Condition must be initialized with SHARED for the old behavior, where
they can be used accross processes.
Updated the two places android that require SHARED conditions.
PRIVATE conditions (and mutexes) use more efficient syscalls.
Change-Id: I9a281a4b88206e92ac559c66554e886b9c62db3a
On binder incalls, the handler thread is given the caller's priority by the
driver, but not the caller's cgroup. We have explicit code that sets the
handler's cgroup to match the caller's, *except* that the system process
explicitly disables this behavior. This led to a siuation in which we were
running binder incalls to the system process at nice=10 but cgroup=fg.
That's fine as far as it goes, except that if a GC happened in the handler
thread, it would be promoted to foreground priority and cgroup both, to avoid
having the GC take forever. Then, when GC finished, the original priority
is reset, and the cgroup set *based on that priority*. This would push the
handler thread into nice=10 cgroup=bg_non_interactive -- which matches the
caller, but is supposed to be impossible in the system process.
The end result of this was that we could be running "lengthy" operations in
the system process in the background. Unfortunately, some of the operations
that wound up like this would hold important global system locks for up to
twenty seconds as a result, making the entire device unresponsive to input
for that period.
This CL fixes the binder incall setup to ensure that within the system process,
a binder incall is always begun from the normal foreground priority as well
as cgroup. In practice now the device still becomes laggy/sluggish when the
offending lock-holding time-consuming incall occurs, but since it still runs
as a foreground task it is able to proceed to completion within a short time
rather than taking 20 seconds.
Fixes bug #2403717
Change-Id: Id046aeabd0e80c48eef94accc37842835eab308d
- AudioPolicyManager: allow platform specific choice for opening a direct output.
Also fixed problems in direct output management.
- AudioFliinger: use shorter standby delay and track inactivity grace period for direct output
thread to free hardware resources as soon as possible.
- AudioSystem: do not use cached output selection in getOutput() when a direct output
can be selected.
Change-Id: If44b50d29237b8402ffd7a5ba1dc43c56f903e9b
if a buffer couldn't be allocated because of an OOM, SF could, in some case dereference
a null pointer.
Change-Id: I5321248c38a21e56d5278b6aada2694e64451378
the framebuffer implementation doesn't do anything special with this
but the surfaceflinger implementation makes sure the surface is not used
by two APIs simultaneously.
Change-Id: Id4ca8ef7093d68846abc2ac814327cc40a64b66b
This loosens our restriction on many manifest attributes requiring
literal string values, to allow various ones to use values from
resources. This is only allowed if the resource value does not change
from configuration changes, and the restriction is still in place
for attributes that are core to security (requesting permissions) or
market operation (used libraries and features etc).
Change-Id: I4da02f6a5196cb6a7dbcff9ac25403904c42c2c8
Part 1 of the fix: when the user doesn't elect to use the car dock
for music and media, the APM was not aware of the device being
docked.
This is fixed by dissociating the notification for the APM of
the docking to the dock from the sink state change of the A2DP
device.
Also missing was forcing the volumes to be reevaluated whenever
the device is docked or undocked, as volumes for docks may
differ, even when the same output device is being used.
Change-Id: If5314e27821a71adbd6df6fdf887c45208241d96
- fix a bug when hacking video buffers into gralloc buffers
where the buffer size was incorrect this was causing the
"direct-form-texture" mode to fail
- also when the above fails, make sure to revert to the
"mdp copy mode" before going to "slow mode"
- finally disable completely the "direct-from-texture" mode
for now. It cannot work because the allocated buffers can't
respect the GPU constraints (alignment and such). We'll
have to find a solution for that.
The problem is that AudioRecord never exits read() when a timeout occurs while trying
to get new PCM data from audio hardware input buffer: it just keeps waiting and retrying until stop() is called.
In the same time, opencore AndroidAudioInput::audin_thread_func() loop cannot be exited when stuck
in AudioRecord::read() because the iExitAudioThread flag can only be sampled when AudioRecord::read()
returns. We remain stuck with the audio input thread running.
The fix consists in modifying AudioRecord behavior in case of timeout when getting new PCM samples.
