Merge commit 'cdf1357b6e0152280dcd611b5f096db4887d8126'
* commit 'cdf1357b6e0152280dcd611b5f096db4887d8126':
Various fixes and improvements in audio effects implementation
Effect API:
- Use different definitions for audio device, channels, formats... in AudioSystem and EffectApi:
Removed media/AudioCommon.h file created for initial version of EffectApi
- Indicate audio session and output ID to effect library when calling EffectCreate(). Session ID can be useful to optimize
the implementation of effect chains in the same audio session. Output ID can be used for effects implemented in audio hardware.
- Renamed EffectQueryNext() function to EffectQueryEffect() and changed operating mode:
now an index is passed for the queried effect instead of implicitly querying the next one.
- Added CPU load and memory usage indication in effects descriptor
- Added flags and commands to indicate changes in audio mode (ring tone, in call...) to effect engine
- Added flag to indicate hardware accelerated effect implementation.
- Renamed EffectFactoryApi.h to EffectsFactoryApi.h for consistency with EffectsFactory.c/h
Effect libraries:
- Reflected changes in Effect API
- Several fixes in reverb implementation
- Added build option TEST_EFFECT_LIBRARIES in makefile to prepare integration of actual effect library.
- Replaced pointer by integer identifier for library handle returned by effects factory
Audio effect framework:
- Added support for audio session -1 in preparation of output stage effects configuration.
- Reflected changes in Effect API
- Removed volume ramp up/down when effect is inserted/removed: this has to be taken care of by effect engines.
- Added some overflow verification on indexes used for deferred parameter updates via shared memory
- Added hardcoded CPU and memory limit check when creating a new effect instance
Change-Id: I43fee5182ee201384ea3479af6d0acb95092901d
Merge commit '256fc04394431cbd332e56747fdbfda4cb4c2e25'
* commit '256fc04394431cbd332e56747fdbfda4cb4c2e25':
fix [2793164] Spam 2x/second with TOT master in SurfaceFlinger
Make sure to not use GL_TEXTURE_EXTERNAL when it's not supported
by the GL. The error was harmless, but annoying.
Change-Id: I571a9a9b05d35da51420950a6a6e95629067efd0
Merge commit 'bc0793e4b77f1f8ec30294a4edac67dfca81f31d'
* commit 'bc0793e4b77f1f8ec30294a4edac67dfca81f31d':
Remember to initialize timestamps in the dispatch allocator
Merge commit 'efcf68aa1fd7fcfd52cf3d2837ed8db8e797194b'
* commit 'efcf68aa1fd7fcfd52cf3d2837ed8db8e797194b':
Start of work on passing around StrictMode policy over Binder calls.
Merge commit 'cefb88587443323d147e687ff78eae9195eb584c'
* commit 'cefb88587443323d147e687ff78eae9195eb584c':
Added support for the GL_TEXTURE_EXTERNAL target
Provides the basic infrastructure for a
NativeActivity's native code to get an object representing
its event stream that can be used to read input events.
Still work to do, probably some API changes, and reasonable
default key handling (so that for example back will still
work).
Change-Id: I6db891bc35dc9683181d7708eaed552b955a077e
Added ANRs handling.
Added event injection.
Fixed a NPE ActivityManagerServer writing ANRs to the drop box.
Fixed HOME key interception.
Fixed trackball reporting.
Fixed pointer rotation in landscape mode.
Change-Id: I50340f559f22899ab924e220a78119ffc79469b7
This is (intendend to be) a no-op change.
At this stage, Binder RPCs just have an additional uint32 passed around
in the header, right before the interface name. But nothing is actually
done with them yet. That value should right now always be 0.
This now boots and seems to work.
Change-Id: I135b7c84f07575e6b9717fef2424d301a450df7b
Merge commit '42bb545a54d89f0ddbb230d7a01ea4210c0f6c00'
* commit '42bb545a54d89f0ddbb230d7a01ea4210c0f6c00':
Even more native input dispatch work in progress.
Added more tests.
Fixed a regression in Vector.
Fixed bugs in pointer tracking.
Fixed a starvation issue in PollLoop when setting or removing callbacks.
Fixed a couple of policy nits.
Modified the internal representation of MotionEvent to be more
efficient and more consistent.
Added code to skip/cancel virtual key processing when there are multiple
pointers down. This helps to better disambiguate virtual key presses
from stray touches (such as cheek presses).
