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l();
@@ -376,7 +376,7 @@
audio_session_t triggerSession,
audio_session_t listenerSession,
sync_event_callback_t callBack,
- wp<RefBase> cookie)
+ const wp<RefBase>& cookie)
: mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession),
mCallback(callBack), mCookie(cookie)
{}
@@ -848,7 +848,7 @@
void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices);
void forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
- std::function<bool(const sp<PlaybackThread>&)> useThread = nullptr);
+ const std::function<bool(const sp<PlaybackThread>&)>& useThread = nullptr);
// AudioStreamIn is immutable, so their fields are const.
// For emphasis, we could also make all pointers to them be "const *",
@@ -861,7 +861,8 @@
sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); }
- AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) :
+ AudioStreamIn(AudioHwDevice *dev, const sp<StreamInHalInterface>& in,
+ audio_input_flags_t flags) :
audioHwDev(dev), stream(in), flags(flags) {}
status_t read(void *buffer, size_t bytes, size_t *read) override {
return stream->read(buffer, bytes, read);
diff --git a/services/audioflinger/AudioHwDevice.h b/services/audioflinger/AudioHwDevice.h
index 1749f3f..d071922 100644
--- a/services/audioflinger/AudioHwDevice.h
+++ b/services/audioflinger/AudioHwDevice.h
@@ -46,7 +46,7 @@
AudioHwDevice(audio_module_handle_t handle,
const char *moduleName,
- sp<DeviceHalInterface> hwDevice,
+ const sp<DeviceHalInterface>& hwDevice,
Flags flags)
: mHandle(handle)
, mModuleName(strdup(moduleName))
diff --git a/services/audioflinger/AutoPark.h b/services/audioflinger/AutoPark.h
index 9ac7b65..83f6b7d 100644
--- a/services/audioflinger/AutoPark.h
+++ b/services/audioflinger/AutoPark.h
@@ -58,4 +58,4 @@
FastThreadState::Command mPreviousCommand;
}; // class AutoPark
-} // namespace
+} // namespace android
diff --git a/services/audioflinger/DeviceEffectManager.cpp b/services/audioflinger/DeviceEffectManager.cpp
index 3a8c1bc..8484f95 100644
--- a/services/audioflinger/DeviceEffectManager.cpp
+++ b/services/audioflinger/DeviceEffectManager.cpp
@@ -160,7 +160,9 @@
return status;
}
-void AudioFlinger::DeviceEffectManager::dump(int fd) {
+void AudioFlinger::DeviceEffectManager::dump(int fd)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
+{
const bool locked = dumpTryLock(mLock);
if (!locked) {
String8 result("DeviceEffectManager may be deadlocked\n");
@@ -264,7 +266,7 @@
return false;
}
-void AudioFlinger::DeviceEffectManager::CommandThread::sendCommand(sp<Command> command) {
+void AudioFlinger::DeviceEffectManager::CommandThread::sendCommand(const sp<Command>& command) {
Mutex::Autolock _l(mLock);
mCommands.push_back(command);
mWaitWorkCV.signal();
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index d2faa70..493800e 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -47,11 +47,11 @@
int32_t sessionId, int32_t deviceId,
sp<EffectHalInterface> *effect);
status_t addEffectToHal(audio_port_handle_t deviceId, audio_module_handle_t hwModuleId,
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
return mAudioFlinger.addEffectToHal(deviceId, hwModuleId, effect);
};
status_t removeEffectFromHal(audio_port_handle_t deviceId, audio_module_handle_t hwModuleId,
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
return mAudioFlinger.removeEffectFromHal(deviceId, hwModuleId, effect);
};
@@ -73,7 +73,7 @@
RELEASE_AUDIO_PATCH,
};
- CommandThread(DeviceEffectManager& manager)
+ explicit CommandThread(DeviceEffectManager& manager)
: Thread(false), mManager(manager) {}
~CommandThread() override;
@@ -94,7 +94,7 @@
class Command: public RefBase {
public:
Command() = default;
- Command(int command, sp<CommandData> data)
+ Command(int command, const sp<CommandData>& data)
: mCommand(command), mData(data) {}
int mCommand = -1;
@@ -117,13 +117,13 @@
class ReleaseAudioPatchData : public CommandData {
public:
- ReleaseAudioPatchData(audio_patch_handle_t handle)
+ explicit ReleaseAudioPatchData(audio_patch_handle_t handle)
: mHandle(handle) {}
audio_patch_handle_t mHandle;
};
- void sendCommand(sp<Command> command);
+ void sendCommand(const sp<Command>& command);
Mutex mLock;
Condition mWaitWorkCV;
@@ -145,7 +145,7 @@
class DeviceEffectManagerCallback : public EffectCallbackInterface {
public:
- DeviceEffectManagerCallback(DeviceEffectManager *manager)
+ explicit DeviceEffectManagerCallback(DeviceEffectManager *manager)
: mManager(*manager) {}
status_t createEffectHal(const effect_uuid_t *pEffectUuid,
@@ -176,10 +176,10 @@
size_t frameCount() const override { return 0; }
uint32_t latency() const override { return 0; }
- status_t addEffectToHal(sp<EffectHalInterface> effect __unused) override {
+ status_t addEffectToHal(const sp<EffectHalInterface>& /* effect */) override {
return NO_ERROR;
}
- status_t removeEffectFromHal(sp<EffectHalInterface> effect __unused) override {