We now wait only one timeout period and try to restart audio record, in case the problem is due to a media_server
process crash. If this fails, we exit read() with a number of bytes read equals to 0 so that
AndroidAudioInput::audin_thread_func() loop can exit.
Also modified Audioflinger::RecordThread() loop so that we attempt to recover from HAL read errors.
In case of read error, the input stream is forced to standby so that next read attempt does a
reconfiguration and restart of the audio input device.
We've gotten lucky to date: the previous calculation of bitmask array
sizes, (maxval+1)/8 only works properly when 'maxval' is one less than
a multiple of 8. Fortunately, this has either been the case for us,
or there has been sufficient 'unused' space at the end of the defined
max value range that we haven't wound up overreading/overwriting the
allocated buffers.
Change-Id: I563a93a86644ab9f19489565e06c28e06bb53abc
Bug #2376231: Apps lose window focus (and back key causes ANR) if the
lock screen is dismissed while the phone is in landscape mode
This is another case where we weren't recomputing the focused window
after changing the visibility policy.
bug #2479958: Investigate source of "Resources don't contain package
for resource number 0x7f0a0000"
Um, okay, so it turns out there were bugs all over the place where
we would load an XML resource from a another application, but not
use the Resources for that application to retrieve its resources...!
I think the only reason any of this stuff was working at all was
because it typically only cared about retrieving the resource
identifiers of the items (it would look up the values later).
Bug #2401082: Passion ERE26 monkey crash - InputMethodManagerService
Add some null checks.
We now only consider a device to be a default keyboard if its name
has "-keypad". A hack, but whatever.
Also add some debug logging for the input state to help identify such
issues in the future.
And related:
- The aapt tool now sets a resource configurations sdk level to match any configs
that have been set (for example if you specify density your sdk level will be
at least 4).
- New option to modify the targetPackage attribute of instrumentation.
- Clean up of aapt options help.
- Fix of UI type values to leave 0 for "unspecified".
- Make the UI mode config APIs public.
It is spamming the log bigtime and can be promoted back to LOGW
or worse by whoever decides to actually investigate the bug.
Change-Id: I72d950155378f641ebdfbacabae774f5736a52bc
This is not a real fix for the issue but a change to make sure that the behavior is consistent regardless of
external condidions (WIFI ON or OFF, music started before call or not, A2DP device same as SCO device...).
As there is now way to guaranty good quality audio over both SCO and A2DP simultaneously, especially when WIFI is on, We will stick to this behavior:
When music is playing and we are docked to the desk dock and a call is answered with a BT SCO headset, A2DP output will be suspended.
If music is restarted during the call, it will appear muted to the user until the call is terminated.
The noise is the residual ring tone that is still playing while the call is answered and the
audio route changed to headset or earpiece.
The fix consists in muting the ringing tone when changing mode from ringtone to in call
and delaying the route change until the audio buffers are emptied.
StringBlock instances containing UTF-8 strings use a cache to convert
into UTF-16, but using that cache and then using a JNI call to NewString
causes the UTF-8 string as well as two copies of the UTF-16 string to
be held in memory. Getting the UTF-8 string directly from the StringPool
eliminates one copy of the UTF-16 string being held in memory.
This is part 1. Part 2 will include ResXMLParser optimizations.
Change-Id: Ibd4509a485db746d59cd4b9501f544877139276c
Add a Flattenable interface to libutils which can be used to flatten
an object into bytestream + filedescriptor stream.
Parcel is modified to handle Flattenable. And GraphicBuffer implements
Flattenable.
Except for the overlay classes libui is now independent of libbinder.
Unicode.cpp used a packed data table for character data that essentially
duplicated ICU's functionality.
Change-Id: Ia68fe4ac94e89dc68d9a3f45f33f6e648a5500b7
Remove some utility functions for discovering character data
that ICU probably took over a while ago.
Change-Id: I97abe4de2f51eb2bf48679941258bc501184c3dc
This feature is currently controled by a system property.
"ro.sf.hwrotation" can be set to either 90 or 270. It'll cause
SF to rotate the screen by 90 and 270 degres respectively.
