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/Threads.cpp b/services/audioflinger/Threads.cpp
index 727a6b5..76c9ad8 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -376,7 +376,7 @@
// 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);
@@ -627,6 +627,7 @@
// 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;
@@ -942,6 +943,7 @@
}
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()));
@@ -1310,7 +1312,9 @@
void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
audio_session_t sessionId,
- bool threadLocked) {
+ bool threadLocked)
+NO_THREAD_SAFETY_ANALYSIS // manual locking
+{
if (!threadLocked) {
mLock.lock();
}
@@ -1794,6 +1798,7 @@
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++) {
@@ -1803,6 +1808,7 @@
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();
@@ -1910,7 +1916,7 @@
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());
@@ -2788,6 +2794,7 @@
// 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;
@@ -2944,7 +2951,7 @@
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) {
@@ -3706,10 +3713,10 @@
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
@@ -3795,8 +3802,8 @@
}
// 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();
@@ -3849,6 +3856,7 @@
}
bool AudioFlinger::PlaybackThread::threadLoop()
+NO_THREAD_SAFETY_ANALYSIS // manual locking of AudioFlinger
{
tlNBLogWriter = mNBLogWriter.get();
@@ -3917,7 +3925,7 @@
// 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
@@ -3937,8 +3945,7 @@
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);
}
@@ -4627,6 +4634,7 @@
// 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);
@@ -4749,8 +4757,8 @@
"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;
@@ -4965,14 +4973,15 @@
// 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);
@@ -5399,7 +5408,7 @@
// 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);
}
@@ -5659,7 +5668,7 @@
// 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);
@@ -5852,12 +5861,12 @@
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.
@@ -6241,12 +6250,12 @@
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__,
@@ -7421,7 +7430,7 @@
void AudioFlinger::DuplicatingThread::threadLoop_mix()
{
// mix buffers...
- if (outputsReady(outputTracks)) {
+ if (outputsReady()) {
mAudioMixer->process();
} else {
if (mMixerBufferValid) {
@@ -7492,7 +7501,7 @@
}
}
-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);
@@ -7596,9 +7605,7 @@
}
}
-
-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();
@@ -7865,14 +7872,16 @@
// 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;
@@ -8220,7 +8229,7 @@
// 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
@@ -8413,8 +8422,9 @@
// 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);
@@ -8424,7 +8434,7 @@
} 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
@@ -8844,7 +8854,6 @@
// 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());
@@ -9035,7 +9044,7 @@
// "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);
}
@@ -9413,7 +9422,7 @@
return out_s8;
}
}
- return String8();
+ return {};
}
void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
@@ -9654,7 +9663,7 @@
maxFilled = filled;
}
}
- if (maxFilled > mRsmpInFrames) {
+ if (maxFilled > static_cast<signed>(mRsmpInFrames)) {
(void)__builtin_sub_overflow(mRsmpInRear, mRsmpInFrames, &oldestFront);
}
return oldestFront;
@@ -9850,7 +9859,7 @@
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),
@@ -10056,8 +10065,10 @@
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();
+ }
}
}
@@ -10294,7 +10305,7 @@
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,
@@ -10324,6 +10335,7 @@
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;
@@ -10341,8 +10353,8 @@
"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;
@@ -10551,6 +10563,7 @@
}
void AudioFlinger::MmapThread::checkInvalidTracks_l()
+NO_THREAD_SAFETY_ANALYSIS // release and re-acquire mLock
{
sp<MmapStreamCallback> callback;
for (const sp<MmapTrack> &track : mActiveTracks) {
@@ -10724,6 +10737,7 @@
}
void AudioFlinger::MmapPlaybackThread::processVolume_l()
+NO_THREAD_SAFETY_ANALYSIS // access of track->processMuteEvent_l
{
float volume;