Current AudioTrack implementation enforces that the requested audio
buffer size is at least corresponding the audio latency.
This requirement is too strong and leads to problems with current
stagefright and AudioSink implementations when playing over output
streams with long latency.
Ultimately, the AudioSink design should be changed to specify a minimum
buffer size in time or frames units but not in buffer count units.
Change-Id: I8ba603956f92ac49143a8249572665aa548f2f0f
Conflicts:
media/libmedia/AudioTrack.cpp
Move in the direction of a more publishable API for configuring a
media player for retransmission. It used to be that we used a custom
invoke and a modified URL (prefixed with aahTX://). There are many
issues with this technique and it was never meant to stand the test of
time.
This CL gets rid of all that. A new (but currently hidden) method was
introduced to the java level MediaPlayer API, called
setRetransmitTarget(InetSocketAddress), which allows an app writer to
set the retransmit target. For now, this method needs to be called
before a call to setDataSource (which is pretty unusual for the
MediaPlayer API) because this mid level code uses this as a cue to
instantiate an aahTX player instead of relying on the data source to
select a player. When retranmit functionality becomes part of the
existing android player implemenation, this
set-retrans-before-set-data-source behavior can go away, along with
the aahTX player itself.
Change-Id: I6ab07d89b2eeb0650e634b8c3b7a0b36aba4e7dd
This reverts commit 64006cb1642b2ec0ee74c66007d869b884391fd1.
Back out this change in order to get ready to implement a longer term,
more media-team approved way of selecting a retransmit player.
Change-Id: I97b68b9859a174eab858598cb00d4445a14fbc17
* Added a MediaPlayer.setMediaPlayerType API that be called to specify the
desired media player implementation before calling setDataSource
* Implemented setDataSource(fd) in the AAH_TxPlayer
Change-Id: I359075d9c7d6fd699dda14eb85ec50da19307639
we need to transition to executing state anyway to be able to properly flush/shutdown
in the future.
Change-Id: Ie48bc09ea31942009ae3a5a45aabc9ffad9fb91f
related-to-bug: 5655016
* commit '481ffa505bb1d8f5089ea98e3b5960d409b6819c':
Fix for issue 5309336 -add videoeditor maximum prefetch YUV frames in media_profiles.xml to limit the total memory usage.
-add videoeditor maximum prefetch YUV frames in media_profiles.xml to limit the total memory usage.
Change-Id: I41ffbc192fcce4c7635e5b0a1f2835852e5ee509
RTP library used to broadcast media from one device to a collection
of listeners. Handles failures/retries/etc.
This is a squashed merge from master-tungsten of the following changes:
commit e1a5101fe627d71739a7c4263bb3a65c7bc44385
Author: Jason Simmons <jsimmons@google.com>
Date: Fri Aug 12 13:24:21 2011 -0700
Hold ThreadWrapper in a ref-counting pointer
Change-Id: Iaf3343182e37bcc0ca99fbaf8f9bbb8c4984072a
commit 89b90d62e164ff3db27c9cba85255fc476d2dd96
Author: Jason Simmons <jsimmons@google.com>
Date: Wed Aug 10 13:08:25 2011 -0700
Update the Tungsten TX player to use HTTPBase
Change-Id: I9f7ecf1b4b496cec1815284dbcdb958a43284169
commit 43be3231034ff8537fdd84422a7954780038671f
Author: John Grossman <johngro@google.com>
Date: Mon Jun 27 18:59:12 2011 -0700
Move libaah_rtp over from the vendor directory.
Also move factor PipeEvent out into utils.
Change-Id: Id3877c66efe22d771cf3ef4877107e431b828e37
Change-Id: I5fe1ea941c09204d7b33f15f4e2b2ab320dc468b
Signed-off-by: Mike J. Chen <mjchen@google.com>
Signed-off-by: John Grossman <johngro@google.com>
Signed-off-by: Jason Simmons <jsimmons@google.com>
Squashed merge from master-tungsten of the following changes:
commit 73d09e18c4557e583a1684d44d598a1a02fd0cf2
Author: John Grossman <johngro@google.com>
Date: Mon Jun 20 13:57:44 2011 -0700
Remove TungstenMisc and rename LinearTransform
Change-Id: Ie8aa3e24e09fdbf6ef8996c26deb9c5640e20d1b
commit 3114aabe76ad733b59929d87e49c68229f5ae2e8
Author: John Grossman <johngro@google.com>
Date: Fri Jun 3 10:47:16 2011 -0700
Name changes and spelling fixes.
+ Replace the term TungstenTime with the Eugene-approved term CommonTime.
+ Fix a spelling error in a comment I noticed.
Change-Id: I8c10d618206826d16055f78c7724e24443bb03fd
commit cbf2903ab6893b6e662514e2f6d670e268a419df
Author: John Grossman <johngro@google.com>
Date: Fri Apr 15 09:27:54 2011 -0700
Migrate Tungsten code from the HC-Tungsten to the Master-Tungsten branch.
