| Commit message (Collapse) | Author | Age | Files | Lines |
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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
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Change-Id: I7d85e04fb9f0caa8460a111ca6181bba6f6497ac
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Change-Id: Ib3982a9c960bfdb0cb7e1b174440b141b194cfbe
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of the dependency on jni.h"
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dependency on jni.h
o related-to-bug: 6214141
Change-Id: I548d84a9888be884d3903dc9dea9235258792165
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Current method implemented by the visualizer to detect that audioflinger has
stopped providing audio buffers does not work if the application
reads pcm captures too fast.
The fix consist in implementing a method based on real time measurement only.
One drawback is that the new method makes use of system calls that add some
overhead to the process and capture functions.
Change-Id: I53bd596b856f1cc7f0f47e08413af3335227100b
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Reorganize SoundPool and JetPlayer code to be ready for the
creation of libmedia_native.
Split SoundPool between libsoundpool (JNI) and libmedia(sound pool implementation).
Remove dependencies on nativehelper/jni.h from JetPlayer.
Change-Id: I130c6014173b714329929dd82c5dfb70b757a610
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* changes:
libmedia: remove skia include
stagefright: remove dependency on skia
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skia is not used in this file, remove the unnecessary include.
Change-Id: Ib4d0e0c0090c6b37ff8cfb816c0d8ba82a9638a4
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skia is only used to write a jpeg file, link directly to libjpeg
instead.
Change-Id: Iea8f4a2347c38328776541d2b74bcbdea3f62041
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Currently not implemented by client or server
Change-Id: Ib11dda57db3eeb871bcc7b546e340078776875f5
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createTrack and openRecord don't need the "old" flags parameter,
which was either audio_policy_output_t or audio_in_acoustics_t
shifted left by 16 bits. But they do need "new" flags, which
are defined by the application use case. Initially, the only
application use case flag is timed output, but others are planned.
For output, the audio_policy_output_t flags are passed to
AudioSystem::getOutput, which returns an audio_io_handle_t, and that
handle is then passed to createTrack. So createTrack doesn't need the
old flags parameter.
For input, the audio_in_acoustics_t flags are passed to
AudioSystem::getInput, which returns an audio_io_handle_t, and that
handle is then passed to openRecord. So openRecord doesn't need the
old flags parameter.
Change-Id: I18a9870911846cca69d420c19fe6a9face2fe8c4
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jni.h is not used in IOMX.h and is not available in pdk builds,
remove it.
Change-Id: I9bc8fd70f617942712d9f684c6fc927bf18be753
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o related-to-bug: 6214141
Change-Id: Ic88d1732b3e014af47532a0809e01f6086e8464d
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Change-Id: I83ad2735eaf8a8dfa5f8f29f30aec1311b3222de
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This reverts commit 0a3edd38df0743dcc7091bb7ebf29e7e7dadc7cb
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This reverts commit c920dee060ac69684be33210ee44b99a5fc3e8b2
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Change-Id: I72474c17757dba5867f55b0e99e76e9e4e32ce7b
related-to-bug: 6217289
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Fix a mistake which came in as part of a merge conflict resolution
during code review of the recent unicast mode refactor of LibAAH_RTP.
Nop packet which were supposed to carry TS transformations for the
pause state accidentally got flagged as Flush operations. The flush
packet successfully carried the TS transformation, but also had the
undesired side effect of constantly flushing the stream.
Change-Id: I4c6aa0043fc274a1d7e880ed1d19cf277f22194b
Signed-off-by: John Grossman <johngro@google.com>
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EOS was being treated as a flush operation which was causing problems.
In particular, the transmitter was delcaring that playback was
complete early (by the clock lead time of the system, which was 1
second in this case). Also, the receiver was treating the EOS message
just like the flush message, immediately destroying the substreams
associated with the program without letting them play out first.
Change the transmitter to send the EOS message like it always does,
but have it wait until the media time of the last sample has arrived
before reporting playback complete to the app level of things.
On the receiver side of things, don't treat the EOS message like the
flush message. Instead, have the EOS message simply put the substream
into EOS mode, allowing it to signal EOS to its decoder and shut off
the isAboutToUnderflow hack.
Change-Id: Ibe3ac01044373f83edb7a5f4b70478bd78c16d11
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Bionic/Android support eventfd, so there is really no reason to have
PipeEvent around any more. This change gets rid of it in LibAAH_RTP
and replaces it with eventfds.