+ status_t removeEffectFromHal(const sp<EffectHalInterface>& /* effect */) override {
return NO_ERROR;
}
@@ -204,11 +204,11 @@
int newEffectId() { return mManager.audioFlinger().nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); }
status_t addEffectToHal(audio_port_handle_t deviceId,
- audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ audio_module_handle_t hwModuleId, const sp<EffectHalInterface>& effect) {
return mManager.addEffectToHal(deviceId, hwModuleId, effect);
}
status_t removeEffectFromHal(audio_port_handle_t deviceId,
- audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ audio_module_handle_t hwModuleId, const sp<EffectHalInterface>& effect) {
return mManager.removeEffectFromHal(deviceId, hwModuleId, effect);
}
private:
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 98829d0..e912bff 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -498,6 +498,7 @@
}
void AudioFlinger::EffectBase::dump(int fd, const Vector<String16>& args __unused)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
String8 result;
@@ -1249,13 +1250,13 @@
return -EINVAL;
}
if (cmdCode == EFFECT_CMD_GET_PARAM &&
- (maxReplySize < sizeof(effect_param_t) ||
+ (maxReplySize < static_cast<signed>(sizeof(effect_param_t)) ||
param->psize > maxReplySize - sizeof(effect_param_t))) {
android_errorWriteLog(0x534e4554, "29251553");
return -EINVAL;
}
if (cmdCode == EFFECT_CMD_GET_PARAM &&
- (sizeof(effect_param_t) > maxReplySize
+ (static_cast<signed>(sizeof(effect_param_t)) > maxReplySize
|| param->psize > maxReplySize - sizeof(effect_param_t)
|| param->vsize > maxReplySize - sizeof(effect_param_t)
- param->psize
@@ -1685,6 +1686,7 @@
}
void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
EffectBase::dump(fd, args);
@@ -1939,7 +1941,7 @@
}
mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
// Client destructor must run with AudioFlinger client mutex locked
- Mutex::Autolock _l(mClient->audioFlinger()->mClientLock);
+ Mutex::Autolock _l2(mClient->audioFlinger()->mClientLock);
mClient.clear();
}
}
@@ -2003,14 +2005,14 @@
}
if (cmdCode == EFFECT_CMD_ENABLE) {
- if (maxResponseSize < sizeof(int)) {
+ if (maxResponseSize < static_cast<signed>(sizeof(int))) {
android_errorWriteLog(0x534e4554, "32095713");
RETURN(BAD_VALUE);
}
writeToBuffer(NO_ERROR, response);
return enable(_aidl_return);
} else if (cmdCode == EFFECT_CMD_DISABLE) {
- if (maxResponseSize < sizeof(int)) {
+ if (maxResponseSize < static_cast<signed>(sizeof(int))) {
android_errorWriteLog(0x534e4554, "32095713");
RETURN(BAD_VALUE);
}
@@ -2034,7 +2036,7 @@
RETURN(INVALID_OPERATION);
}
- if (maxResponseSize < sizeof(int)) {
+ if (maxResponseSize < (signed)sizeof(int)) {
android_errorWriteLog(0x534e4554, "32095713");
RETURN(BAD_VALUE);
}
@@ -2043,7 +2045,7 @@
// No need to trylock() here as this function is executed in the binder thread serving a
// particular client process: no risk to block the whole media server process or mixer
// threads if we are stuck here
- Mutex::Autolock _l(mCblk->lock);
+ Mutex::Autolock _l2(mCblk->lock);
// keep local copy of index in case of client corruption b/32220769
const uint32_t clientIndex = mCblk->clientIndex;
const uint32_t serverIndex = mCblk->serverIndex;
@@ -2146,6 +2148,7 @@
}
void AudioFlinger::EffectHandle::dumpToBuffer(char* buffer, size_t size)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
@@ -2400,7 +2403,7 @@
}
} else {
effect->setInBuffer(mInBuffer);
- if (idx_insert == previousSize) {
+ if (idx_insert == static_cast<ssize_t>(previousSize)) {
if (idx_insert != 0) {
mEffects[idx_insert-1]->configure();
mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
@@ -2460,7 +2463,7 @@
}
// remember position of first insert effect and by default
// select this as insert position for new effect
- if (idx_insert == size) {
+ if (idx_insert == static_cast<ssize_t>(size)) {
idx_insert = i;
}
// remember position of last insert effect claiming
@@ -2690,6 +2693,7 @@
}
void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
String8 result;
@@ -3027,7 +3031,7 @@
}
status_t AudioFlinger::EffectChain::EffectCallback::addEffectToHal(
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
status_t result = NO_INIT;
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
@@ -3043,7 +3047,7 @@
}
status_t AudioFlinger::EffectChain::EffectCallback::removeEffectFromHal(
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
status_t result = NO_INIT;
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
@@ -3185,15 +3189,20 @@
return t->frameCount();
}
-uint32_t AudioFlinger::EffectChain::EffectCallback::latency() const {
+uint32_t AudioFlinger::EffectChain::EffectCallback::latency() const
+NO_THREAD_SAFETY_ANALYSIS // latency_l() access
+{
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
return 0;
}
+ // TODO(b/275956781) - this requires the thread lock.