Change-Id: I95372d913a0761d90168edb4016f5ece0ea74502
commit bc7c46aa629f9883e959ef23de8da297f9eb508b
Author: Jason Simmons <jsimmons@google.com>
Date: Mon Jun 20 13:59:17 2011 -0700
Create a separate class for timed AudioTracks
commit 43be3231034ff8537fdd84422a7954780038671f
Author: John Grossman <johngro@google.com>
Date: Mon Jun 27 18:59:12 2011 -0700
Move libaah_rtp over from the vendor directory.
Also move factor PipeEvent out into utils.
Change-Id: Id3877c66efe22d771cf3ef4877107e431b828e37
commit 17526eb3148c9c3d4365b6d5b47e8dc13bca71b6
Author: John Grossman <johngro@google.com>
Date: Mon Jun 27 17:06:49 2011 -0700
Name changes for the TRTP Players s/tungsten/aah/g
Change-Id: I55e9ad13003f6aa6a36955b54426a7efbe31ac51
commit 423fc1bfc0fda799c421a650c83c4b9293b1a08c
Author: Jason Simmons <jsimmons@google.com>
Date: Mon Jun 20 17:56:09 2011 -0700
More timed AudioFlinger changes requested by code review:
* change trimTimedBufferQueue to trimTimedBufferQueue_l
* create one timed audio buffer heap per client process instead of one per track
* grow the silence buffer on demand
* some error handling fixes in timed getNextBuffer
* calculate the next output PTS in all mixer and track hooks
Change-Id: Ifc51a08b55029b7c48902ab2f22933ad7bafe1ad
commit a148e2674b1d3cb73289b82b85c333f0a66824a9
Author: John Grossman <johngro@google.com>
Date: Mon Jun 20 17:02:24 2011 -0700
Move the A@H time service into frameworks/base
Change-Id: I5c570cde70e8931e205516cb33517585804ce841
commit dfa438fa49bdaeeb2ec5fd0d17b30d881608b6b1
Author: John Grossman <johngro@google.com>
Date: Mon Jun 20 11:55:36 2011 -0700
Fix the build after Mike's code moving.
Change-Id: Ia883643ded252168bcc5a70584ab6ce97bb05266
commit 04489474ec8e73efe1bf52918831f41659033162
Author: John Grossman <johngro@google.com>
Date: Fri Jun 17 14:19:50 2011 -0700
Refactor the local/common clock services.
This change is one of a set of 5 changes made to different repositories. Look
for this comment in all of them.
Refactor the local/common clock services in tungsten to match android best
practice. Notable changes include
+ The kernel no longer knows anything about common time. Common time has been
moved completely up into user land. This has an impact on the accuracy of the
timesync debugging code, and the netfilter assisted approach to network based
timesync is going to have to be modified.
+ The timesync driver used by A@H is now just local time driver.
+ The kernel no longer needs access to the linear transform math code, and it
has been removed.
+ A new HAL has been introduced to expose the concept of local time to the
system.
+ A non-slewable stub implementation of the local time HAL based on
CLOCK_MONOTONIC has been added.
+ The TungstenTime library has been eliminated. Its functionality has been
distributed among the common time binder service, the local time hal and the
linear transform utility code.
+ All clients of the old TungstenTime library have been changed to be clients of
the binder service, the hal and the utility code.
+ The reset_tt utilities have been removed, they no longer have a purpose in the
system.
+ more progress has been made in eliminating the word "tungsten" from the code.
Things left to do include
+ Finish getting rid of tungsten from the time service.
+ Move the time service into the framework; AudioFlinger's new timed mode
depends on it and the service cannot continue to live in vendor tungsten.
Change-Id: I999b6cfb4a9d267818a86d747c35eecfc6693101
commit d48194545eed1116a84d81e2fb53315d2b0701a7
Author: Jason Simmons <jsimmons@google.com>
Date: Thu Jun 16 14:22:46 2011 -0700
Change the interface of the AudioMixer and AudioBufferProvider to accept a presentation timestamp
Change-Id: Ice2df5628d45a7f77100e7008103b35b3d3160a4
commit 02561419db82b01ffb28df38000716c612988427
Author: John Grossman <johngro@google.com>
Date: Tue May 10 14:00:21 2011 -0700
Put in a hack for controling master volume in the policy manager.
Fix initial master volume reporting.
Change-Id: Ia6caf2bbc6083c5f99fab852baa40fff10fc5fc7
commit 549cdc3ba115dc654cdade261fb055c72c6cdb79
Author: John Grossman <johngro@google.com>
Date: Wed May 4 11:46:17 2011 -0700
Make certain the logic for computing the output stream mixing point is hardened
against underflow and overflow when input and output sample rates don't match.