Change-Id: I841fcb71bf5015d521d7517c69f44eac0ea92278
Signed-off-by: John Grossman <johngro@google.com>
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Add support for unicast mode to the AAH RXPlayer. At the API level,
things should be pretty simple. To use unicast mode, instead of
passing the multicast address and port in the data source URL, just
pass the unicast address and port of the transmitters command and
control port. For example, instead of
aahRX://224.128.60.5:8867
one might instead pass
aahRX://192.168.63.5:55476
Change-Id: I7b40716983d7a91def86dcf40f093dda4255aae3
Signed-off-by: John Grossman <johngro@google.com>
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Fix a bug discovered while working on adding unicast mode to the TX/RX
players. Also some general cleanup/consolidation regarding timeout
code.
The bug went like this. When a TX player had hit EOS, it would send
an EOS command payload to its receivers. Later, when application
level code shutdown and cleaned up the player, it would send another.
In situations where there is massive packet loss, there is a chance
that not only did both of the EOS packets get dropped, but that they
never got filled in by the retry algorithm because the receiver gave
up on the RTP gap due to an aboutToUnderflow situation in at least one
of its active substreams.
When this happens, there are two major problems. First, all of the
substreams associated with the TX player which has now gone away have
become effectively leaked. They will only get cleaned up if the
entire RTP stream (the TX Group) goes away for 10 seconds or more, or
when the RX Player itself is reset by application level code or a
fatal error. These substreams are holding decoder and renderer
resources which are probably in very short supply, which is a Bad
Thing.
Second, there is now at least one substream in the RX player which is
never going to receive another payload (its TX player source is gone),
but is still considered to be active by the rx player. Assuming that
this substream's program was in the play state when the track ended,
there is now at least one substream which is always
"aboutToUnderflow". From here on out, when the retry algorithm is
attempting to decide whether or not it has the time to attempt to fill
in a gap in the muxed RTP sequence, it always decides that it does not
have the time because of the orphaned substream which is stuck in its
about to underflow state. This effectively means that the retry
algorithm is completely shut off until the rx player gets reset
somehow (something which does not happen during normal operation).
Since the environment had to be extremely lossy to trigger this chain
of events in the first place, and its probably no better now, your
playback is just going to be chock full of gaps which produces
horrible stuttering in the presentation stage of the system.
Two new failsafes have been introduced to keep the double EOS drop
from causing this. First, a timeout has been introduced on the
substream level, in addition to the already existing RTP level
timeout. If a substream fails to receive an activity for 10 seconds
(same timeout as the master RTP timeout), it will be automatically
flushed and purged.
Second, the nature of the master RTP timeout on the transmitter side
has been changed. Instead of just sending an empty NOP command packet
to indicate that the main RTP stream is still alive, the transmitter
now sends a new time of command packet; the Active Program Update
packet. This packet contains a list of all the active program ID
attached to this TX group. Upon receiving one of these APU packets,
RX players reset the inactivity timers for all substreams which are
members of the programs listed in the packet, but they also
immediately purge any substreams associated with programs not present
in the APU.
Between the two of these, no matter how nasty and selective the packet
smashing gremlins in your system happen to be, substreams will always
eventually clean up and avoid getting stuck in a perma-stutter
situation.
Also in this CL:
+ Extract some common utility code into a utils.cpp file so that it
can be shared across the library.
+ Stop using custom timeout logic in the RXPlayer. Instead, use the
common Timeout helper class in utils.cpp.
Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I350869942074f2cae020f719c2911d9092ba8055
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Significantly refactor the TXGroup code to allow transmit groups to
operate in a unicast fanout mode in addition to the traditional pure
multicast mode. Important changes include...
+ Each transmit group active in the system now has its own socket to
send and receive traffic on. In the past, this socket was used to
listen for retry requests from clients. Now it is also used to
listen for group membership reports (IGMPv3 style) from unicast
clients. Having an individual socket per transmit group allows
unicast clients to join the group needing only the IP address and
port of the transmitters socket, and not needing any additional
"group id" to be sent to the client beforehand.
+ Setup for the transmitter is now slightly different. As before, to
setup for multicast mode, a user can call setRetransmitEndpoint
passing an IPv4 multicast address and specific port to transmit to.
It used to also be the case that a user could pass a specific
unicast address and port to transmit to as well. This is no longer
allowed. Instead, to operate in unicast mode, a user passes 0.0.0.0
(IPADDR_ANY) as the IP address. In addition, they need to pass
either 0 for a port to create a new unicast mode TX group, or they
need to pass a specific port to cause the player to attempt to use
an existing unicast mode TX group. The specific port should be the
command and control port of the TX group which was bound to when the
group was originally created.
+ A magic invoke was added to allow clients to fetch the command and
control port on which a TX Player's TX Group is listening.
The API described above is most likely temporary and should eventually
be replaced with one where TX groups are formal top level objects with
their own independent interface and life-cycle management.