Change-Id: I2a7d2cce0195afb9125b23378baa94fd2fc6671c
Refactored the code to eliminate potential deadlocks due to re-entrant
calls from the policy into the dispatcher. Also added some plumbing
that will be used to notify the framework about ANRs.
Change-Id: Iba7a10de0cb3c56cd7520d6ce716db52fdcc94ff
The old dispatch mechanism has been left in place and continues to
be used by default for now. To enable native input dispatch,
edit the ENABLE_NATIVE_DISPATCH constant in WindowManagerPolicy.
Includes part of the new input event NDK API. Some details TBD.
To wire up input dispatch, as the ViewRoot adds a window to the
window session it receives an InputChannel object as an output
argument. The InputChannel encapsulates the file descriptors for a
shared memory region and two pipe end-points. The ViewRoot then
provides the InputChannel to the InputQueue. Behind the
scenes, InputQueue simply attaches handlers to the native PollLoop object
that underlies the MessageQueue. This way MessageQueue doesn't need
to know anything about input dispatch per-se, it just exposes (in native
code) a PollLoop that other components can use to monitor file descriptor
state changes.
There can be zero or more targets for any given input event. Each
input target is specified by its input channel and some parameters
including flags, an X/Y coordinate offset, and the dispatch timeout.
An input target can request either synchronous dispatch (for foreground apps)
or asynchronous dispatch (fire-and-forget for wallpapers and "outside"
targets). Currently, finding the appropriate input targets for an event
requires a call back into the WindowManagerServer from native code.
In the future this will be refactored to avoid most of these callbacks
except as required to handle pending focus transitions.
End-to-end event dispatch mostly works!
To do: event injection, rate limiting, ANRs, testing, optimization, etc.
Change-Id: I8c36b2b9e0a2d27392040ecda0f51b636456de25
It was possible for stylesStrings to claim to start past the end of the
data area thereby making mStringPoolSize larger than the data area.
Change-Id: Ibc4d5b429e3a388516135801c8abc3681daae291
Surfaces can now be parcelized and sent to remote
processes. When a surface crosses a process
boundary, it looses its connection with the
current process and gets attached to the new one.
Change-Id: I39c7b055bcd3ea1162ef2718d3d4b866bf7c81c0
The aapt dump reading had less error checking than the actual parsing,
so this change brings it more into parity so that bad APKs don't crash
"aapt dump"
Change-Id: Ib30e63e41be5c652645c4aa0de580a87b184529d
this is called for each relayout() and used to create a full Surface (cpp)
which in turn did some heavy work (including an IPC with surfaceflinger),
most of the time to destroy it immediatelly when the returned surface
(the one in the parcel) was the same.
we now more intelligentely read from the parcel and construct the new
object only if needed.
Change-Id: Idfd40d9ac96ffc6d4ae5fd99bcc0773e131e2267
simplified things a lot, the biggest change is that the concept
of "ClientID" is now gone, instead we simply use references.
Change-Id: Icbc57f80865884aa5f35ad0d0a0db26f19f9f7ce
First drop of audio framework modifications for audio effects support.
- AudioTrack/AudioRecord:
Added support for auxiliary effects in AudioTrack
Added support for audio sessions
Fixed left right channel inversion in setVolume()
- IAudioFlinger:
Added interface methods for effect enumeraiton and instantiation
Added support for audio sessions.
- IAudioTrack:
Added method to attach auxiliary effect.
- AudioFlinger
Created new classes to control effect engines in effect library and manage effect connections to tracks or
output mix:
EffectModule: wrapper object controlling the effect engine implementation in the effect library. There
is one EffectModule per instance of an effect in a given audio session
EffectChain: group of effects associated to one audio session. There is one EffectChain per audio session.
EffectChain for session 0 is for output mix effects, other chains are attached to audio tracks
with same session ID. Each chain contains a variable number of EffectModules
EffectHandle: implements the IEffect interface. There is one EffectHandle object for each application
controlling (or using) an effect module. THe EffectModule maintians a list of EffectHandles.
Added support for effect modules and effect chains creation in PlaybackThread.
modified mixer thread loop to allow track volume control by effect modules and call effect processing.
-AudioMixer
Each track now specifies its output buffer used by mixer for accumulation
Modified mixer process functions to process tracks by groups of tracks with same buffer
Modified track process functions to support accumulation to auxiliary channel
Change-Id: I26d5f7c9e070a89bdd383e1a659f8b7ca150379c
opaque 32-bits windows are now allocated as RGBX_8888 buffers and
SurfaceFlinger always uses GL_MODULATE instead of trying to
optimize to GL_REPLACE when possible (makes no sense on
h/w accelerated GL).
we still have a small hack for devices that don't support
RGBX_8888 in their gralloc implementation where we revert to
RGBA_8888.