return t->latency_l();
}
-void AudioFlinger::EffectChain::EffectCallback::setVolumeForOutput(float left, float right) const {
+void AudioFlinger::EffectChain::EffectCallback::setVolumeForOutput(float left, float right) const
+NO_THREAD_SAFETY_ANALYSIS // setVolumeForOutput_l() access
+{
sp<ThreadBase> t = thread().promote();
if (t == nullptr) {
return;
@@ -3437,7 +3446,7 @@
}
status_t AudioFlinger::DeviceEffectProxy::addEffectToHal(
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
if (mHalEffect == nullptr) {
return NO_INIT;
}
@@ -3446,7 +3455,7 @@
}
status_t AudioFlinger::DeviceEffectProxy::removeEffectFromHal(
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
if (mHalEffect == nullptr) {
return NO_INIT;
}
@@ -3484,7 +3493,9 @@
return audio_channel_count_from_in_mask(channelMask());
}
-void AudioFlinger::DeviceEffectProxy::dump(int fd, int spaces) {
+void AudioFlinger::DeviceEffectProxy::dump(int fd, int spaces)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
+{
const Vector<String16> args;
EffectBase::dump(fd, args);
@@ -3563,7 +3574,7 @@
}
status_t AudioFlinger::DeviceEffectProxy::ProxyCallback::addEffectToHal(
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
sp<DeviceEffectProxy> proxy = mProxy.promote();
if (proxy == nullptr) {
return NO_INIT;
@@ -3572,7 +3583,7 @@
}
status_t AudioFlinger::DeviceEffectProxy::ProxyCallback::removeEffectFromHal(
- sp<EffectHalInterface> effect) {
+ const sp<EffectHalInterface>& effect) {
sp<DeviceEffectProxy> proxy = mProxy.promote();
if (proxy == nullptr) {
return NO_INIT;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 78788df..bad86bc 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -45,8 +45,8 @@
// Non trivial methods usually implemented with help from ThreadBase:
// pay attention to mutex locking order
virtual uint32_t latency() const { return 0; }
- virtual status_t addEffectToHal(sp<EffectHalInterface> effect) = 0;
- virtual status_t removeEffectFromHal(sp<EffectHalInterface> effect) = 0;
+ virtual status_t addEffectToHal(const sp<EffectHalInterface>& effect) = 0;
+ virtual status_t removeEffectFromHal(const sp<EffectHalInterface>& effect) = 0;
virtual void setVolumeForOutput(float left, float right) const = 0;
virtual bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) = 0;
virtual void checkSuspendOnEffectEnabled(const sp<EffectBase>& effect,
@@ -159,8 +159,8 @@
bool isPinned() const { return mPinned; }
void unPin() { mPinned = false; }
- void lock() { mLock.lock(); }
- void unlock() { mLock.unlock(); }
+ void lock() ACQUIRE(mLock) { mLock.lock(); }
+ void unlock() RELEASE(mLock) { mLock.unlock(); }
status_t updatePolicyState();
@@ -459,10 +459,10 @@
void process_l();
- void lock() {
+ void lock() ACQUIRE(mLock) {
mLock.lock();
}
- void unlock() {
+ void unlock() RELEASE(mLock) {
mLock.unlock();
}
@@ -605,8 +605,8 @@
size_t frameCount() const override;
uint32_t latency() const override;
- status_t addEffectToHal(sp<EffectHalInterface> effect) override;
- status_t removeEffectFromHal(sp<EffectHalInterface> effect) override;
+ status_t addEffectToHal(const sp<EffectHalInterface>& effect) override;
+ status_t removeEffectFromHal(const sp<EffectHalInterface>& effect) override;
bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) override;
void setVolumeForOutput(float left, float right) const override;
@@ -721,8 +721,8 @@
size_t removeEffect(const sp<EffectModule>& effect);
- status_t addEffectToHal(sp<EffectHalInterface> effect);
- status_t removeEffectFromHal(sp<EffectHalInterface> effect);
+ status_t addEffectToHal(const sp<EffectHalInterface>& effect);
+ status_t removeEffectFromHal(const sp<EffectHalInterface>& effect);
const AudioDeviceTypeAddr& device() { return mDevice; };
bool isOutput() const;
@@ -766,8 +766,8 @@
size_t frameCount() const override { return 0; }
uint32_t latency() const override { return 0; }
- status_t addEffectToHal(sp<EffectHalInterface> effect) override;
- status_t removeEffectFromHal(sp<EffectHalInterface> effect) override;
+ status_t addEffectToHal(const sp<EffectHalInterface>& effect) override;
+ status_t removeEffectFromHal(const sp<EffectHalInterface>& effect) override;
bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) override;
void setVolumeForOutput(float left __unused, float right __unused) const override {}
diff --git a/services/audioflinger/FastCaptureDumpState.cpp b/services/audioflinger/FastCaptureDumpState.cpp
index b8b3866..243dfa5 100644
--- a/services/audioflinger/FastCaptureDumpState.cpp
