In constructors, initialize member fields in the initialization list
rather than constructor body where possible. This allows more fields
to be const, provided they are never modified.
Also initialize POD fields in constructor, unless it's obvious they
don't need to be initialized. In that case, put a comment instead.
Remove explicit clear() in destructors on fields that are now const.
Give AudioSessionRef a default constructor, so it's immutable fields can
be marked const.
Add comment about ~TrackBase() trick.
Initialize fields in declaration order to make it easier to confirm that
all fields are set.
Move initialization of mHardwareStatus from onFirstRef() to constructor.
Use NULL not 0 to initialize raw pointers in initialization list.
Rename field mClient to mAudioFlingerClient, and getter from client()
to audioFlingerClient().
Change-Id: Ib36cf6ed32f3cd19003f40a5d84046eb4c122052
Don't check that pointer is non-NULL before delete.
Don't leave deleted member fields non-NULL, except in a destructor,
since it could be misleading in a dump or debugger. (mRsmpOutBuffer)
Change-Id: Ic0492a6b752f74a67f4c96dfb89ca2de4e69eecf
Using the builtin is faster on some platforms, for example on ARM it's
19 instructions instead of 13, and is O(1) instead of O(n). Of course,
track creation is an inherently slow operation, so this doesn't matter
much now. But if we add support for virtual tracks, then physical tracks
will be allocated/freed more frequently. Also just on principle ...
Change-Id: I3f590934092bd7a1869cbedbc7357928aa5cc8ff
Several source files privately defined macros LIKELY and UNLIKELY in terms
of __builtin_expect. But <cutils/compiler.h> already has CC_LIKELY and
CC_UNLIKELY which are intended for this purpose. So rename the private
uses to use the standard names.
In addition, AudioFlinger was relying on the macro expanding to extra ( ).
Change-Id: I2494e087a0c0cac0ac998335f5e9c8ad02955873
The calculation done in prepareTracks_l() for the minimum amount
off frames needed to mix one output buffer had 2 issues:
- the additional sample needed for interpolation was not included
- the fact that the resampler does not acknowledge the frames consumed
immediately after each mixing round but only once all frames requested have been used
was not taken into account.
Thus the number of frames available in track buffer could be considered sufficient although
it was not and the resampler would abort producing a short silence perceived as a click.
Issue 5727099.
Change-Id: I7419847a7474c7d9f9170bedd0a636132262142c
Replace series of if/then/elses by easier-to-read switch. Also return
void instead of status_t, since callers weren't checking it. Assert on
bad input parameters.
Change-Id: Ie1f0a297977b28501d20e1af819afed9b4750616
Return void, not status_t, from setActiveTrack and setBufferProvider.
These methods returned status_t, but the callers never checked the
return value. Since these aren't externally visible APIs, they now
return void, and assert on bad input parameters.
Change-Id: I530ed29484596ae41e8659826ca425149c51c2a1
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 problem is that when an AudioRecord using the resampler is restarted,
the resampler state is not reset (as there is no reset function in the resampler).
The consequence is that the first time the record thread loop runs, it calls the resampler
which consumes the remaining data in the input buffer and when this buffer is released
the input index is incremented over the limit.
The fix consists in implementing a reset function in the resampler.
A similar problem was also present for playback but unoticed because the track buffer is always
drained by the mixer when a track stops. The only problem for playback was that the initial
phase fraction was wrong when restarting a track after stop (it was correct after a pause).
Change-Id: Ifc2585d685f4402d29f4afc63f6efd1d69265de3
When resampling too short sound, AudioMixer uses previous
tracks buffer. So we re-initialize the temporary buffer per
loop to avoid it.
Change-Id: I55a59a3b14faa8445e09c450478fe79cef704760
moved surfaceflinger, audioflinger, cameraservice
all native services should now reside in this location.
Change-Id: Iee42b83dd2a94c3bf5107ab0895fe2dfcd5337a8
