Merge tag 'android-4.4.1_r1' into HEAD
Android 4.4.1 Release 1
Conflicts:
media/libmediaplayerservice/StagefrightRecorder.cpp
media/libstagefright/AudioSource.cpp
media/libstagefright/AwesomePlayer.cpp
media/libstagefright/DataSource.cpp
media/libstagefright/OMXCodec.cpp
services/audioflinger/RecordTracks.h
services/audioflinger/Threads.cpp
services/audioflinger/Threads.h
services/audioflinger/Tracks.cpp
Change-Id: Ie17f9030275faae7a0f907b8ccc3195a7657742d
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 7bc1c7d..2145e17 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -473,6 +473,7 @@
pid_t tid,
int *sessionId,
String8& name,
+ int clientUid,
status_t *status)
{
sp<PlaybackThread::Track> track;
@@ -508,6 +509,7 @@
}
pid_t pid = IPCThreadState::self()->getCallingPid();
+
client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
@@ -535,7 +537,7 @@
ALOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus);
+ channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -1590,8 +1592,11 @@
// create new record track.
// The record track uses one track in mHardwareMixerThread by convention.
+ // TODO: the uid should be passed in as a parameter to openRecord
recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
- frameCount, lSessionId, flags, tid, &lStatus);
+ frameCount, lSessionId,
+ IPCThreadState::self()->getCallingUid(),
+ flags, tid, &lStatus);
LOG_ALWAYS_FATAL_IF((recordTrack != 0) != (lStatus == NO_ERROR));
}
if (lStatus != NO_ERROR) {
@@ -2734,6 +2739,7 @@
strategy,
sessionId,
effect->id());
+ AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
}
effect = chain->getEffectFromId_l(0);
}
@@ -2748,6 +2754,7 @@
strategy,
sessionId,
removed[i]->id());
+ AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
}
}
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 97de2f6..c0c9b99 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -122,6 +122,7 @@
pid_t tid,
int *sessionId,
String8& name,
+ int clientUid,
status_t *status);
#ifdef QCOM_HARDWARE
virtual sp<IDirectTrack> createDirectTrack(
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index a2e2511..43b77f3 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -31,6 +31,7 @@
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ int uid,
IAudioFlinger::track_flags_t flags);
virtual ~Track();
@@ -165,7 +166,8 @@
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
- int sessionId);
+ int sessionId,
+ int uid);
virtual ~TimedTrack();
class TimedBuffer {
@@ -208,7 +210,8 @@
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
- int sessionId);
+ int sessionId,
+ int uid);
void timedYieldSamples_l(AudioBufferProvider::Buffer* buffer);
void timedYieldSilence_l(uint32_t numFrames,
@@ -255,7 +258,8 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount);
+ size_t frameCount,
+ int uid);
virtual ~OutputTrack();
virtual status_t start(AudioSystem::sync_event_t event =
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 4b0c4d8..8e20d5a 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -31,7 +31,8 @@
#ifdef QCOM_HARDWARE
uint32_t flags,
#endif
- int sessionId);
+ int sessionId,
+ int uid);
virtual ~RecordTrack();
virtual status_t start(AudioSystem::sync_event_t event, int triggerSession);
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 7559d4c..c315101 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -283,6 +283,7 @@
// mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are
// set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters()
mParamStatus(NO_ERROR),
+ //FIXME: mStandby should be true here. Is this some kind of hack?
mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
// mName will be set by concrete (non-virtual) subclass
@@ -521,17 +522,7 @@
void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid)
{
- if (mPowerManager == 0) {
- // use checkService() to avoid blocking if power service is not up yet
- sp<IBinder> binder =
- defaultServiceManager()->checkService(String16("power"));
- if (binder == 0) {
- ALOGW("Thread %s cannot connect to the power manager service", mName);
- } else {
- mPowerManager = interface_cast<IPowerManager>(binder);
- binder->linkToDeath(mDeathRecipient);
- }
- }
+ getPowerManager_l();
if (mPowerManager != 0) {
sp<IBinder> binder = new BBinder();
status_t status;
@@ -571,6 +562,41 @@
}
}
+void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) {
+ Mutex::Autolock _l(mLock);
+ updateWakeLockUids_l(uids);
+}
+
+void AudioFlinger::ThreadBase::getPowerManager_l() {
+
+ if (mPowerManager == 0) {
+ // use checkService() to avoid blocking if power service is not up yet
+ sp<IBinder> binder =
+ defaultServiceManager()->checkService(String16("power"));
+ if (binder == 0) {
+ ALOGW("Thread %s cannot connect to the power manager service", mName);
+ } else {
+ mPowerManager = interface_cast<IPowerManager>(binder);
+ binder->linkToDeath(mDeathRecipient);
+ }
+ }
+}
+
+void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) {
+
+ getPowerManager_l();
+ if (mWakeLockToken == NULL) {
+ ALOGE("no wake lock to update!");
+ return;
+ }
+ if (mPowerManager != 0) {
+ sp<IBinder> binder = new BBinder();
+ status_t status;
+ status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array());
+ ALOGV("acquireWakeLock_l() %s status %d", mName, status);
+ }
+}
+
void AudioFlinger::ThreadBase::clearPowerManager()
{
Mutex::Autolock _l(mLock);
@@ -1029,6 +1055,7 @@
: ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type),
mNormalFrameCount(0), mMixBuffer(NULL),
mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ mActiveTracksGeneration(0),
// mStreamTypes[] initialized in constructor body
mOutput(output),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false),
@@ -1212,6 +1239,7 @@
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
+ int uid,
status_t *status)
{
sp<Track> track;
@@ -1358,10 +1386,10 @@
if (!isTimed) {
track = new Track(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId, *flags);
+ channelMask, frameCount, sharedBuffer, sessionId, uid, *flags);
} else {
track = TimedTrack::create(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId);
+ channelMask, frameCount, sharedBuffer, sessionId, uid);
}
if (track == 0 || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
@@ -1497,6 +1525,9 @@
track->mResetDone = false;
track->mPresentationCompleteFrames = 0;
mActiveTracks.add(track);
+ mWakeLockUids.add(track->uid());
+ mActiveTracksGeneration++;
+ mLatestActiveTrack = track;
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
@@ -1970,7 +2001,7 @@
mNumWrites++;
mInWrite = false;
-
+ mStandby = false;
return bytesWritten;
}
@@ -2199,6 +2230,8 @@
// FIXME could this be made local to while loop?
writeFrames = 0;
+ int lastGeneration = 0;
+
cacheParameters_l();
sleepTime = idleSleepTime;
@@ -2255,6 +2288,8 @@
break;
}
releaseWakeLock_l();
+ mWakeLockUids.clear();
+ mActiveTracksGeneration++;
ALOGV("wait async completion");
mWaitWorkCV.wait(mLock);
ALOGV("async completion/wake");
@@ -2285,6 +2320,8 @@
}
releaseWakeLock_l();
+ mWakeLockUids.clear();
+ mActiveTracksGeneration++;
// wait until we have something to do...
ALOGV("%s going to sleep", myName.string());
mWaitWorkCV.wait(mLock);
@@ -2309,11 +2346,18 @@
// mMixerStatusIgnoringFastTracks is also updated internally
mMixerStatus = prepareTracks_l(&tracksToRemove);
+ // compare with previously applied list
+ if (lastGeneration != mActiveTracksGeneration) {
+ // update wakelock
+ updateWakeLockUids_l(mWakeLockUids);
+ lastGeneration = mActiveTracksGeneration;
+ }
+
// prevent any changes in effect chain list and in each effect chain
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
- }
+ } // mLock scope ends
if (mBytesRemaining == 0) {
mCurrentWriteLength = 0;
@@ -2402,7 +2446,6 @@
}
}
- mStandby = false;
} else {
usleep(sleepTime);
}
@@ -2445,6 +2488,8 @@
}
#endif
releaseWakeLock();
+ mWakeLockUids.clear();
+ mActiveTracksGeneration++;
ALOGV("Thread %p type %d exiting", this, mType);
return false;
@@ -2458,6 +2503,8 @@
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove.itemAt(i);
mActiveTracks.remove(track);
+ mWakeLockUids.remove(track->uid());
+ mActiveTracksGeneration++;
ALOGV("removeTracks_l removing track on session %d", track->sessionId());
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
@@ -3666,6 +3713,12 @@
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
+ // Only consider last track started for volume and mixer state control.