+++ b/services/audioflinger/FastCaptureDumpState.cpp
@@ -51,4 +51,4 @@
periodSec * 1e3, mSilenced ? "true" : "false");
}
-} // android
+} // namespace android
diff --git a/services/audioflinger/FastCaptureDumpState.h b/services/audioflinger/FastCaptureDumpState.h
index a1b8706..34ce456 100644
--- a/services/audioflinger/FastCaptureDumpState.h
+++ b/services/audioflinger/FastCaptureDumpState.h
@@ -38,6 +38,6 @@
bool mSilenced = false; // capture is silenced
};
-} // android
+} // namespace android
#endif // ANDROID_AUDIO_FAST_CAPTURE_DUMP_STATE_H
diff --git a/services/audioflinger/FastCaptureState.cpp b/services/audioflinger/FastCaptureState.cpp
index c4d5e45..918ba9c 100644
--- a/services/audioflinger/FastCaptureState.cpp
+++ b/services/audioflinger/FastCaptureState.cpp
@@ -42,4 +42,4 @@
LOG_ALWAYS_FATAL("%s", __func__);
}
-} // android
+} // namespace android
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
index 3f20282..d041882 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -203,4 +203,4 @@
}
}
-} // android
+} // namespace android
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 9b91cbc..294ef78 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -81,6 +81,6 @@
TimestampVerifier<int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
};
-} // android
+} // namespace android
#endif // ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
diff --git a/services/audioflinger/FastThread.h b/services/audioflinger/FastThread.h
index 3f6b206..2f0f73f 100644
--- a/services/audioflinger/FastThread.h
+++ b/services/audioflinger/FastThread.h
@@ -92,6 +92,6 @@
}; // class FastThread
-} // android
+} // namespace android
#endif // ANDROID_AUDIO_FAST_THREAD_H
diff --git a/services/audioflinger/FastThreadDumpState.cpp b/services/audioflinger/FastThreadDumpState.cpp
index 964a725..e91073f 100644
--- a/services/audioflinger/FastThreadDumpState.cpp
+++ b/services/audioflinger/FastThreadDumpState.cpp
@@ -56,4 +56,4 @@
}
#endif
-} // android
+} // namespace android
diff --git a/services/audioflinger/FastThreadDumpState.h b/services/audioflinger/FastThreadDumpState.h
index 1ce0914..0b20e55 100644
--- a/services/audioflinger/FastThreadDumpState.h
+++ b/services/audioflinger/FastThreadDumpState.h
@@ -67,6 +67,6 @@
}; // struct FastThreadDumpState
-} // android
+} // namespace android
#endif // ANDROID_AUDIO_FAST_THREAD_DUMP_STATE_H
diff --git a/services/audioflinger/FastThreadState.h b/services/audioflinger/FastThreadState.h
index 54c0dc6..9fb4e06 100644
--- a/services/audioflinger/FastThreadState.h
+++ b/services/audioflinger/FastThreadState.h
@@ -50,6 +50,6 @@
static const char *commandToString(Command command);
}; // struct FastThreadState
-} // android
+} // namespace android
#endif // ANDROID_AUDIO_FAST_THREAD_STATE_H
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index 68a3800..5555766 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -199,7 +199,7 @@
return mRecord.handle() != AUDIO_PATCH_HANDLE_NONE ||
mPlayback.handle() != AUDIO_PATCH_HANDLE_NONE; }
- void setThread(sp<ThreadBase> thread) { mThread = thread; }
+ void setThread(const sp<ThreadBase>& thread) { mThread = thread; }
wp<ThreadBase> thread() const { return mThread; }
// returns the latency of the patch (from record to playback).
diff --git a/services/audioflinger/StateQueue.cpp b/services/audioflinger/StateQueue.cpp
index 9d4188f..38ce2c2 100644
--- a/services/audioflinger/StateQueue.cpp
+++ b/services/audioflinger/StateQueue.cpp
@@ -187,7 +187,9 @@
} // namespace android
-// hack for gcc
+// Hack to avoid explicit template instantiation of
+// template class StateQueue<FastCaptureState>;
+// template class StateQueue<FastMixerState>;
#ifdef STATE_QUEUE_INSTANTIATIONS
-#include STATE_QUEUE_INSTANTIATIONS
+#include STATE_QUEUE_INSTANTIATIONS // NOLINT(bugprone-suspicious-include)
#endif
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 16c72cc..6956e86 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -374,7 +374,7 @@
// try three times to get the clock offset, choose the one
// with the minimum gap in measurements.
const int tries = 3;
- nsecs_t bestGap, measured;
+ nsecs_t bestGap = 0, measured = 0; // not required, initialized for clang-tidy
for (int i = 0; i < tries; ++i) {
const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC);
const nsecs_t tbase = systemTime(clockbase);
@@ -623,6 +623,7 @@
// sendConfigEvent_l() must be called with ThreadBase::mLock held
// Can temporarily release the lock if waiting for a reply from processConfigEvents_l().