Change-Id: I5ebea07c9938107b435bec7413418622767e4e16
commit 8043d8ed63f51e76d452d22be7d453d4a7794530
Author: Jason Simmons <jsimmons@google.com>
Date: Wed Apr 27 18:06:27 2011 -0700
Add the patch for timed audio support to the mono resampler
Change-Id: I526f34ae9d1e8e3b0ed2fb05af3d024d5c5fe711
commit 2be89486ef23f0b0b0cc2dc25a4c0ee691043f00
Author: John Grossman <johngro@google.com>
Date: Wed Apr 27 10:38:57 2011 -0700
Extend the AudioHWInterface to allow it to specify the initial master volume used by AudioFlinger.
Change-Id: I8823330801c927494cf7ca31a6b8f9264fbfbb26
commit ff89a4d5e37e6a05a2b03f79ab4e97833dd66393
Author: John Grossman <johngro@google.com>
Date: Wed Apr 27 09:07:14 2011 -0700
Fix an issue with inconsistent volume reporting.
Changed masterVolume() to return the same value as the last call
to setMasterVolume when the HW layer is implementing master
volume control. The masterVolume/setMasterVolume API seems to be
an idea which was abandonded a long time ago; as of today the
system only ever sets it to 1.0 at startup and then never changes
it. Until we can figure out how the concept of external
amplifier gain control fits into the Android audio framework,
Tungsten is exposing this API via a hack-tastic invoke back door
in the TungstenRXPlayer and needs the getter/setter results to be
consistent.
Change-Id: I2ac730fa8fc9ee28c88f1a8e6f2e493eb5b65544
commit 086511b2d19cceb976747ac23e12b73fc7c28bea
Author: Jason Simmons <jsimmons@google.com>
Date: Mon Apr 25 16:07:19 2011 -0700
Add handling of timed audio tracks in the generic resampling mixer
Change-Id: Ic3be1d21b1117f1b233808be543c28a0dcec4792
Change-Id: I6ec5d2bca9b8ebc0acd395a7dd92e1a48fcdfa9b
Signed-off-by: Mike J. Chen <mjchen@google.com>
Signed-off-by: John Grossman <johngro@google.com>
Signed-off-by: Jason Simmons <jsimmons@google.com>
This change moves the ANativeWindow connect and disconnect logic from
MediaPlayer to MediaPlayerService::Client.
Bug: 5502654
Change-Id: Ifc43b98b01ad8f35d62d7ece43110724ec7fda3d
This change makes OMXCodec push RGB 565 buffers filled with black to an
ANativeWindow when tearing down after decoding to protected gralloc
buffers. This allows the OMX tear down to zero out any protected
buffers that were used without the possibility that the buffer is still
being used by SurfaceFlinger or HWComposer.
Bug: 5483222
Change-Id: I8acedd81a7bb67dfdc2fd15733e3375b6ce8d560
This change fixes an issue in Stagefright where the state of an OMXCodec
object can get out of sync with the state of the OMX component. In
particular, if one of the ANativeWindow functions failed and put the
OMXCodec into the ERROR state, this would cause Stagefright to skip
doing the Executing -> Idle transition. Without this transition the
freeBuffersOnPort call would never be made, and the MediaBuffers would
end up being leaked (which would also leak the Gralloc buffers they
reference).
Bug: 5333695
Change-Id: I85ea0cf92d18e7ef6d35c7d1e2a7b4e2c9745d34
The native_window_set_crop() is called when port reconfig event callback comes from decoder's and
crop parameters are changed from default getconfig() OMX_IndexConfigCommonOutputCrop values.
Since the default crop params are same as port reconfig crop params, the native_window_set_crop()
is not called, hence resulting in displaying the whole frame(paddedWidth x paddedHeight).
By calling native_window_set_crop() during initilaization of output port of decoder ensures
in setting up ANative window to crop region.
Change-Id: I68926464a1f5c7e6053804615c8b9bd32ea85688
Signed-off-by: Lakshman Gowda <lakshman79@ti.com>
Return BAD_VALUE error upon detection of wrongly formatted files.
The client should abort the initialization upon error detection.
The current CHECK() interrupts the configurecodec() preventing a graceful
exit.
Change-Id: Ic79313fa76a63284897df5d91635de87d06f3100
Signed-off-by: Gilles-Arnaud Bleu-Laine <gilles@ti.com>
Fixed problem in AudioTrack::restoreTrack_l() causing a permanent
failure if the IAudioTrack interface to AudioFlinger could not be
restored at the first attempt.
Change-Id: I039d4fe2dca8d3baf71f1a6c51119f27a67b6611
The list of directories to skip are configurable via setprop.
The main motivation is that some test data folder takes long time
to scan, and media scanner may compete for CPU time against perf
tests therefore skewing the results.
Bug: 5263115
Change-Id: I568213e2a4babf6033021c1d336ef0347c0e3315