+ // In theory an older track could underrun and restart after the new one starts
+ // but as we only care about the transition phase between two tracks on a
+ // direct output, it is not a problem to ignore the underrun case.
+ sp<Track> l = mLatestActiveTrack.promote();
+ bool last = l.get() == track;
// The first time a track is added we wait
// for all its buffers to be filled before processing it
@@ -3675,11 +3728,6 @@
} else {
minFrames = 1;
}
- // Only consider last track started for volume and mixer state control.
- // This is the last entry in mActiveTracks unless a track underruns.
- // As we only care about the transition phase between two tracks on a
- // direct output, it is not a problem to ignore the underrun case.
- bool last = (i == (count - 1));
if ((track->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
@@ -3706,7 +3754,7 @@
} else {
// clear effect chain input buffer if the last active track started underruns
// to avoid sending previous audio buffer again to effects
- if (!mEffectChains.isEmpty() && (i == (count -1))) {
+ if (!mEffectChains.isEmpty() && last) {
mEffectChains[0]->clearInputBuffer();
}
@@ -3718,7 +3766,8 @@
// TODO: implement behavior for compressed audio
size_t audioHALFrames = (latency_l() * mSampleRate) / 1000;
size_t framesWritten = mBytesWritten / mFrameSize;
- if (mStandby || track->presentationComplete(framesWritten, audioHALFrames) ||
+ if (mStandby || !last ||
+ track->presentationComplete(framesWritten, audioHALFrames) ||
track->isTerminated()) {
if (track->isStopped()) {
track->reset();
@@ -3732,6 +3781,8 @@
if (--(track->mRetryCount) <= 0) {
ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
tracksToRemove->add(track);
+ // indicate to client process that the track was disabled because of underrun;
+ // it will then automatically call start() when data is available
android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
} else if (last) {
mixerStatus = MIXER_TRACKS_ENABLED;
@@ -3996,6 +4047,8 @@
mPausedBytesRemaining(0),
mPreviousTrack(NULL)
{
+ //FIXME: mStandby should be set to true by ThreadBase constructor
+ mStandby = true;
}
void AudioFlinger::OffloadThread::threadLoop_exit()
@@ -4032,24 +4085,13 @@
}
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
- if (mPreviousTrack != NULL) {
- if (t.get() != mPreviousTrack) {
- // Flush any data still being written from last track
- mBytesRemaining = 0;
- if (mPausedBytesRemaining) {
- // Last track was paused so we also need to flush saved
- // mixbuffer state and invalidate track so that it will
- // re-submit that unwritten data when it is next resumed
- mPausedBytesRemaining = 0;
- // Invalidate is a bit drastic - would be more efficient
- // to have a flag to tell client that some of the
- // previously written data was lost
- mPreviousTrack->invalidate();
- }
- }
- }
- mPreviousTrack = t.get();
- bool last = (i == (count - 1));
+ // Only consider last track started for volume and mixer state control.
+ // In theory an older track could underrun and restart after the new one starts
+ // but as we only care about the transition phase between two tracks on a
+ // direct output, it is not a problem to ignore the underrun case.
