| Commit message (Collapse) | Author | Age | Files | Lines |
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Eventually we may want to use uint64_t, but will need to confirm atomicity.
Bug: 12381724
Change-Id: Ia2c591d262d22b47b6f7dab4b9d9faa14b86d865
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Contains the necessary changes to make frameworks/av build and work
on a 64-bit machine.
Signed-off-by: Craig Barber <craig.barber@arm.com>
Signed-off-by: Kévin PETIT <kevin.petit@arm.com>
Signed-off-by: Ashok Bhat <ashok.bhat@arm.com>
Signed-off-by: Marcus Oakland <marcus.oakland@arm.com>
Change-Id: I725feaae50ed8eee25ca2c947cf15aee1f395c43
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Unless AudioFlinger was built with FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE
enabled, AudioFlinger would deny using the fast path (and internally
fall back to the normal codepath) when it realized that resampling
was required. Since the buffer size calculations within AudioFlinger
don't take resampling into account properly (see the calculation
below "AUDIO_OUTPUT_FLAG_FAST denied" in audioflinger/Threads.cpp,
just below the hunk that this patch changes), make sure AudioTrack
doesn't try to use the fast path if resampling is required.
This removes the possibility to enable
FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE in AudioFlinger since it
AudioTrack now won't even try to use the fast path for content
that requires resampling, regardless of the AudioFlinger configuration.
Change-Id: Icf0f8ad50bf0fdb84657f518c0120aa0535f23f9
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Do partial read in MemoryLeakTrackUtil dumpMemoryAddresses
to avoid using more memory than what is allocated.
Change-Id: I94feb4e00647407f938571167b981c7371f39e3d
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When restoring an AudioTrack, the next position callback point
should not be modified and set ahead of current buffer head.
Otherwise, as frames are dropped, the new position is never reached
and an application relying on position callbacks to reload the buffer
would be stalled.
Bug: 11868603.
Change-Id: I93b2a311642a0c89944b78bcc0482d4ceed98ae4
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AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
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Change-Id: I158f147295eebcea96e4047d7618069bc48bdd7d
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* commit '9cae217050aa1347d4ac5053c305754879e3f97f':
Assign blame for playback wakelocks.
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Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Cherrypicked from master.
b/9464621
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
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* commit '95c03858e2ab4fb693a2bfe47b3caa806e43c044':
Allow releaseBuffer after flush
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After AudioTrack start checks for pending flush,
allow releaseBuffer on any previously obtained buffer.
For example, this can happen if the resampler has obtained
a buffer but not released the whole buffer yet.
Note that the resampler will be reading obsolete data.
Bug: 11285590
Change-Id: I0614fbb62e43604aac3089cce4b7797c87a306b5
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start" into klp-dev
* commit 'f8f15b05fe051009945c9042a1a9260280e0feb2':
Fix race condition in AudioTrack::pause followed by start
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Bug: 11148722
Change-Id: Iec88f00c8510363d4418e4b8d5b34feb06ecf04d
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* commit '120a88471a607c85c4d60300d73c3be0a1e8f8c8':
AudioTrack: fix head position after restore
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The head position transfered to the new track
by restoreTrack_l() must take into account the frames that
are dropped from the old track to avoid a non recoverable
offset in the playback head position returned to applications.
Bug: 11230062.
Change-Id: I51143a08b95e8f264ed709ae2054360315f2b8b1
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* commit 'b3cb72a17d9a472883e9e2faa18b42eac533fe99':
SoundPool: handle new audio track event
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If the AudioTrack is torn down, SoundPool will never
receive the buffer end event and the track will stay active
for ever.
The fix consists in stopping the AudioTrack when a new audiotrack
event is received.
Bug: 11193583.