That is, if the driver reports 800x480 for instance, and
ro.sf.hwrotation is set to 90, applications will "see" a
480x800 display and will run in portrait.
This is implemented by introducing an extra "display"
transformation in the GraphicPlane.
We now always first try to use the EGLImageKHR directly before
making a copy with copybit. The copy may be needed when
EGLImage doesn't support the requested format, which is
currently the case with YUV.
At some point the implementation became complicated because of
SurfaceFlinger's special needs, since we are now relying on gralloc
we can go back to much simpler MemoryDealer.
Removed HeapInterface and AllocatorInterface, since those don't need
to be paramterized anymore. Merged SimpleMemory and Allocation.
Made SimplisticAllocator non virtual.
Removed MemoryDealer flags (READ_ONLY, PAGE_ALIGNED)
Removed a lot of unneeded code.
Added getRenderPosition() API to IAudioFlinger to retreive number of audio frames
written by AudioFlinger to audio HAL and by DSP to DAC.
Added getRenderPosition() API to AudioHardwareInterface to retreive number of audio frames
written by DSP to DAC.
Exposed AudioTrack::getPosition() to AudioSink() to make it available to media player.
Removed excessive log in AudioHardwareGeneric.
Merge commit '425324e97bba75cd69bb6c81de6248529540e6fe'
* commit '425324e97bba75cd69bb6c81de6248529540e6fe':
Fix failure to open AVRCP input device due to EPERM.
Modified AudioService.getActiveStreamType() so that STREAM_VOICE_CALL is selected when a track using this stream
type is playing.
Chanded isMusicActive() for a more generic isStreamActive(stream) method in AudioSystem, IAudioFlinger and AudioFlinger.
Sleep for 100us and try to open the input device again if it fails, with a
maximum of 10 attempts.
We need the retry logic because setting permissions on a new input device is
racy. The init process watches for new input device (via uevent) and sets the
permission on them in devices.c:make_device(). However at the same time
EventHub.cpp watches for new input devices from the system_server process, and
immediately tries to open them. I can't see a simple way to avoid this race
condition.
As best as I can tell this race condition has always exisited.
There must have been some timing change that happened recently that causes us
to hit this race condition much more often. See repro notes in referenced bug.
Bug: 2375632
make sure to fallback properly to software when copybit operation fails.
with this change, the preview image will at least be displayed in b&w
(since GL doesn't support the yuv format). This would also fix
2363506, but that one is now handled more cleanly.
First implementations of audio policy manager in Eclair branch have shown that most code is common to all platforms.
Creating AudioPolicyManagerBase base class will improve code maintainability and readability.
Audio policy manager code for platforms using generic audio previously in AudioPolicyManagerGeneric is replaced by AudioPolicyManagerBase.
Audio policy manager test code previously in AudioPolicyManagerGeneric is moved to AudioPolicyManagerBase.
Also added a wake lock for delayed commands in AudioPolicyService.
Modified AudioFlinger duplicating output thread so that audio tracks are not mixed until both outputs (A2DP and hardware) have exited standby mode. This avoids to have one output far ahead of the other and audio frames dropped because the compensation mechanism cannot keep up.
Also calculate the maximum wait time in OutputTrack::write() based the on smallest frame count of all output threads instead of the frame count of the thread the OutputTrack is connected to. This avoids starving the thread with the smallest frame count by waiting too long on the other thread.
Since the frame count was reduced on hardware output to reduce latency the difference between A2DP and hardware outputs frame counts had become problematic.
Also increased the number of overflow buffers to cope with bigger timing differences among outputs.
Merge commit 'f9b0e826689cca5ecbd40aa49f3ea7f7c73ad2a2' into eclair-mr2
* commit 'f9b0e826689cca5ecbd40aa49f3ea7f7c73ad2a2':
fix [2269582] [TOP-10][Passion_1506][APT:Camera]Sometimes camera preview screen is truncated after launching and back to home screen by home key repeatedly
When a surface is removed from the screen while it holds a "freeze lock", the
release of that lock happens in the destructor as a "safety net". However, it
doesn't trigger an update at that point.