To allow use of the native CursorWindow class outside of the core framework jni
Change-Id: I72e8dcb91a2c691130c33cdfd9a25d343da1c592
Signed-off-by: Mike Lockwood <lockwood@android.com>
SurfaceComposerClient now only exist on the WindowManager side,
the client side uses the new SurfaceClient class, which only
exposes what a client needs.
also instead of keeping mappings from IBinder to SurfaceComposerClients
we have a SurfaceClient per Surface (referring to the same IBinder), this
is made possible by the fact that SurfaceClient is very light.
Change-Id: I6a1f7015424f07871632a25ed6a502c55abfcfa6
The problem is that the code in AudioPolicyManagerBase::checkAndSetVolume() that forces
voice volume to max when setting bluetooth SCO volume is not called if the bluetooth stream
volume did not actually change. So even if we re apply volumes when switching to bluetooth
device, the volume voice volume is not changed and remains what it was when routed to earpiece
What makes things worse on Passion is that stream volumes are limited when connected to bluetooth
and their actual value does not change as soon as they exceed the limit threshold.
Change-Id: I18265e5e6686db0a1f30fc37a31e2ecde4f3fbc6
the new native_window_set_buffers_geometry allows
to specify a size and format for all buffers to be
dequeued. the buffer will be scalled to the window's
size.
Change-Id: I2c378b85c88d29cdd827a5f319d5c704d79ba381
this method can be used to change the number of buffers
associated to a native window. the default is two.
Change-Id: I608b959e6b29d77f95edb23c31dc9b099a758f2f
this change introduces R/W locks in the right places.
on the server-side, it guarantees that setBufferCount()
is synchronized with "retire" and "resize".
on the client-side, it guarantees that setBufferCount()
is synchronized with "dequeue", "lockbuffer" and "queue"
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).
The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.
AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.
AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.
AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.
Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
Merge commit '900b6157f5dee2ed7b2c73cf320b2baf293230ff' into kraken
* commit '900b6157f5dee2ed7b2c73cf320b2baf293230ff':
Only hold a weak pointer on SurfaceComposerClients
Some variables and structure members should be renamed to reflect the fact that they contain the
number of channels in a track (channel count) or the actual channels used by a track (channel mask).
Especially member "channels" of track control block (struct audio_track_cblk_t) is actually the
number of channels (channels count).
Change-Id: I220c8dede9fc00c8a5693389e790073b6ed307b8
the new TextureMagager class now handle texture creation and upload
as well as EGL image creation and binding to GraphicBuffers. This is
used indirectly by Layer and directly by LayerBuffer
the new BufferManager class handles the set of buffers used for a
Layer (Surface), it abstracts how many buffer there is as well as
the use of EGLimage vs. regular texture ops (glTexImage2D).
Change-Id: I2da1ddcf27758e6731400f6cc4e20bef35c0a39a
this hack was used for gpus that don't support cached buffers
for s/w clients. currently we have no gpu with this issue.
this removes quite a bit of complexity.
Change-Id: I72564669f124f92805030e61983711f61c76b6d9
- forward setMode() and getInputBufferSize() calls to underlying audio hardware interface.
- Allow capture of more than one output stream (previous implementation was only capturing
the first output opened, namely the hardware output).
- Allow capture of input streams: previous implementation was only simulating input streams
when more than one was open at a time by reading from a file on SD card). Now the default
behavior is to capture PCM data read from input stream if it was successfully opened or
simulate capture otherwise.
Change-Id: I7e2892b25e295fc3c19c7eb0f71bfaea5816b73a
There is a bug in the way notification client list is managed when the client binder
interface dies that makes that the dead client is not removed from the list: the week
reference passed by binderDied() cannot be promoted and compared to the strong
references in the list.
The fix consists in creating a new NotificationClient class that implements the
binder DeathRecipient and holds a strong reference to the IAudioFlingerClient interface.
A new instance of this class is created for each cient and a strong reference to this
object is added to the notification client list maintained by AudioFlinger.
When binderDied() is called on this object, it is removed from the list preventing
AudioFlinger to notify this client for further io changes.
Also added code to disable LifeVibes effects when the client that has enabled the
enhancements dies.