Change-Id: I9876eb2a8f75c601368f669acd67b0accf6e2736
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* commit 'fee4ce338d78eeb58af1f66831ead53322d3859e':
Cleanup openRecord error handling
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Bug: 10888816
Change-Id: I84897dd7d30b370640b54e928f230604b873cb68
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* commit '56ce726019f700a95ce5b45beebceadae4836e30':
IOMX: Add prepareForAdaptivePlayback method
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prepareForAdaptivePlayback is the fallback mechanism to support
seamless resolution change for devices that do not support dynamic
output buffers. It is up to the codecs to handle this appropriately,
but codecs that do not handle dynamic output buffers would
request enough buffers up to the requested size in this method
to avoid port reconfiguration on resolution changes.
Change-Id: I58d4aa8ef1359ea3472735bbe9140c3132039b3d
Signed-off-by: Lajos Molnar <lajos@google.com>
Bug: 10192531
Related-to-bug: 7093648
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into klp-dev
* commit '1adf20ce868b80a24f7387daa6549364d5509c6a':
fix volume and effect enable delay on offloaded tracks
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Volume: add a method to wake up the mediaserver playback
thread when a volume command is received on an offloaded track.
Effects: call effect chain process on offloaded playback threads
asynchronously from writes to allow effect state updates while
waiting for async write callback.
Bug: 10796540.
Change-Id: Id2747ae88783575d1d7ffd6fc86fbd054ab2c739
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klp-dev
* commit '1c7f35d1f25eb7160314fdef536463fc34deb1ea':
soundpool: allocate shared memory heap by client
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Current SoundPool implementation allocates the shared memory heap
containing decoded PCM samples in mediaserver process.
When mediaserver process crashes, the shared memory heaps allocated by
AudioCache cannot be mapped anymore in the new instance of mediaserver.
This causes a silent failure to end playback of new sounds because
AudioFlinger believes the new AudioTracks are opened in streaming mode
and not static mode: it sees a NULL shared memory pointer when the track
is created.
The fix consists in allocating the memory heap in the client process. Thus
the heap is not lost when mediaserver restarts. The global memory usage is
the same as this is shared memory.
Also added a way to detect that a shared memory is passed when the track is
created but cannot be mapped on mediaserver side.
Also fix a crash in SoundPool when ALOGV is enabled.
Bug: 10894793.
Change-Id: Ice6c66ec3b2a409d75dc903a508b6c6fbfb2e8a7
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klp-dev
* commit '402dfba6dcd68f5fd8d8921f9751f3e47eb1449d':
Add support for level measurements in Visualizer
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New commands to set a measurement mode and perform peak + RMS
measurements.
Bug 8413913
Change-Id: Ib25254065c79d365ebb34f9dc9caa0490e2d300d
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* commit '1d2536f460d4678770f423f50cbf6a61a13d4d11':
AudioTrack: fix music resume
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Fix regression introduced by commit 5a6cd22 in AudioTrack resume:
the callback thread was not signaled if paused internaly.
Bug: 10895013.
Change-Id: Ic356b115132d6fccbcee2d9bb855e92671dc20c5
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klp-dev
* commit '4b701cc041d635e5ec56e382043a4c5d01aedd80':
Revert "Workaround slow AudioTrack destruction"
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This reverts commit 8bbbd7da02fac3de40139af19f7cf7a7cc3cc824.
Change-Id: I269a6c445cbce33451b6a9e74223e36e6abbdbe0
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klp-dev
* commit '3b3cfcfa272c8e3e16c89765b8817f5a8de0c505':
Fix slow AudioTrack and AudioRecord destruction
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There were two causes for the slowness:
When thread was paused, it used nanosleep and sleep. These usually
run to completion (except for POSIX signal, which we avoid because it
is low-level). Instead, replace the nanosleep and sleep by condition
timed wait, as that can be made to return early by a condition signal.
Another advantage of condition timed wait is that a condition wait was
already being used at top of thread loop, so it is a simpler change.
The AudioRecord destructor was missing a proxy interrupt that was correct
in AudioTrack. This proxy interrupt is needed in case another thread
is blocked in proxy obtainBuffer.