Make sure that "freeze locks" are released from the transaction at the point
a surface is removed from the screen (if it's not on screen, it shouldn't
prevent the screen to redraw, and therefore cannot hold a freeze lock).
The refresh corresponding to that transaction will pick it up as soon as possible.
Merge commit '083a557c25e0032bc4900f335b6643d0badd09ce' into eclair-mr2
* commit '083a557c25e0032bc4900f335b6643d0badd09ce':
fix [2319255] crash in openGL : from the media recorder stress test.
Merge commit '76169da0e84b0fcf621aeac6141af3ee85bc7c1e' into eclair-mr2
* commit '76169da0e84b0fcf621aeac6141af3ee85bc7c1e':
fix [2315900] Monochrome camera preview screen after launching camera
Merge commit 'd8c752ef74bc6d8b412defe35caf1a19be15eb8b' into eclair-mr2
* commit 'd8c752ef74bc6d8b412defe35caf1a19be15eb8b':
improve video performance to minimize the tearing effect seen in 720p movies
Allows "aapt dump --values resource" to print out whether a string in a
ResStringPool is in UTF-8 or UTF-16 encoding.
Change-Id: I6478884a70a3b46fee862dece6cb33454fc34843
this was introduced by a recent change. when we try to figure out the size of
the yuv->rgb temporary buffer, the output resolution has not been computed yet
and an invalid buffer size is used. most of the time the allocation fails
and the system reverts to "standard" GL will uses onle the Y plane.
the allocation of the temporary buffer is moved to onDraw(), the first
time it is called, by that time, the window is positioned properly.
always rescale videos to their target size using copybit during yuv->rgb
conversion. this improves performance of the GPU pass and doesn't require
linear filtering to be enabled. Also always use 16-bits buffers.
the average processing time for 720p dropped from ~50ms to ~30ms
This is a very simply implementation: upon receiving an IPC, if the handling
thread is at a background priority (the driver will have taken care of
propagating this from the calling thread), then stick it in to the background
scheduling group. Plus an API to turn this off for the process, which is
used by the system process.
This also pulls some of the code for managing scheduling classes out of
the Process JNI wrappers and in to some convenience methods in thread.h.
Allows the use of UTF-8 for packing resources instead of the
default of UTF-16 for Java. When strings are extracted from the
ResStringPool, they are converted to UTF-16 and the result is
cached for subsequent calls.
When using aapt to package, add in the "-8" switch to pack the
resources using UTF-8. This will result in the value, key, and
type strings as well as the compiled XML string values taking
significantly less space in the final application package in
most scenarios.
Change-Id: I129483f8b3d3b1c5869dced05cb525e494a6c83a
The ToneGenerator failed to initialize because no more tracks were available in AudioFlinger mixer.
All tracks were used because the duplicating output was failing to free the tracks on audio hardware output mixer when exiting due to a misplaced test on output activity: output tracks where only freed if the duplicating output was active when exiting.
The fix consists in freeing the output tracks when the duplicating thread is destroyed without condition.
Merge commit '6d42d80653f2c41f3e72a878a1d9a6f9693b89f7' into eclair-mr2
* commit '6d42d80653f2c41f3e72a878a1d9a6f9693b89f7':
Fix issue 2304669: VoiceIME: starting and canceling voice IME yields persistent "error 8" state on future attempts and breaks voice search.
Fixed AudioFlinger::openInput() broken in change ddb78e7753be03937ad57ce7c3c842c52bdad65e
so that an invalid IO handle (0) is returned in case of failure.
Applied the same correction to openOutput().
Modified RecordThread start procedure so that a failure occuring during the first read from audio input stream is detected and causes
the record start to fail.
Modified RecordThread stop procedure to make sure that audio input stream fd is closed before we exit the stop function.
Fixed AudioRecord JAVA and JNI implementation to take status of native AudioRecord::start() into account
and not change mRecordingState to RECORDSTATE_RECORDING if start fails.
Merge commit '0019215fc395ef12c191049b1903eeabf70859cf' into eclair-mr2
* commit '0019215fc395ef12c191049b1903eeabf70859cf':
Revert "When using MDP, we needed to use a texture for diming."
