Fix the trivial warnings that are hidden by the use of -isystem to
include frameworks/av and caught by -Werror.
Test: m -j checkbuild
Bug: 31751828
Change-Id: I155f9f772ce0a84b364313814cb7cf528b9de4eb
Normally the flush is processed on ObtainBuffer.
But that is not called when the track is paused.
So it was not possible to flush and re-prime a
track that was paused.
Now we flush synchronously for inactive tracks,
and flush from prepareTracks_l for active tracks.
Bug: 19193985
Bug: 27791443
Change-Id: I39a7e4921e45041c3a51cf91fd3995b5edee6dd4
Signed-off-by: Phil Burk <philburk@google.com>
Provide server timestamps if the HAL doesn't provide it.
Provide monotonic - boottime translation.
Bug: 17472992
Bug: 26682703
Bug: 27749434
Change-Id: I6c9b213d9f9284092e34d57f52870e02c72df62a
The server was waiting for a full buffer.
But the buffer was only getting partly filled.
So the stream was not starting.
The fix involves having the server look at the adjustable threshold.
Bug: 27505889
Change-Id: I5dbf686413e670dacbbecc9e0f838744e465f44f
Signed-off-by: Phil Burk <philburk@google.com>
When audio flinger mixer removes an AudioTrack from the
active list in case of underrun, it is possible that the
client has written a full buffer just after the underrun detection and
is blocked waiting for more space to write. In this case, the client
will never detect the DISABLED flag and the track never be restarted.
Also implement missing DISABLE flag detection in server side audio tracks
(OutputTrack and PatchTrack).
bug: 27567768
Change-Id: I8d0753429d4113498258b1f61bd8ac5939a612f0
Provide server timestamps if the HAL doesn't provide it.
Provide monotonic - boottime translation.
Integrate record timestamps and playback timestamps together.
Bug: 17472992
Bug: 22871200
Bug: 26400089
Bug: 26682703
Change-Id: If1974f94232fcce7ba0bbcdf63d9e54ed51918ff
Also add getBufferCapacityInFrames().
These can be used to dynamically raise or lower latency.
Bug: 21019153
Signed-off-by: Phil Burk <philburk@google.com>
Change-Id: I02ca7f6f5cc4e089fcd81cc8a2b6ff234e0381a8
Use this struct to handle the parameters for TimestretchBufferProvider all
across the system.
Add stretch mode and fallback mode to TimestretchBuffer Provider.
Change-Id: I19099924a7003c62e48bb6ead56c785cb129fba2
Allow restoration of loop and position.
Make position and loop synchronously readable.
Bug: 17964637
Change-Id: I8cfb5036e665f55fdff5c67d27e1363ce9a8665d
StaticAudioTrackServerProxy::framesReady() previously returned
only the contiguous frames, update to return the total
available frames. This resolves short-count looping in
SoundPool for FastTracks.
Also (1) Removes the racy condition of reading two variables
and (2) Fixes buffer->mNonContig to return the correct value
and (3) Restores behavior that loop count of 1 goes back to
loopStart once during playback.
Bug: 11830751
Bug: 12070295
Bug: 17456842
Change-Id: I64906e6036bb00a1d7375b03efe6deb69d6478ca
This structure is passed between 64 and 32 bit processes via shared
memory, so ensure it's the same size, and that the members we care
about are in the same place.
Bug: 17569156
Change-Id: Id776bc825af1fbf43a6dd3407cca064f6d932902
Add support for audio device connections between different audio
hw modules.
The patch is performed by creating a bridge between the playback
thread connected to the sink device and the record thread connected
to the source device using a pair of specialized PlaybackTrack and
RecordTrack.
- Added PatchTrack and PatchRecord classes.
- Added TrackBase type to indicate more clearly the track behavior.
- A TrackBase can allocate the buffer or reuse an existing one.
- Factored some code in openOutput() and openInput() for internal use
by PatchPanel.
Bug: 14815883.
Change-Id: Ib9515fcda864610458a4bc81fa8f59096ff4d7db
This will allow (eventually) a greater dynamic range for gains.
However there are still a few remaining places in effects and mixer
that will also need to be changed in order to get the full benefit.
Also fixes a minor bug: was not checking for NaN in AudioTrack C++.
Change-Id: I63bce9e82e0a61546d8ff475fb94bcb700d99c96
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
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
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
using a new timestamp latch in PlaybackThread, and
AudioTrackServerProxy::framesReleased() which returns mServer.
Change-Id: I1ebfba968c773faaab95648c272fd3ebd74718d6
This fixes a regression that was introduced earlier
by commit 9f80dd223d
called "New control block for AudioTrack and AudioRecord".
That commit broke underrun reporting for fast tracks.
Also remove Track::mUnderrunCount, which counted the number of underrun
events, and was only used by dumpsys media.audio_flinger.
Now dumpsys media.audio_flinger reports the number of underrun frames,
Isolated underrun-related control block accesses via the proxy, so that
the server is not directly poking around in the control block.
The new proxy APIs are AudioTrackServerProxy::getUnderrunFrames() and
AudioTrackServerProxy::tallyUnderrunFrames(). getUnderrunFrames() returns
a rolling counter for streaming tracks, or zero for static buffer tracks
which never underrun, but do a kind of 'pause' at end of buffer.
tallyUnderrunFrames() increments the counter by a specified number of frames.
Change-Id: Ib31fd73eb17cbb23888ce3af8ff29f471f5bd5a2
This is part of a series of CLs to clean up the shared memory
control block, by removing any fields that don't have to be there.
Change-Id: I6e51003a1293b6800258c31b22cff2eba42162e7
- start() returns a status so that upper layers can
recreate a non offloaded track in case of error.
- Added states to handle offloaded tracks specific:
- waiting for stream end (drain) notification by
audio flinger
- allow pause while waiting for stream end notification
- getPosition() queries the render position directly from
audio HAL.
- disable APIs not applicable to offloaded tracks
- Modified track restoring behavior for invalidated
offloaded tracks: just send the callback and wait for
upper layers to create a new track.
- Added wait for stream end management in audio track client
proxy. Similar to obtainBuffer and should be factored in.
Change-Id: I0fc48117946364cb255afd653195498891f622bd
Signed-off-by: Eric Laurent <elaurent@google.com>
- Added specialized playback thread class for offload playback,
derived from directoutput thread.
This thread type handles specific state transitions for offloaded
tracks and offloading commands (pause/resume/drain/flush..) to audio HAL.
As opposed to other threads, does not go to standby if the track is paused.
- Added support for asynchronous write and drain operations at audio HAL.
Use a thread to handle async callback events from HAL: this avoids locking
playback thread mutex when executing the callback and cause deadlocks when
calling audio HAL functions with the playback thread mutex locked.
- Better accouting for track activity: call start/stop and release Output
methods in audio policy manager when tracks are actually added and removed
from the active tracks list.
Added a command thread in audio policy service to handle stop/release commands
asynchronously and avoid deadlocks with playback thread.
- Track terminated status is not a state anymore. This condition is othogonal
to state to permitted state transitions while terminated.
Change-Id: Id157f4b3277620568d8eace7535d9186602564de
Glenn Kasten
Andy Hung
Colin Cross
Phil Burk
Eric Laurent
Ricardo Garcia
Marco Nelissen
Elliott Hughes
Richard Fitzgerald