+ sp<Track> l = mLatestActiveTrack.promote();
+ bool last = l.get() == track;
+
if (track->isPausing()) {
track->setPaused();
if (last) {
@@ -4097,6 +4139,31 @@
}
if (last) {
+ sp<Track> previousTrack = mPreviousTrack.promote();
+ if (previousTrack != 0) {
+ if (track != previousTrack.get()) {
+ // Flush any data still being written from last track
+ mBytesRemaining = 0;
+ if (mPausedBytesRemaining) {
+ // Last track was paused so we also need to flush saved
+ // mixbuffer state and invalidate track so that it will
+ // re-submit that unwritten data when it is next resumed
+ mPausedBytesRemaining = 0;
+ // Invalidate is a bit drastic - would be more efficient
+ // to have a flag to tell client that some of the
+ // previously written data was lost
+ previousTrack->invalidate();
+ }
+ // flush data already sent to the DSP if changing audio session as audio
+ // comes from a different source. Also invalidate previous track to force a
+ // seek when resuming.
+ if (previousTrack->sessionId() != track->sessionId()) {
+ previousTrack->invalidate();
+ mFlushPending = true;
+ }
+ }
+ }
+ mPreviousTrack = track;
// reset retry count
track->mRetryCount = kMaxTrackRetriesOffload;
mActiveTrack = t;
@@ -4115,14 +4182,18 @@
track->mState = TrackBase::STOPPING_2; // so presentation completes after drain
// do not drain if no data was ever sent to HAL (mStandby == true)
if (last && !mStandby) {
- sleepTime = 0;
- standbyTime = systemTime() + standbyDelay;
- mixerStatus = MIXER_DRAIN_TRACK;
- mDrainSequence += 2;
+ // do not modify drain sequence if we are already draining. This happens
+ // when resuming from pause after drain.
+ if ((mDrainSequence & 1) == 0) {
+ sleepTime = 0;
+ standbyTime = systemTime() + standbyDelay;
+ mixerStatus = MIXER_DRAIN_TRACK;
+ mDrainSequence += 2;
+ }
if (mHwPaused) {
// It is possible to move from PAUSED to STOPPING_1 without
// a resume so we must ensure hardware is running
- mOutput->stream->resume(mOutput->stream);
+ doHwResume = true;
mHwPaused = false;
}
}
@@ -4146,6 +4217,9 @@
ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list",
track->name());
tracksToRemove->add(track);
+ // indicate to client process that the track was disabled because of underrun;
+ // it will then automatically call start() when data is available
+ android_atomic_or(CBLK_DISABLED, &cblk->mFlags);
} else if (last){
mixerStatus = MIXER_TRACKS_ENABLED;
}
@@ -4159,7 +4233,7 @@
// If a flush is pending and a track is active but the HW is not paused, force a HW pause
// before flush and then resume HW. This can happen in case of pause/flush/resume
// if resume is received before pause is executed.