Merge commit '121a31ac3901fcb81c808da2b4a9a7cf66c12b7c' into eclair-mr2
* commit '121a31ac3901fcb81c808da2b4a9a7cf66c12b7c':
fix [2291418] Camera preview cannot work in Emulator
The image buffer used by glTexImage2d() would be uninitialized when no copybit engine
can be found.
We now always initialize images, since the abscence of copybit is not necessarily fatal.
Merge commit '1ac56b602aa6a1ac54c608e5a8b76f44638db23b' into eclair-mr2
* commit '1ac56b602aa6a1ac54c608e5a8b76f44638db23b':
Fix issue 2292062: Audio freezes for three seconds when choosing ringtones with a headset connected and music playing.
There was bug in the logic that calculated the relative timeout, the start time was
reset each time an event was received, which caused the timeout to never occur if
an application was constantly redrawing.
Now we always check for a timeout when we come back from the waitEvent() and
process the "anti-freeze" if needed, regardless of whether an event was received.
The problem comes from a deadlock with AudioPolicyService mutex: When the second ringtone starts,
this mutex is locked by AudioPolicyService::startOutput() which in turn calls setParameters() to change the output device.
Audioflinger::ThreadBase::setParameters() signals the parameter change to the AudioFlinger mixer thread and waits for a condition
indicating that the parameter change has been processed.
At the same time, the mixer thread detects that the audio track corresponding to the first ring tone has been killed and calls its destructor.
This calls AudioPolicyService::releaseOutput() which tries to lock the AudioPolicyService mutex.
If this happens before the mixer thread can process the setParameters() command we are deadlocked.
The deadlock ends because setParameters() uses a timeout when waiting for the condition.
This regression was introduced by change 33736 fixing issue 2265163.
The fix consists in calling AudioPolicyService::releaseOutput() from Track::destroy() instead of from Track destructor: as detroy() is never called from the mixer thread loop (as opposed to the destructor) the deadlock described above cannot occur.
Binary XML file line #37: Error inflating class <unknown> after adding a secondary account
Now that I have these debug logs, I want to keep them since they will make
debugging these kinds of issues a lot easier in the future. (Note in this
case there was no problem in the framework.)
Change-Id: If2b0bbeda4706b7c5dc1ba4a5db04b74f40e1543
This is a second attempt to fix the audio routed to earpiece syndrom.
The root cause identified this time is the crash of an application having an active AudioTrack playing on the VOICE_CALL stream type.
When this happens, the AudioTrack destructor is not called and the audio policy manager is not notified of the track stop.
Results a situation where the VOICE_CALL stream is considered as always in use by audio policy manager which makes that audio is routed to earpiece.
The fix consists in moving the track start/stop/close notification to audio policiy manager from AudioTrack to AudioFlinger Track objet.
The net result is that in the case of a client application crash, the AudioFlinger TrackHandle object (which implements the remote side of the IAudioTrack binder interface) destructor is called which in turn destroys the Track object and we can notify the audio policy manager of the track stop and removal.
The same modification is made for AudioRecord although no bug related to record has been reported yet.
Also fixed a potential problem if record stop is called while the record thread is exiting.
since we're using the GPU for composition, don't use a texture for dimming,
instead simply use an alpha-blended quad.
also workaround what looks like a GL driver bug by calling glFinish() before
glReadPixels().
2206097: Broken suggestions while composing message
2166583: Color artifacts with MDP dithering
2261119: Passion transition animations are rough
2216759: Screen flicker when dropdown list in background window shows or hides
This is part of enabling GPU composition instead of using the MDP. This change
is dependent on another change in the vendor project.
Specifically this change disables the use of EGLImageKHR for s/w buffers
for cache coherency reasons. memcpy is used instead.
Surface::validate() could sometimes dereference a null pointer before checking it wasn't null.
This will prevent the application to crash when given bad parameters or used incorrectly.
However, the bug above probably has another cause.
in the kernel requires a guard page, so 1M allocations fragment memory very
badly. Subtracting a couple of pages so that they fit in a power of
two allows the kernel to make more efficient use of its virtual address space.