status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
+NO_THREAD_SAFETY_ANALYSIS // condition variable
{
status_t status = NO_ERROR;
@@ -936,6 +937,7 @@
}
void AudioFlinger::ThreadBase::dump(int fd, const Vector<String16>& args)
+NO_THREAD_SAFETY_ANALYSIS // conditional try lock
{
dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input",
this, mThreadName, getTid(), type(), threadTypeToString(type()));
@@ -1304,7 +1306,9 @@
void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
audio_session_t sessionId,
- bool threadLocked) {
+ bool threadLocked)
+NO_THREAD_SAFETY_ANALYSIS // manual locking
+{
if (!threadLocked) {
mLock.lock();
}
@@ -1768,6 +1772,7 @@
void AudioFlinger::ThreadBase::lockEffectChains_l(
Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::lock()
{
effectChains = mEffectChains;
for (size_t i = 0; i < mEffectChains.size(); i++) {
@@ -1777,6 +1782,7 @@
void AudioFlinger::ThreadBase::unlockEffectChains(
const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
+NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::unlock()
{
for (size_t i = 0; i < effectChains.size(); i++) {
effectChains[i]->unlock();
@@ -1884,7 +1890,7 @@
template <typename T>
void AudioFlinger::ThreadBase::ActiveTracks<T>::updatePowerState(
- sp<ThreadBase> thread, bool force) {
+ const sp<ThreadBase>& thread, bool force) {
// Updates ActiveTracks client uids to the thread wakelock.
if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) {
thread->updateWakeLockUids_l(getWakeLockUids());
@@ -2716,6 +2722,7 @@
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
+NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
status_t status = ALREADY_EXISTS;
@@ -2868,7 +2875,7 @@
if (initCheck() == NO_ERROR && mOutput->stream->getParameters(keys, &out_s8) == OK) {
return out_s8;
}
- return String8();
+ return {};
}
status_t AudioFlinger::DirectOutputThread::selectPresentation(int presentationId, int programId) {
@@ -3578,10 +3585,10 @@
size_t numSamples = mNormalFrameCount
* (audio_channel_count_from_out_mask(mMixerChannelMask)
+ mHapticChannelCount);
- status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
+ const status_t allocateStatus = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
numSamples * sizeof(effect_buffer_t),
&halInBuffer);
- if (result != OK) return result;
+ if (allocateStatus != OK) return allocateStatus;
#ifdef FLOAT_EFFECT_CHAIN
buffer = halInBuffer ? halInBuffer->audioBuffer()->f32 : buffer;
#else
@@ -3667,8 +3674,8 @@
}
// detach all tracks with same session ID from this chain
- for (size_t i = 0; i < mTracks.size(); ++i) {
- sp<Track> track = mTracks[i];
+ for (size_t j = 0; j < mTracks.size(); ++j) {
+ sp<Track> track = mTracks[j];
if (session == track->sessionId()) {
track->setMainBuffer(reinterpret_cast<effect_buffer_t*>(mSinkBuffer));
chain->decTrackCnt();
@@ -3721,6 +3728,7 @@
}
bool AudioFlinger::PlaybackThread::threadLoop()
+NO_THREAD_SAFETY_ANALYSIS // manual locking of AudioFlinger
{
tlNBLogWriter = mNBLogWriter.get();
@@ -3789,7 +3797,7 @@
// is more informational.
if (mAudioFlinger->mLock.tryLock() == NO_ERROR) {
std::vector<PatchPanel::SoftwarePatch> swPatches;
- double latencyMs;
+ double latencyMs = 0.; // not required; initialized for clang-tidy
status_t status = INVALID_OPERATION;
audio_patch_handle_t downstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (mAudioFlinger->mPatchPanel.getDownstreamSoftwarePatches(id(), &swPatches) == OK
@@ -3809,8 +3817,7 @@
ALOGVV("new downstream latency %lf ms", latencyMs);
} else {
ALOGD("out of range downstream latency %lf ms", latencyMs);
- if (latencyMs < minLatency) latencyMs = minLatency;
- else if (latencyMs > maxLatency) latencyMs = maxLatency;
+ latencyMs = std::clamp(latencyMs, minLatency, maxLatency);
}
mDownstreamLatencyStatMs.add(latencyMs);
}
@@ -4492,6 +4499,7 @@
// removeTracks_l() must be called with ThreadBase::mLock held
void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove)
+NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
for (const auto& track : tracksToRemove) {
mActiveTracks.remove(track);
@@ -4612,8 +4620,8 @@
"as it does not support audio patches",
patch->sinks[i].ext.device.type);
type = static_cast<audio_devices_t>(type | patch->sinks[i].ext.device.type);
- deviceTypeAddrs.push_back(AudioDeviceTypeAddr(patch->sinks[i].ext.device.type,
- patch->sinks[i].ext.device.address));
+ deviceTypeAddrs.emplace_back(patch->sinks[i].ext.device.type,
+ patch->sinks[i].ext.device.address);
}
audio_port_handle_t sinkPortId = patch->sinks[0].id;
@@ -4821,14 +4829,15 @@
// When it wakes up after a maximum latency, it runs a few cycles quickly before
// finally blocking. Note the pipe implementation rounds up the request to a power of 2.
MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
- const NBAIO_Format offers[1] = {format};
- size_t numCounterOffers = 0;
+ const NBAIO_Format offersFast[1] = {format};
+ size_t numCounterOffersFast = 0;
#if !LOG_NDEBUG
ssize_t index =
#else
(void)
#endif
- monoPipe->negotiate(offers, 1, NULL, numCounterOffers);
+ monoPipe->negotiate(offersFast, std::size(offersFast),
+ nullptr /* counterOffers */, numCounterOffersFast);
ALOG_ASSERT(index == 0);
monoPipe->setAvgFrames((mScreenState & 1) ?
(monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
@@ -5240,7 +5249,7 @@
// tallyUnderrunFrames() is called to update the track counters
// with the number of underrun frames for a particular mixer period.
// We defer tallying until we know the final mixer status.