Does not address the 1 second polling for NS_WHENEVER.
Bug: 10822765
Change-Id: Id665994551e87e4d7da9c7b015f424fd7a0b5560
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* commit 'feb6d27bf61cd266cf753215e9cae16b9bc9dbbd':
Workaround slow AudioTrack destruction
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Bug: 10809586
Change-Id: I5f30d4deb1233e8ade8967568e40684ef680c395
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* commit '9a98b6de791aeb130192df10744f5b35f8b6ef1a':
Partial fix for SoundPool not terminating
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SoundPool was waiting for EVENT_UNDERRUN only to indicate end of clip. In
J, AudioTrack delivered both EVENT_UNDERRUN followed by EVENT_BUFFER_END.
However, as of K, AudioTrack is only delivering EVENT_BUFFER_END (this
lack of EVENT_UNDERRUN is another bug which still needs to be fixed).
The workaround is to also respond to EVENT_BUFFER_END in SoundPool.
Bug: 10787103
Change-Id: Id68a23bddd6dd9df6c49c55138197260d71ca468
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klp-dev
* commit '18f861404efc054da0a2ea6c582e293940f63bc8':
Fix underruns when fast track denied due to SRC
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OpenSL ES requests a fast track. If sample rate conversion is needed,
the request is denied by server, and a larger client buffer is used
to handle the higher latency of a normal track. However the client
notification period was calculated based on buffer being divided into
2 sub-buffers. That resulted in the notification period being too long.
The server pulls chunks that are smaller than half the total buffer.
So now the client uses 3 sub-buffers when there is SRC.
Also removed the 'defer wake' optimization because it was incorrect.
This optimization attempted to reduce the number of wakeups of client,
when server releaseBuffer knows that another releaseBuffer will be
following. But there is no way for the first releaseBuffer to predict
how soon the second releaseBuffer will occur. In some cases it was
a long time, and the client underran. So now the client is woken up
immediately if the total number of available frames to client is >=
the minimum number the client wants to see (the notification period).
Also fix bug where minimum frame count was not being used in the
calculation of notification period.
Bug: 10342804
Change-Id: I3c246f4e7bc3684a344f2cf08268dc082e338e2a
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storage. So it can not guarantee works well when multithread environment. AudioFlinger has multithread. so strtok_r is more safe."
* commit 'fc270954192ef7e15ac2c88daadd8890d22096e3':
strtok stores its values in thread local storage. So it can not guarantee works well when multithread environment. AudioFlinger has multithread. so strtok_r is more safe.
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can not guarantee works well when multithread environment. AudioFlinger has multithread. so strtok_r is more safe."
* commit 'e56f3c96fa6e7550b67e8b049f999aaa2ada1192':
strtok stores its values in thread local storage. So it can not guarantee works well when multithread environment. AudioFlinger has multithread. so strtok_r is more safe.
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So it can not guarantee works well when multithread environment.
AudioFlinger has multithread.
so strtok_r is more safe.
Change-Id: I6d77ef9cc49a4478dd856dcdca14e4920ce955c6
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The change I7370d6e59a7ef26dfb284a8b058d5ab2e0a42ccf caused a regression
in SoundPool looping when using SoundPool's streaming implementation.
This reverts a portion of that change.
Bug: https://code.google.com/p/android/issues/detail?id=58113
Bug: 10171337
Change-Id: I8af0dc8683a7c7f225c80f0eb4d39770667b52e5
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This is done by configuring SoundPool for shared memory and fast track.
Previously SoundPool used a streaming track, and looping in streaming
mode relied on the ability to loop the most recently enqueued data.
That 'feature' was lost in the new implementation of streaming, so we're
now switching from streaming mode to shared memory mode. Shared memory
mode had always been desired, but was blocked by bug 2801375 which is fixed now.
Bug: 10171337
Change-Id: I2a938e3ffafa2a74d5210b4198b50db20ad5da0e
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unsupported" into klp-dev
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