Signed-off-by: Rebecca Schultz Zavin <rebecca@android.com>
This builds on the EGLImage solution. We simply use copybit to convert from the
YUV frame into an EGLImage created for that purpose and proceed with the
regular EGLImage code.
We need to do this because "regular" GL doesn't support YUV textures.
We could improve upon this by detecting exacly what the GL supports and bypass
this extra step if not required, but we'll do this later if needed.
This change goes with a kernel driver change that reduces the audio buffer size from 4800 bytes (~27ms) to 3072 bytes (~17ms).
- The AudioFlinger modifcations in change 0bca68cfff161abbc992fec82dc7c88079dd1a36 have been removed: the short sleep period was counter productive when the AudioTrack is using the call back thread as it causes to many preemptions.
- AudioFlinger mixer thread now detects long standby exit time and in this case anticipates start by writing 0s as soon as a track is enabled even if not ready for mixing.
- AudioTrack::start() is modified to start call back thread before starting the IAudioTrack so that thread startup time is masked by IAudioTrack start and mixer thread wakeup time.
To prevent buggy command implementations from poisoning binder threads'
scheduling class & priority for future command execution, we now reset the
cgroup and thread priority to foreground/normal when a binder service thread
finishes executing the designated command.
Change-Id: Ibc0ab2485751453f6dc96fdb4eb877fd02796e3f
we lost the concept of vertical stride when moving video playback to EGLImage.
Here we bring it back in a somewhat hacky-way that will work only for the
softgl/mdp backend.
Reduce sleep time in AudioFlinger mixer thread when no data has been written to output to speed up startup time when exiting standby.
The rest of the modifications for this issues is in kernel driver:
commit 0dbb0ee136ed8de757df1ae26d84556c1751deae for buffer size modification from 8192 to 4800 bytes.
Another kernel improvement that is not submitted yes will reduce delay when audio output is exiting standby.
add a way to convert a mapped "pushbuffer" buffer to a gralloc handle
which then can be safely used by surfaceflinger, without including
gralloc_priv.h
Temporarily make a function public that doesn't need to be. When
host gcc-4.0.3 is gone from the build servers we can undo this.
(Cherry-picked from eclair-mr2.)
Use EGLImageKHR instead of copybit directly.
We now have the basis to use streaming YUV textures (well, in fact
we already are). When/if we use the GPU instead of the MDP we'll
need to make sure it supports the appropriate YUV format.
Also make sure we compile if EGL_ANDROID_image_native_buffer is not supported
Instead of using glTex{Sub}Image2D() to refresh the textures, we're using an EGLImageKHR object
backed up by a gralloc buffer. The data is updated using memcpy(). This is faster than
glTex{Sub}Image2D() because the texture is not swizzled. It also uses less memory because
EGLImageKHW is not limited to power-of-two dimensions.
Added setVoiceVolume() method to AudioSystem, AudioFlinger, IAudioFlinger, AudioPolicyService.
Removed call to AudioHardwareInterface::setVoiceVolume() from AudioFlinger::setStreamVolume().
Added a workarouond to request the A2DP output standby directly to audio hardware when the sink is suspended as it seems that the suspend request often fails.
Also take into account resume requests received while a suspend request is pending.
when running out of memory, a null handle is returned but the error code may not be set.
In that case we need to return NO_MEMORY instead of NO_ERROR, so that the calling code
won't try to dereference the null pointer.
When switching rapidily orientation back and forth, surfaces end-up
acquiring the freeze-lock when the first orientation change happens,
but never release it because by the time the 2nd orientation change
comes in, the surface size is back to its original size and
doesn't appear to have resized.
we now always release the freeze-lock when we receive a buffer of the
expected size.
This also fixes [2152536] ANR in browser
When SF is enqueuing buffers faster than SF dequeues them.
The update flag in SF is not counted and under some situations SF will only
dequeue the first buffer. The state at this point is not technically
corrupted, it's valid, but just delayed by one buffer.
In the case of the Browser ANR, because the last enqueued buffer was delayed
the resizing of the current buffer couldn't happen.
The system would always fall back onto its feet if anything -else- in
tried to draw, because the "late" buffer would be picked up then.