- void tallyUnderrunFrames(sp<Track> track, size_t underrunFrames) {
+ void tallyUnderrunFrames(const sp<Track>& track, size_t underrunFrames) {
mUnderrunFrames.emplace_back(track, underrunFrames);
}
@@ -5489,7 +5498,7 @@
// during last round
size_t desiredFrames;
const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate();
- AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
+ const AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate();
desiredFrames = sourceFramesNeededWithTimestretch(
sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed);
@@ -5673,12 +5682,12 @@
AudioMixer::SAMPLE_RATE,
(void *)(uintptr_t)reqSampleRate);
- AudioPlaybackRate playbackRate = proxy->getPlaybackRate();
mAudioMixer->setParameter(
trackId,
AudioMixer::TIMESTRETCH,
AudioMixer::PLAYBACK_RATE,
- &playbackRate);
+ // cast away constness for this generic API.
+ const_cast<void *>(reinterpret_cast<const void *>(&playbackRate)));
/*
* Select the appropriate output buffer for the track.
@@ -6062,12 +6071,12 @@
mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate);
for (const auto &track : mTracks) {
const int trackId = track->id();
- status_t status = mAudioMixer->create(
+ const status_t createStatus = mAudioMixer->create(
trackId,
track->mChannelMask,
track->mFormat,
track->mSessionId);
- ALOGW_IF(status != NO_ERROR,
+ ALOGW_IF(createStatus != NO_ERROR,
"%s(): AudioMixer cannot create track(%d)"
" mask %#x, format %#x, sessionId %d",
__func__,
@@ -7114,7 +7123,7 @@
void AudioFlinger::DuplicatingThread::threadLoop_mix()
{
// mix buffers...
- if (outputsReady(outputTracks)) {
+ if (outputsReady()) {
mAudioMixer->process();
} else {
if (mMixerBufferValid) {
@@ -7185,7 +7194,7 @@
}
}
-void AudioFlinger::DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::DuplicatingThread::dumpInternals_l(int fd, const Vector<String16>& args)
{
MixerThread::dumpInternals_l(fd, args);
@@ -7289,9 +7298,7 @@
}
}
-
-bool AudioFlinger::DuplicatingThread::outputsReady(
- const SortedVector< sp<OutputTrack> > &outputTracks)
+bool AudioFlinger::DuplicatingThread::outputsReady()
{
for (size_t i = 0; i < outputTracks.size(); i++) {
sp<ThreadBase> thread = outputTracks[i]->thread().promote();
@@ -7620,14 +7627,16 @@
// pipe will be shared directly with fast clients, so clear to avoid leaking old information
memset(pipeBuffer, 0, pipeSize);
Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer);
- const NBAIO_Format offers[1] = {format};
- size_t numCounterOffers = 0;
- [[maybe_unused]] ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
+ const NBAIO_Format offersFast[1] = {format};
+ size_t numCounterOffersFast = 0;
+ [[maybe_unused]] ssize_t index = pipe->negotiate(offersFast, std::size(offersFast),
+ nullptr /* counterOffers */, numCounterOffersFast);
ALOG_ASSERT(index == 0);
mPipeSink = pipe;
PipeReader *pipeReader = new PipeReader(*pipe);
- numCounterOffers = 0;
- index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
+ numCounterOffersFast = 0;
+ index = pipeReader->negotiate(offersFast, std::size(offersFast),
+ nullptr /* counterOffers */, numCounterOffersFast);
ALOG_ASSERT(index == 0);
mPipeSource = pipeReader;
mPipeFramesP2 = pipeFramesP2;
@@ -7975,7 +7984,7 @@
// copy to the right place. Permitted because mRsmpInBuffer was over-allocated.
int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1);
- ssize_t framesRead;
+ ssize_t framesRead = 0; // not needed, remove clang-tidy warning.
const int64_t lastIoBeginNs = systemTime(); // start IO timing
// If an NBAIO source is present, use it to read the normal capture's data
@@ -8168,8 +8177,9 @@
// straight from RecordThread buffer to RecordTrack buffer.
AudioBufferProvider::Buffer buffer;
buffer.frameCount = framesOut;
- status_t status = activeTrack->mResamplerBufferProvider->getNextBuffer(&buffer);
- if (status == OK && buffer.frameCount != 0) {
+ const status_t getNextBufferStatus =
+ activeTrack->mResamplerBufferProvider->getNextBuffer(&buffer);
+ if (getNextBufferStatus == OK && buffer.frameCount != 0) {
ALOGV_IF(buffer.frameCount != framesOut,
"%s() read less than expected (%zu vs %zu)",
__func__, buffer.frameCount, framesOut);
@@ -8179,7 +8189,7 @@
} else {
framesOut = 0;
ALOGE("%s() cannot fill request, status: %d, frameCount: %zu",
- __func__, status, buffer.frameCount);
+ __func__, getNextBufferStatus, buffer.frameCount);
}
} else {
// process frames from the RecordThread buffer provider to the RecordTrack
@@ -8599,7 +8609,6 @@
// or using a separate command thread
recordTrack->mState = TrackBase::STARTING_1;
mActiveTracks.add(recordTrack);
- status_t status = NO_ERROR;
if (recordTrack->isExternalTrack()) {
mLock.unlock();
status = AudioSystem::startInput(recordTrack->portId());
@@ -8790,7 +8799,7 @@
// "best effort" behavior of the API.