Change-Id: Icedc4af171759e9ae9a575d82d44784b4e8267e8
Change the way zip archives are handled. This is necessary to deal with
very large (~1GB) APK files, for which our current approach of mapping
the entire file falls over.
We now do the classic scavenger hunt for the End Of Central Directory
magic on a buffer of data read from the file, instead of a memory-mapped
section. We use what we find to create a map that covers the Central
Directory only.
If the caller is interested in unpacking the file contents, we have to
do an additional file read to discover the size of the Local File Header
section so we can skip past it.
This is based on Change I745fb15abb in the dalvik tree. Both
implementations share a common ancestry, but the cost of unifying them
outweighs the benefits of wrapping C calls.
Change-Id: Iddacb50fe913917c2845708a530872d65fdbe620
Merge commit '56aed6bde0c52658d2cb1207c0cfe8ba0a764c59' into kraken
* commit '56aed6bde0c52658d2cb1207c0cfe8ba0a764c59':
fix [2664345] Flash: Bad flicker at the end of a pinch zoom.
the window manger puts SurfaceViews up before they have been
rendered into, because of that surfaceflinger doesn't have
anything ready to draw for that surface when an udpate occurs
and responds by filling the surface with black.
With this fix, we only fill those areas of the framebuffer
that would otherwise be undefined (no content at all).
in the Flash case, the "flash" window is not drawn at all
until it has some content, instead the underlaying browser
window is shown.
Change-Id: Ifb610f7f8c27b88edf83e09adc4803fc295c15a1
Merge commit 'ca48c88c3d5733c4405a2fc4f7d9bb7fbba3d43f' into kraken
* commit 'ca48c88c3d5733c4405a2fc4f7d9bb7fbba3d43f':
Make static versions of libutils and libbinder.
Fix some small static-initialization-order issues (and a static-
initializers-missing issue) that result from doing so. The static
libraries don't actually get used for anything real at the moment --
they're used for perf tests of bug 2660235.
Bug: 2660235
Change-Id: Iee2f38f79cc93b395e8d0a5a144ed92461f5ada0
Not complete, only for experimentation at this point.
This includes a reworking of how screen size configurations are matched,
so that if you are on a larger screen we can select configurations for
smaller screens if there aren't any exactly matching the current screen.
The screen size at which we switch to xlarge has been arbitrarily
chosen; the compatibility behavior has not yet been defined.
Change-Id: I1a33b3818eeb51a68fb72397568c39ab040a07f5
the reason for the above change is that waitForCondition() had become
large over time, mainly to handle error cases, using inlines to
evaluate the condition doesn't buys us much anymore while it increases
code size.
Change-Id: I2595d850832628954b900ab8bb1796c863447bc7
in the undoDequeue() case, 'tail' was recalculated from 'available' and 'head'
however there was a race between this and retireAndLock(), which could cause
'tail' to be recalculated wrongly.
the interesting thing though is that retireAndLock() shouldn't have any impact
on the value of 'tail', which is client-side only attribute.
we fix the race by saving the value of 'tail' before dequeue() and restore it
in the case of undoDequeue(), since we know it doesn't depend on retireAndLock().
Change-Id: I4bcc4d16b6bc4dd93717ee739c603040b18295a0
get rid of the "fake rtti" code, and use polymorphism instead.
also simplify how we log SF's state (using polymorphism)
Change-Id: I2bae7c98de4dd207a3e2b00083fa3fde7c467922
also increase the dirtyregion size from 1 to 6 rectangles.
Overall we now need 27KiB process instead of 4KiB
Change-Id: Iebda5565015158f49d9ca8dbcf55e6ad04855be3
In case of A2DP write errors, there is an overflow in the calculation
of the sleep duration to simulate the timing of a successful write.
Change-Id: Ic4e570aebf07fac69735aab1bbc2fc73512ee795
Merge commit '26f6163557980062dbb203388b3d0794ee0d06f7' into kraken
* commit '26f6163557980062dbb203388b3d0794ee0d06f7':
fix [2599939] "cannot play video" after open/close a video player a dozen of times
get rid off the MAP_ONCE flag is MemoryHeapBase (as well as it's functionality),
this feature should not be used anymore.
the software renderer was incorrectly using the default ctor which set MAP_ONCE,
causing the leak. the software renderer itself is incorrectly used while coming
back from sleep.