A window is created and the browser is about to render into it the
very first time, at that point it does an IPC to SF to request a new
buffer. Meanwhile, the window manager removes that window from the
list and the shared memory block it uses is marked as invalid.
However, at that point, another window is created and is given the
same index (that just go freed), but a different identity and resets
the "invalid" bit in the shared block. When we go back to the buffer
allocation code, we're stuck because the surface we're allocating for
is gone and we don't detect it's invalid because the invalid bit has
been reset.
It is not sufficient to check for the invalid bit, I should
also check that identities match.
This change is a complement to the main fix in kernel driver for the same issue (partner change #1250).
It removes clicks sometimes heard after the end of the tones while audio flinger is sending 0s to the audio output stream.
The problem was that the sleep time between two writes was more than the duration of one audio output stream buffer which could cause some underrun.
Also fixed a recent regression in ToneGenerator that made that the end of previous tone was repeated at the beginning of current one under certain timing circumstances when the maximum tone duration was specified.
When EGLImage extension is not available, SurfaceFlinger will fallback to using
glTexImage2D and glTexSubImage2D instead, which requires 50% more memory and an
extra copy. However this code path has never been exercised and had some bugs
which this patch fix.
Mainly the scale factor wasn't computed right when falling back on glDrawElements.
We also fallback to this mode of operation if a buffer doesn't have the adequate
usage bits for EGLImage usage.
This changes only code that is currently not executed. Some refactoring was needed to
keep the change clean. This doesn't change anything functionaly.
The ANR is caused by SurfaceFlinger waiting for buffers of a removed surface to become availlable.
When it is removed from the current list, a Surface is marked as NO_INIT, which causes SF to return
immediately in the above case. For some reason, the surface here wasn't marked as NO_INIT.
This change makes the code more robust by always (irregadless or errors) setting the NO_INIT status
in all code paths where a surface is removed from the list.
Additionaly added more information in the logs, should this happen again.
The core logging in BackupManagerService and in the Google backup transport are
still enabled at this point.
Change-Id: I10abfa565bbd1097dd3631051b6aca163e4af33a
Wait for the parameter set completed condition with a time out in ThreadBase::setParameters().
Also lock AudioFlinger mutex before accessing thread list in AudioFlinger::setParameters() and keep a strong reference
on the thread being used in case it is exited while processing the request.
* changes:
fix [2152247] Windows sometimes drawn scaled up.
invalidate the surface when the physical changes
introduce the notion of the requested size in the Layer state
remove unused code
We were emitting GL commands, calling composition complete and releasing clients
without ever calling eglSwapBuffers(), which is completely wrong on non-direct
renders. This could cause transient drawing artifacts when unfreezing the
screen (upon orientaion change for instance) and could also block the clients
for ever as they are waiting for their previous buffer to be rendered.
Turning off backup in the Settings UI constitutes an opt-out of the whole
mechanism. For privacy reasons we instruct the backend to wipe all of the data
belonging to this device when the user does this. If the attempt fails it is
rescheduled in the future based on the transport's requestBackupTime()
suggestion. If network connectivity changes prompt the transport to indicate a
backup pass is appropriate "now," any pending init operation is processed before
the backup schedule is resumed.
The broadcasts used internally to the backup manager are now fully protected;
third party apps can neither send nor receive them.
(Also a minor logging change; don't log 'appropriate' EOF encountered during
parsing of a backup data stream.)
There was a regression introduced in AudioFlinger by change 24114 for suspended output:
The suspended output was not reading and mixing audio tracks.
When the phone is ringing, the A2DP output is suspended if the SCO headset and A2DP headset are the same. As the ringtone is played over the duplicated output, the fact that the A2DP output was not reading data was causing the hardware output to be stalled from time to time.
This appears to fix the sim-eng build on the gDapper build machines.