- if (doHwPause || (mFlushPending && !mHwPaused && (count != 0))) {
+ if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) {
mOutput->stream->pause(mOutput->stream);
if (!doHwPause) {
doHwResume = true;
@@ -4169,7 +4243,7 @@
flushHw_l();
mFlushPending = false;
}
- if (doHwResume) {
+ if (!mStandby && doHwResume) {
mOutput->stream->resume(mOutput->stream);
}
@@ -4291,6 +4365,7 @@
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(mMixBuffer, writeFrames);
}
+ mStandby = false;
return (ssize_t)mixBufferSize;
}
@@ -4325,7 +4400,8 @@
mSampleRate,
mFormat,
mChannelMask,
- frameCount);
+ frameCount,
+ IPCThreadState::self()->getCallingUid());
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f);
mOutputTracks.add(outputTrack);
@@ -4427,7 +4503,6 @@
snprintf(mName, kNameLength, "AudioIn_%X", id);
readInputParameters();
- mClientUid = IPCThreadState::self()->getCallingUid();
}
@@ -4459,7 +4534,11 @@
nsecs_t lastWarning = 0;
inputStandBy();
- acquireWakeLock(mClientUid);
+ {
+ Mutex::Autolock _l(mLock);
+ activeTrack = mActiveTrack;
+ acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1);
+ }
// used to verify we've read at least once before evaluating how many bytes were read
bool readOnce = false;
@@ -4472,6 +4551,12 @@
{ // scope for mLock
Mutex::Autolock _l(mLock);
checkForNewParameters_l();
+ if (mActiveTrack != 0 && activeTrack != mActiveTrack) {
+ SortedVector<int> tmp;
+ tmp.add(mActiveTrack->uid());
+ updateWakeLockUids_l(tmp);
+ }
+ activeTrack = mActiveTrack;
if (mActiveTrack == 0 && mConfigEvents.isEmpty()) {
standby();
@@ -4484,7 +4569,7 @@
// go to sleep
mWaitWorkCV.wait(mLock);
ALOGV("RecordThread: loop starting");
- acquireWakeLock_l(mClientUid);
+ acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1);
continue;
}
if (mActiveTrack != 0) {
@@ -4720,6 +4805,7 @@
audio_channel_mask_t channelMask,
size_t frameCount,
int sessionId,
+ int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status)
@@ -4793,9 +4879,9 @@
track = new RecordTrack(this, client, sampleRate,
#ifdef QCOM_HARDWARE
- format, channelMask, frameCount, *flags, sessionId);
+ format, channelMask, frameCount, *flags, sessionId, uid);
#else
- format, channelMask, frameCount, sessionId);
+ format, channelMask, frameCount, sessionId, uid);
#endif
if (track->getCblk() == 0) {
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 0d01d05..8036a28 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -245,6 +245,9 @@
void acquireWakeLock_l(int uid = -1);
void releaseWakeLock();
void releaseWakeLock_l();
+ void updateWakeLockUids(const SortedVector<int> &uids);
+ void updateWakeLockUids_l(const SortedVector<int> &uids);
+ void getPowerManager_l();
void setEffectSuspended_l(const effect_uuid_t *type,
bool suspend,
int sessionId);
@@ -425,6 +428,7 @@
int sessionId,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
+ int uid,
status_t *status);
AudioStreamOut* getOutput() const;
@@ -499,6 +503,9 @@
void setMasterMute_l(bool muted) { mMasterMute = muted; }
protected:
SortedVector< wp<Track> > mActiveTracks; // FIXME check if this could be sp<>
+ SortedVector<int> mWakeLockUids;
+ int mActiveTracksGeneration;
+ wp<Track> mLatestActiveTrack; // latest track added to mActiveTracks
// Allocate a track name for a given channel mask.
// Returns name >= 0 if successful, -1 on failure.
@@ -759,7 +766,7 @@
bool mFlushPending;
size_t mPausedWriteLength; // length in bytes of write interrupted by pause
size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume
- Track *mPreviousTrack; // used to detect track switch
+ wp<Track> mPreviousTrack; // used to detect track switch
};
class AsyncCallbackThread : public Thread {
@@ -877,6 +884,7 @@
audio_channel_mask_t channelMask,
size_t frameCount,
int sessionId,
+ int uid,
IAudioFlinger::track_flags_t *flags,
pid_t tid,
status_t *status);
@@ -957,5 +965,4 @@
// For dumpsys
const sp<NBAIO_Sink> mTeeSink;
- int mClientUid;
};
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 358d74a..ab83955 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -48,6 +48,7 @@
#endif
const sp<IMemory>& sharedBuffer,
int sessionId,
+ int uid,
bool isOut);
virtual ~TrackBase();
@@ -57,6 +58,7 @@
sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
int sessionId() const { return mSessionId; }
+ int uid() const { return mUid; }
virtual status_t setSyncEvent(const sp<SyncEvent>& event);
protected:
@@ -137,6 +139,7 @@
uint32_t mFlags;
#endif
const int mSessionId;
+ int mUid;
Vector < sp<SyncEvent> >mSyncEvents;
const bool mIsOut;
ServerProxy* mServerProxy;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index dfae5d9..8297b5c 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -71,6 +71,7 @@
#endif
const sp<IMemory>& sharedBuffer,
int sessionId,
+ int clientUid,
bool isOut)
: RefBase(),
mThread(thread),
@@ -99,6 +100,18 @@
mId(android_atomic_inc(&nextTrackId)),
mTerminated(false)
{
+ // if the caller is us, trust the specified uid
+ if (IPCThreadState::self()->getCallingPid() != getpid_cached || clientUid == -1) {
+ int newclientUid = IPCThreadState::self()->getCallingUid();
+ if (clientUid != -1 && clientUid != newclientUid) {
+ ALOGW("uid %d tried to pass itself off as %d", newclientUid, clientUid);
+ }
+ clientUid = newclientUid;
+ }
+ // clientUid contains the uid of the app that is responsible for this track, so we can blame
+ // battery usage on it.