Change-Id: I123621f8d140550b864f352bbcd8a5729db12b57
Merge commit 'e7d5a2f9ae47d8ea8face3f1e451314ed36f4026' into kraken
* commit 'e7d5a2f9ae47d8ea8face3f1e451314ed36f4026':
fix [2594950] Flash: Zooming in on some content crashes the Nexus One and causes it to reboot (runtime restart)
We now limit the size of the surface to the maximum size supported by the GPU.
On Nexus One this will 2048 -- it could be different on other devices.
Surface creation fails if the limit is exceeded.
Change-Id: I9ecfc2e9c58c9e283782b61ebfc6b590f71df785
This changes fixes the issue for the direct output thread that was not
addressed by commit 71f37cd8a175ee00635cb91506d6810fd02b5b51.
Change-Id: I1bbe26be5f444415dd97270e49257650f5d2858f
Each process maintains an array of active SurfaceComposerClient
objects, so that they can be reused as new surfaces are parceled
across. When a SurfaceComposerClient is disposed, it will remove
itself from this list. However, because the list maintains a strong
reference on the object, a reference cycle is created, and the
client is never deleted.
This patch changes the list to maintain weak pointers on the clients
instead.
Change-Id: I93dc8155fe28b4e350366a3400cdf22a8c77cdd3
The problem is a bug in AudioFlinger::MixerThread::prepareTracks_l() that makes that even if the TrackHandle
is destroyed, the corresponding Track will remain active as long as frames are ready for mixing.
If the track uses shared memory (static mode) and the sound is looped, this track will play for ever.
The fix consists in removing the track from active list immediately if the track is terminated.
Change-Id: I4582aa1d981079ab79be442fb6185f5afaed5cf3
[Sorted|Keyed]Vector<TYPE> would leak their whole storage when resized
from the end and TYPE had trivial dtor and copy operators.
Change-Id: I8555bb1aa0863df72de27d67ae50e20706e90cf5
Vector::sort() is using _do_copy() incorrectly; _do_copy() calls the
copy constructor, not the assignment operator, so we need to destroy
the "destination" before copying the item.
Change-Id: Iaeeac808fa5341a7d219edeba4aa63d44f31473c
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.
This addresses a few parts of the bug:
- There was a small issue in the window manager where we could show a window
too early before the transition animation starts, which was introduced
by the recent wallpaper work. This was the cause of the flicker when
starting the dialer for the first time.
- There was a much larger problem that has existing forever where moving
an application token to the front or back was not synchronized with the
application animation transaction. This was the cause of the flicker
when hanging up (now that the in-call screen moves to the back instead
of closing and we always have a wallpaper visible). The approach to
solving this is to have the window manager go ahead and move the app
tokens (it must in order to keep in sync with the activity manager), but
to delay the actual window movement: perform the movement to front when
the animation starts, and to back when it ends. Actually, when the
animation ends, we just go and completely rebuild the window list to
ensure it is correct, because there can be ways people can add windows
while in this intermediate state where they could end up at the wrong
place once we do the delayed movement to the front or back. And it is
simply reasuring to know that every time we finish a full app transition,
we re-evaluate the world and put everything in its proper place.
Also included in this change are a few little tweaks to the input system,
to perform better logging, and completely ignore input devices that do not
have any of our input classes. There is also a little cleanup of evaluating
configuration changes to not do more work than needed when an input
devices appears or disappears, and to only log a config change message when
the config is truly changing.
Change-Id: Ifb2db77f8867435121722a6abeb946ec7c3ea9d3
In practice, no one ever writes an apostrophe in an aapt string with the
intent of using it to quote whitespace -- they always mean to include a
literal apostrophe in the string and then are surprised when they find
the apostrophe missing. Make this an error so that it is discovered
right away instead of waiting until late in QA or after the strings have
already been sent for translation. (And fix a recently-introduced string
that has exactly this problem.)
Silence the warning about an empty span in a string, since this seems to
annoy people instead of finding any real problems.
Make the error about having a translated string with no base string into
a warning, since this is a big pain when making changes to an application
that has already had some translations done, and the dead translations
should be removed by a later translation import anyway.
In AudioFlinger::MixerThread::putTracks(), change the mFillingUpStatus flag to FS_FILLING for active tracks so that mute request is executed without ramping volume down when the track is moved from A2DP to hardware output.
Also modified AudioFlinger::setStreamOutput() so that the notification of the change is sent only once to AudioSystem.