Basic problem is that LayerBuffer::OverlaySource has a constructor that
calls SurfaceFlinger.signalEvent(). SurfaceFlinger lists LayerBuffer
as a friend, but that's not enough to convince gcc that the embedded
OverlaySource class is also a friend. I don't see a way to make them
friendly, so I marked signalEvent() as public.
a new method, compostionComplete() is added to the framebuffer hal, it is used by surfaceflinger to signal the driver that the composition is complete, BEFORE it releases its client. This gives a chance to the driver to
The fix consists in locking AudioFlinger::mLock mutex in the TrackBase destructor before clearing the strong pointer to the shared memory client. The mutex is not locked in removeclient() any more which implies that we must make sure that the Client destructor is always called from the TrackBase destructor or that we hold the mLock mutex before calling deleting the Client.
Take 2. We needed to check that the usage flags are "good enough" as opposed to "the same".
This reverts commit 8f17a762fe9e9f31e4e86cb60ff2bfb6b10fdee6.
The problem comes from the fact that when the duplicated output is closed after BT headset disconnection, the OUTPUT_CLOSED notification is not sent to AudioSystem. Then the mapping between notification stream and duplicated output cached in AudioSystem is not cleared and next time a notification is played, the duplicated output is selected and the createTrack() request is refused by AudioFlinger as the selected output doesn't exist.
The notification is ignored by AudioFlinger because when it is sent by the terminating playback thread, the thread has already been removed from the playback thread list.
The fix consists in sending the notification in closeOutput() and not when exiting the playback thread.
The same fix is applied to record threads.
This is due to a regression introduced by change 24114: when no audio tracks are ready for mixing, 0s are written to audio hardware. However this should only happen if tracks have already been mixed since the audio flinger thread woke up.
Also do not write 0s to audio hardware in direct output threads when audio format is not linear PCM.
Appears to have been broken by:
commit 9779b221e999583ff89e0dfc40e56398737adbb3
Author: Mathias Agopian <mathias@google.com>
Date: Mon Sep 7 16:32:45 2009 -0700
fix [2068105] implement queueBuffer/lockBuffer/dequeueBuffer properly
For some reason we don't like to have "-lpthread" globally -- it's a no-op
on device builds, but required for many host tools and all sim binaries --
so adding the use of pthread calls requires adding the library explicitly.
AudioFlinger: verify that mCblk is not null before using it in Track and RecordTrack contructors.
IAudioFlinger: check result of remote transaction before reading IAudioTrack and IAudioRecord.
IAudioTrack and IAudioRecord: check result of remote transaction before reading IMemory.
we could have several thread waiting on the condition and they all need to wake-up.
also added a debug "mTid" field in the class, which contains the tid of the thread (as opposed to pthread_t), this
is useful when debugging under gdb for instance.
we ended-up locking a Mutex that had been destroyed.
This happened because we gave an sp<Source> to the outside world,
and were called after LayerBuffer had been destroyed.
Instead we now give a wp<LayerBuffer> to the outside and have it
do the destruction.
Add a parameter to ToneGenerator.startTone() allowing the caller to specify the tone duration. This is used by the phone application to have a precise control on the DTMF tone duration which was not possible with the use of delayed messaged.
Also modified AudioFlinger output threads so that 0s are written to the audio output stream when no more tracks are ready to mix instead of just sleeping. This avoids an issue where the end of a previous DTMF tone could stay in audio hardware buffers and be played just before the beginning of the next DTMF tone.
Rewrote SurfaceFlinger's buffer management from the ground-up.
The design now support an arbitrary number of buffers per surface, however the current implementation is limited to four. Currently only 2 buffers are used in practice.
The main new feature is to be able to dequeue all buffers at once (very important when there are only two).
A client can dequeue all buffers until there are none available, it can lock all buffers except the last one that is used for composition. The client will block then, until a new buffer is enqueued.
The current implementation requires that buffers are locked in the same order they are dequeued and enqueued in the same order they are locked. Only one buffer can be locked at a time.
eg. Allowed sequence: DQ, DQ, LOCK, Q, LOCK, Q
eg. Forbidden sequence: DQ, DQ, LOCK, LOCK, Q, Q
Do not ramp volume if the first frame of a track is processed after the track was stopped.
In the case of very short sounds, the track stop request can be received by AudioFlinger just after the start request before the first frame is mixed by AudioMixer. In this case, the track is already in stopped state and initial volume is applied with a ramp for the first frame processed which should not be the case: initial volume change is always applied immediatelly.
