WebView needs more fine-grained control over the behavior of the
framework upon execution of the display lists. The new status_t
allows WebView to requests its functor to be re-executed directly
without causing a redraw of the entire hierarchy.
Change-Id: I97a8141dc5c6eeb6805b6024cc1e76fce07d24cc
Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
Fortunately audio_track_cblk_t doesn't have a destructor, but for clarity
remove the double destruction.
Also add warning not to add any virtuals to audio_track_cblk_t.
Change-Id: I70ebe1a70460c7002145b2cdf10f9f137396e6f3
Always read and write track volumes atomically. In most places this was
already being done, but there were a couple places where the left and
right channels were read independently.
Changed constant MAX_GAIN_INT to be a uint32_t instead of a float.
It is always used as a uint32_t in comparisons and assignments.
Use MAX_GAIN_INT in more places.
Now that volume is always accessed atomically, removed the union
and alias for uint16_t volume[2], and kept only volumeLR.
Removed volatile as it's meaningless.
In AudioFlinger, clamp the track volumes read from shared memory
before applying master and stream volume.
Change-Id: If65e2b27e5bc3db5bf75540479843041b58433f0
Add an API to control block for getting/setting send level.
This allow us to make the mSendLevel field private.
Document the lack of barriers.
Use 0.0f to initialize floating-point values (for doc only).
Change-Id: I59f83b00adeb89eeee227e7648625d9a835be7a4
except in the control block, where we don't have room.
In AudioFlinger::ThreadBase::TrackBase::getBuffer,
read the frame size from control block only once.
Change-Id: Id6c4bccd4ed3e07d91df6bbea43bae45524f9f4e
Added Surface.setPosition(float, float) which allows to set a surface's
position in float.
Bug: 5239859
Change-Id: I903aef4ad5b5999142202fb8ea30fe216d805711
Add the concept of synchronous dequeueBuffer in SurfaceTexture
Implement {Surface|SurfaceTextureClient}::setSwapInterval()
Add SurfaceTexture logging
fix onFrameAvailable
Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.
The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.
Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
The first fix (commit 913af0b4) is problematic because it makes threads
in mediaserver process block on the cblk mutex. This is not permitted
as it can cause audio to skip or worse have a malicious application
prevent all audio playback by keeping the mutex locked.
The fix consists in using atomic operations when modifying the control
block flags.
Also fixed audio_track_cblk_t::framesReady() so that it doesn't block
when called from AudioFlinger (only applies when a loop is active).
Change-Id: Ibf0abb562ced3e9f64118afdd5036854bb959428
This change fixes the stability problems experienced when using
a bluetooth headset supporting both A2DP and SCO. Problems occur
when starting the video chat at which time the A2DP output is being
stopped to start SCO. At that time, active AudioTracks are invalidated
by AudioFlinger so that a new AudioTrack binder interface can be
recreated by the client process on the new mixer thread with correct parameters.
The problem was that the process to restore the binder interface was not
protected against concurrent requests which caused 2 binder interfaces
to be created sometimes. This could lead to permanent client deadlock
if one of the client threads was waiting for a condition of the first
created binder interface while the second one was created (as the AudioFlinger
would only signal conditions on the last one created).
This concurrent request situation is more likely to happen when a client
uses the JAVA AudioTrack as the JNI implementation uses simultaneously the
native AudioTrack callback and write push mechanisms. By doing so, the code
that checks if the binder interface should be restored (in obtainBuffer()) is
much more likely to be called concurrently from two different threads.
The fix consists in protecting the critical binder interface restore phase
with a flag in the AudioTrack control block. The first thread acting upon the binder
interface restore request will raise the flag and the second thread will just wait for
a condition to be signaled when the restore process is complete.
Also protected all accesses to the AudioTrack control block by a mutex to prevent
access while the track is being destroyed and restored. If a mutex cannot be held
(e.g because we call a callback function), acquire a strong reference on the IAudioTrack
to prevent its destruction while the cblk is being accessed.
Modified AudioTrack JNI to use GetByteArrayElements() instead of
GetPrimitiveArrayCritical() when writing audio buffers. Entering a critical section would
cause the JNI to abort if a mediaserver crash occurs during a write due to the AudioSystem
callback being called during the critical section when media server process restarts.
Anyway with current JNI implementation, either versions do not copy data most of the times
and the criticial version does not guaranty no data copy.
The same modifications have been made to AudioRecord.
Change-Id: Idc5aa711a04c3eee180cdd03f44fe17f3c4dcb52
now that we removed the notion of a "inUse" buffer in surfaceflinger
a lot of code can be simplified / removed.
noteworthy, the whole concept of "unlockClient" wrt. "compositionComplete"
is also gone.
Change-Id: I210413d4c8c0998dae05c8620ebfc895d3e6233d
There is a new ANativeWindow::cancelBuffer() API that can be used to
cancel any dequeued buffer, BEFORE it's been enqueued. The buffer is
returned to the list of availlable buffers. dequeue and cancel are not
mutually thread safe, they must be called from the same thread or
external synchronization must be used.
Change-Id: I86cc7985bace8b6a93ad2c75d2bef5c3c2cb4d61
This issue showed that when an AudioTrack underruns during a too long period
of time and is therefore disabled by audioflinger mixer, it takes an additional
delay of up to 3 seconds to recover.
This fix adds a simple mechanism to recover immediately when the client application
is ready to write data again in the AudioTrack buffer
Also throttle warnings on record overflows
Change-Id: I8b2c71578dd134b9e60a15ee4d91b70f3799cb3d
this situation happened when the last buffer needed to be resized
(or allocated, the first time). the assumption was that the buffer
was in use by SF itself as the current buffer (obviously, this
assumption made no sense when the buffer had never been allocated, btw).
the system would wait until some other buffer became the "front" buffer.
we fix this problem by entirely removing the requirement that the
buffer being resized cannot be the front buffer. instead, we just
allocate a new buffer and replace the front buffer by the new one.
the downside is that this uses more memory (an extra buffer) for a
brief amount of time while the old buffer is being reallocated and
before it has actually been replaced.
Change-Id: I022e4621209474ceb1c671b23deb4188eaaa7285
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
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
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
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