(in this case the state is dumped without the proper locks held which could result to a crash)
in addition, the last transaction and swap times are printed to the dump as well as the time spent
*currently* in these function. For instance, if SF is unresponsive because eglSwapBuffers() is stuck,
this will show up here.
what happened is that the efective pixel format is calculated by SF but Surface nevew had access to it directly.
in particular this caused query(FORMAT) to return the requested format instead of the effective format.
this would happen is the window is made visible but the client didn't render yet into it. This happens often with SurfaceView.
Instead of filling the window with solid black, SF would simply ignore it which could lead to more disturbing artifacts.
in theory the window manager should not display a window before it has been drawn into, but it does happen occasionnaly.
Apparently the problem is caused by the fact that A2dpAudioStreamOut::standby() calls a2dp_stop() after the headset has been powered down.
The workaround consists in indicating to A2DP audio hardware that a close request is pending and that stanby() must be bypassed.
This change makes SurfaceHolder.setType(GPU) obsolete (it's now ignored).
Added an API to android_native_window_t to allow extending the functionality without ever breaking binary compatibility. This is used to implement the new set_usage() API. This API needs to be called by software renderers because the default is to use usage flags suitable for h/w.
This is because the AudioFlinger duplicating thread is closed while the output tracks are still active. This cause the output tracks objects to be destroyed at a time where they can be in use by the destination output mixer.
The fix consists in adding the OutputTrack to the track list (mTracks) of its destination thread so that a strong reference is help during the mixer processed and the track is detroyed only when safe by destination thread.
Also added detection of problems when creating the output track (e.g. no more tracks in mixer). In this case the output track is not added to output track list of duplicating thread.
When changing the audio output stream sampling rate with setParameters() make sure that all tracks have a sampling rate less or equal to 2 times the new output sampling rate.
Merge change 7419 from master that may help eliminate the problem.
This change was for a different use case (when disabling A2DP to switch output to SCO) but without a repro case it is worth trying.
The BT headset detection now makes the difference between car kits and headsets, which can be used by audio policy manager.
The headset connection is also detected earlier, that is when the headset is connected and not when the SCO socket is connected as it was the case before. This allows the audio policy manager to suspend A2DP output while ringing if a SCO headset is connected.
There was no garanty that the corresponding thread destructor had been already called when exiting the closeOutput() or closeInput() functions.
This contructor could be called by the thread after the exit condition is signalled. By way of consequence, closeOutputStream() could be called after
we exited closeOutput() function.
To solve the problem, the call to closeOutputStream() or closeInputStream() is moved to closeOutput() or closeInput().
The function checkForNewParameters_l() is called with the ThreadBase mutex mLock locked. In the case where the parameter change implies
an audio parameter modification (e.g. sampling rate) the function sendConfigEvent() is called which tries to lock mLock creating a deadlock.
The fix consists in creating a function equivalent to sendConfigEvent() that must be called with mLock locked and does not lock mLock.
Also added the possibility to have more than one set parameter request pending.
Use integers instead of void* as input/output handles at IAudioFlinger and IAudioPolicyService interfaces.
AudioFlinger maintains an always increasing count of opened inputs or outputs as unique ID.
* changes:
update most gl tests to use EGLUtils
added two EGL helpers for selecting a config matching a certain pixelformat or native window type
added NATIVE_WINDOW_FORMAT attribute to android_native_window_t
The major things going on here:
- The MotionEvent API is now extended to included "pointer ID" information, for
applications to keep track of individual fingers as they move up and down.
PointerLocation has been updated to take advantage of this.
- The input system now has logic to generate MotionEvents with the new ID
information, synthesizing an identifier as new points are down and trying to
keep pointer ids consistent across events by looking at the distance between
the last and next set of pointers.
- We now support the new multitouch driver protocol, and will use that instead
of the old one if it is available. We do NOT use any finger id information
coming from the driver, but always synthesize pointer ids in user space.
(This is simply because we don't yet have a driver reporting this information
from which to base an implementation on.)
- Increase maximum number of fingers to 10. This code has only been used
with a driver that reports up to 2, so no idea how more will actually work.
- Oh and the input system can now detect and report physical DPAD devices.
If the output stream handler passed was not the A2DP output stream, the request was ignored instead of being forwarded downstream to hardware interface.
there was several issues:
- when a surface was made non-current, the last frame wasn't shown and the buffer could stay locked
- when a surface was made current the 2nd time, it would not dequeue a new buffer
now, queue/dequeue are done when the surface is made current.
for this to work, a new query() hook had to be added on android_native_window_t, it allows to retrieve some attributes of a window (currently only width and height).