if (sharedOffset < 0) {
sharedAudioStartFrames = mRsmpInRear;
- } else if (sharedOffset > mRsmpInFrames) {
+ } else if (sharedOffset > static_cast<signed>(mRsmpInFrames)) {
sharedAudioStartFrames =
audio_utils::safe_sub_overflow(mRsmpInRear, (int32_t)mRsmpInFrames);
}
@@ -9165,7 +9174,7 @@
return out_s8;
}
}
- return String8();
+ return {};
}
void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
@@ -9400,7 +9409,7 @@
maxFilled = filled;
}
}
- if (maxFilled > mRsmpInFrames) {
+ if (maxFilled > static_cast<signed>(mRsmpInFrames)) {
(void)__builtin_sub_overflow(mRsmpInRear, mRsmpInFrames, &oldestFront);
}
return oldestFront;
@@ -9592,7 +9601,7 @@
AudioFlinger::MmapThread::MmapThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, sp<StreamHalInterface> stream, bool systemReady, bool isOut)
+ AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady, bool isOut)
: ThreadBase(audioFlinger, id, (isOut ? MMAP_PLAYBACK : MMAP_CAPTURE), systemReady, isOut),
mSessionId(AUDIO_SESSION_NONE),
mPortId(AUDIO_PORT_HANDLE_NONE),
@@ -9797,8 +9806,10 @@
mHalVolFloat = -1.0f;
} else if (!track->isSilenced_l()) {
for (const sp<MmapTrack> &t : mActiveTracks) {
- if (t->isSilenced_l() && t->uid() != client.attributionSource.uid)
+ if (t->isSilenced_l()
+ && t->uid() != static_cast<uid_t>(client.attributionSource.uid)) {
t->invalidate();
+ }
}
}
@@ -10024,7 +10035,7 @@
if (initCheck() == NO_ERROR && mHalStream->getParameters(keys, &out_s8) == OK) {
return out_s8;
}
- return String8();
+ return {};
}
void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
@@ -10054,6 +10065,7 @@
status_t AudioFlinger::MmapThread::createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle)
+NO_THREAD_SAFETY_ANALYSIS // elease and re-acquire mLock
{
status_t status = NO_ERROR;
@@ -10071,8 +10083,8 @@
"as it does not support audio patches",
patch->sinks[i].ext.device.type);
type = static_cast<audio_devices_t>(type | patch->sinks[i].ext.device.type);
- sinkDeviceTypeAddrs.push_back(AudioDeviceTypeAddr(patch->sinks[i].ext.device.type,
- patch->sinks[i].ext.device.address));
+ sinkDeviceTypeAddrs.emplace_back(patch->sinks[i].ext.device.type,
+ patch->sinks[i].ext.device.address);
}
deviceId = patch->sinks[0].id;
numDevices = mPatch.num_sinks;
@@ -10275,6 +10287,7 @@
}
void AudioFlinger::MmapThread::checkInvalidTracks_l()
+NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
sp<MmapStreamCallback> callback;
for (const sp<MmapTrack> &track : mActiveTracks) {
@@ -10429,6 +10442,7 @@
}
void AudioFlinger::MmapPlaybackThread::processVolume_l()
+NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
{
float volume;
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index ce90767..63ad4e6 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -163,7 +163,7 @@
class SetParameterConfigEventData : public ConfigEventData {
public:
- explicit SetParameterConfigEventData(String8 keyValuePairs) :
+ explicit SetParameterConfigEventData(const String8& keyValuePairs) :
mKeyValuePairs(keyValuePairs) {}
virtual void dump(char *buffer, size_t size) {
@@ -175,7 +175,7 @@
class SetParameterConfigEvent : public ConfigEvent {
public:
- explicit SetParameterConfigEvent(String8 keyValuePairs) :
+ explicit SetParameterConfigEvent(const String8& keyValuePairs) :
ConfigEvent(CFG_EVENT_SET_PARAMETER) {
mData = new SetParameterConfigEventData(keyValuePairs);
mWaitStatus = true;
@@ -789,7 +789,7 @@
// ThreadBase thread.
void clear();
// periodically called in the threadLoop() to update power state uids.
- void updatePowerState(sp<ThreadBase> thread, bool force = false);
+ void updatePowerState(const sp<ThreadBase>& thread, bool force = false);
/** @return true if one or move active tracks was added or removed since the
* last time this function was called or the vector was created.
@@ -1260,7 +1260,7 @@
template <typename T>
class Tracks {
public:
- Tracks(bool saveDeletedTrackIds) :
+ explicit Tracks(bool saveDeletedTrackIds) :
mSaveDeletedTrackIds(saveDeletedTrackIds) { }
// SortedVector methods
@@ -1289,7 +1289,7 @@
return mTracks.end();
}
- size_t processDeletedTrackIds(std::function<void(int)> f) {
+ size_t processDeletedTrackIds(const std::function<void(int)>& f) {
for (const int trackId : mDeletedTrackIds) {
f(trackId);
}
@@ -1411,7 +1411,7 @@
class IsTimestampAdvancing {
public:
// The timestamp will not be checked any faster than the specified time.