+ mUid = clientUid;
+
// client == 0 implies sharedBuffer == 0
ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -374,14 +387,15 @@
size_t frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ int uid,
IAudioFlinger::track_flags_t flags)
#ifdef QCOM_HARDWARE
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
((audio_stream_type_t)streamType == AUDIO_STREAM_VOICE_CALL)? IAudioFlinger::TRACK_VOICE_COMMUNICATION:0x0,
- sharedBuffer, sessionId, true /*isOut*/),
+ sharedBuffer, sessionId, uid, true /*isOut*/),
#else
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
- sessionId, true /*isOut*/),
+ sessionId, uid, true /*isOut*/),
#endif
mFillingUpStatus(FS_INVALID),
// mRetryCount initialized later when needed
@@ -905,6 +919,7 @@
dstChain->strategy(),
AUDIO_SESSION_OUTPUT_MIX,
effect->id());
+ AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled());
}
status = playbackThread->attachAuxEffect(this, EffectId);
}
@@ -1030,13 +1045,14 @@
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
- int sessionId) {
+ int sessionId,
+ int uid) {
if (!client->reserveTimedTrack())
return 0;
return new TimedTrack(
thread, client, streamType, sampleRate, format, channelMask, frameCount,
- sharedBuffer, sessionId);
+ sharedBuffer, sessionId, uid);
}
AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
@@ -1048,9 +1064,10 @@
audio_channel_mask_t channelMask,
size_t frameCount,
const sp<IMemory>& sharedBuffer,
- int sessionId)
+ int sessionId,
+ int uid)
: Track(thread, client, streamType, sampleRate, format, channelMask,
- frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED),
+ frameCount, sharedBuffer, sessionId, uid, IAudioFlinger::TRACK_TIMED),
mQueueHeadInFlight(false),
mTrimQueueHeadOnRelease(false),
mFramesPendingInQueue(0),
@@ -1543,9 +1560,10 @@
uint32_t sampleRate,
audio_format_t format,
audio_channel_mask_t channelMask,
- size_t frameCount)
+ size_t frameCount,
+ int uid)
: Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount,
- NULL, 0, IAudioFlinger::TRACK_DEFAULT),
+ NULL, 0, uid, IAudioFlinger::TRACK_DEFAULT),
mActive(false), mSourceThread(sourceThread), mClientProxy(NULL)
{
@@ -1808,13 +1826,14 @@
#ifdef QCOM_HARDWARE
uint32_t flags,
#endif
- int sessionId)
+ int sessionId,
+ int uid)
#ifdef QCOM_HARDWARE
: TrackBase(thread, client, sampleRate, format, channelMask, frameCount,
- flags, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
+ flags, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/),
#else
: TrackBase(thread, client, sampleRate, format,
- channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, false /*isOut*/),
+ channelMask, frameCount, 0 /*sharedBuffer*/, sessionId, uid, false /*isOut*/),
#endif
mOverflow(false)
{