- IsTimestampAdvancing(nsecs_t minimumTimeBetweenChecksNs)
+ explicit IsTimestampAdvancing(nsecs_t minimumTimeBetweenChecksNs)
: mMinimumTimeBetweenChecksNs(minimumTimeBetweenChecksNs)
{
clear();
@@ -1700,7 +1700,7 @@
void dumpInternals_l(int fd, const Vector<String16>& args) override;
private:
- bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
+ bool outputsReady();
protected:
// threadLoop snippets
virtual void threadLoop_mix();
@@ -2066,7 +2066,7 @@
#include "MmapTracks.h"
MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, sp<StreamHalInterface> stream, bool systemReady,
+ AudioHwDevice *hwDev, const sp<StreamHalInterface>& stream, bool systemReady,
bool isOut);
virtual ~MmapThread();
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 20bfbb0..42f7b47 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -122,7 +122,7 @@
* This may be called without the thread lock.
*/
virtual double bufferLatencyMs() const {
- return mServerProxy->framesReadySafe() * 1000 / sampleRate();
+ return mServerProxy->framesReadySafe() * 1000. / sampleRate();
}
/** returns whether the track supports server latency computation.
@@ -430,7 +430,7 @@
{
public:
using Timeout = std::optional<std::chrono::nanoseconds>;
- PatchTrackBase(sp<ClientProxy> proxy, const ThreadBase& thread,
+ PatchTrackBase(const sp<ClientProxy>& proxy, const ThreadBase& thread,
const Timeout& timeout);
void setPeerTimeout(std::chrono::nanoseconds timeout);
template <typename T>
diff --git a/services/audioflinger/TrackMetrics.h b/services/audioflinger/TrackMetrics.h
index 6fc70d6..ed3928a 100644
--- a/services/audioflinger/TrackMetrics.h
+++ b/services/audioflinger/TrackMetrics.h
@@ -113,7 +113,8 @@
mDeviceStartupMs.add(startupMs);
}
- void updateMinMaxVolume(int64_t durationNs, double deviceVolume) {
+ void updateMinMaxVolume_l(int64_t durationNs, double deviceVolume)
+ REQUIRES(mLock) {
if (deviceVolume > mMaxVolume) {
mMaxVolume = deviceVolume;
mMaxVolumeDurationNs = durationNs;
@@ -165,7 +166,7 @@
mDeviceTimeNs += durationNs;
mCumulativeTimeNs += durationNs;
}
- updateMinMaxVolume(durationNs, mVolume); // always update.
+ updateMinMaxVolume_l(durationNs, mVolume); // always update.
mVolume = volume;
mLastVolumeChangeTimeNs = timeNs;
}
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 300ad9f..123d5a9 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -303,7 +303,7 @@
return NO_ERROR;
}
-AudioFlinger::ThreadBase::PatchTrackBase::PatchTrackBase(sp<ClientProxy> proxy,
+AudioFlinger::ThreadBase::PatchTrackBase::PatchTrackBase(const sp<ClientProxy>& proxy,
const ThreadBase& thread,
const Timeout& timeout)
: mProxy(proxy)
@@ -1298,8 +1298,9 @@
// must be called with thread lock held
void AudioFlinger::PlaybackThread::Track::flushAck()
{
- if (!isOffloaded() && !isDirect())
+ if (!isOffloaded() && !isDirect()) {
return;
+ }
// Clear the client ring buffer so that the app can prime the buffer while paused.
// Otherwise it might not get cleared until playback is resumed and obtainBuffer() is called.
@@ -1812,23 +1813,23 @@
//To be called with thread lock held
bool AudioFlinger::PlaybackThread::Track::isResumePending() {
-
- if (mState == RESUMING)
+ if (mState == RESUMING) {
return true;
+ }
/* Resume is pending if track was stopping before pause was called */
if (mState == STOPPING_1 &&
- mResumeToStopping)
+ mResumeToStopping) {
return true;
+ }
return false;
}
//To be called with thread lock held
void AudioFlinger::PlaybackThread::Track::resumeAck() {
-
-
- if (mState == RESUMING)
+ if (mState == RESUMING) {
mState = ACTIVE;
+ }
// Other possibility of pending resume is stopping_1 state
// Do not update the state from stopping as this prevents
@@ -2094,7 +2095,10 @@
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
- pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
+ const size_t bufferSize = inBuffer.frameCount * mFrameSize;
+ pInBuffer->mBuffer = malloc(bufferSize);
+ LOG_ALWAYS_FATAL_IF(pInBuffer->mBuffer == nullptr,
+ "%s: Unable to malloc size %zu", __func__, bufferSize);
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->raw = pInBuffer->mBuffer;
memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
@@ -2258,7 +2262,7 @@
buf.mFrameCount = buffer->frameCount;
buf.mRaw = buffer->raw;
mPeerProxy->releaseBuffer(&buf);
- TrackBase::releaseBuffer(buffer);
+ TrackBase::releaseBuffer(buffer); // Note: this is the base class.
}
status_t AudioFlinger::PlaybackThread::PatchTrack::obtainBuffer(Proxy::Buffer* buffer,
@@ -2872,7 +2876,7 @@
{
void *ptr = nullptr;
(void)posix_memalign(&ptr, alignment, size);
- return std::unique_ptr<void, decltype(free)*>(ptr, free);
+ return {ptr, free};
}
AudioFlinger::RecordThread::PassthruPatchRecord::PassthruPatchRecord(