Signed-off-by: John Grossman <johngro@google.com>
Change-Id: Ib4e9737c10660d36c50f1825c9824fff5390b1c7
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Rename AAH_TXSender to AAH_TXGroup in preparation for refactoring to
support unicast retransmission.
Signed-off-by: John Grossman <johngro@google.com>
Change-Id: I3984db27d1c61c6155d5d7cb9c38eead421b9249
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ics-mr1
* commit '14958e21c12f922d7501d32c3bec05109eb342d5':
audioflinger: fix issue with camcorder and A2DP
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* commit '2a0d685ed62ff7a5e5a40be0748860c092165984':
MediaPlayerService: fix AudioSink latency
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* commit '3fe7ee651db0aae9485ead227c89db1e24b9e245':
AudioTrack: relax check on minimum buffer size
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* commit '4d7bc65538c7cd9fbb1fbbcf22d1da47fcee1219':
Doc change: String changes for Android Market
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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
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transmitter media player."
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
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Change-Id: Ie8298eb9d253fc6ede448da87660a60d23170987
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Add a bandaid to prevent a segfault which can occur while handling
timed audio buffers. There is a deeper problem which should
eventually be addressed, but for now this fix should prevent any
crashing.
The deeper problem is as follows.
When the AudioFlinger mixer gets data to mix from an AudioTrack, it
ends up getting a structure filled out which points into an IMemory
region owned by the AudioTrack. Unfortunately, this structure is not
holding a refcount on the IMemory which it points into. If the
IMemory refcount hits 0 and the chunk of RAM is retuned to the binder
heap it came from, there can still be a Buffer object being held by
the AudioFlinger mixer which points into the region of memory which
was retuned to the binfer heap. If AF reads from this buffer, it
could read corrupt data (if the region of memory gets handed back out
to a writer), or it could segfault (if the heap has been freed and the
pages unmapped). Similar problems could happen if AF attempts to
write to the buffer, heap corruption in one case, segfaulting in the
other.
In the past, this has not been an issue for AF, because tracks
allocate a single IMemory (which serves as a ring buffer) and the
IMemory lives for as long as the track lives. As an artifact of the
way the code came out, the mixer cannot be holding a Buffer structure
pointing into the IMemory which used to be owned by a track if the
track no longer exists. Tracks cannot come into or out of existence
during a mix operation, which is the only thing which makes this safe.
TimedTracks work differently, however. Timed tracks each allocate a
small binder heap, and then hand out IMemory instances broken out of
this heap. The heap lives as long as the track, so the worst which
could happen here is that a TimedTrack's IMemory gets returned to the
heap while there is still a buffer structure in flight pointing into
the memory region, then the region gets handed out again and
overwritten by new data causing the mixer to mix the wrong audio. The
timing to cause this to happen is very difficult to encounter, and you
to generate the timing conditions required, you need to be in a pretty
bad failure state where audio is already breaking up and skipping, so
its unlikely that anyone would notice (which is why I'm band-aiding
the segfault and letting the deeper issue slide for now).
In general, however, it might be a good idea to revisit this buffering
design. On principal, if someone is going to hold pointers into a
refcounted object, they should be holding a ref on the object at the
same time. Failure to do this will usually lead to a situation where
there are corruption or segfault issues, or to a system where the
refcounted object's lifetime must be implicitly managed very carefully
in ways which are usually non-obvious and are easy to break by new
engineers on a project.
Change-Id: Ib391075395ed0ef46a03c37aa38a82d09e88abeb
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Check the string returned by a HAL's implementation of get_parameters
for NULL before attempting to make use of it. That way, we won't
bring down the mediaserver because of a poorly written HAL.
Change-Id: I45f0fd3de003151f98acf32c36c42f58d053f3a0
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Change the CCHelper class to be an instanced instead of a static
pattern. The CCHelper instances all share an interface to the common
clock service and register/unregister a callback handler in response
to there being CCHelper instance in the system or not. This brings
usage of the CCHelper into like with the new auto-disable
functionality of the common time service. For any given process,
whenever there are CCHelper instances active, the process will
maintain a callback target to the common clock service and will be
considered to be an active client.
Also change all of the users of the CCHelper interface to manage the
lifecycle of their new CCHelper instances.
Change-Id: I7c28c5d70d9b07ba7407b4ac706e7e7d7253001b
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Define a native service interface for configuring and controlling the
common time service. Implement the native marshallers and stub the
implementation of the new interface.
Change-Id: Ia6a6a20ef3d221e8829c55be1dd5f98ed996c610
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Add marshallers and stub implementations for new methods in the common
clock interface to support new functionality being added in the
process of integrating the common time service more closely with the
Java level of Android.
Change-Id: Iac2d3fb405d1b64cea1d8e13f988160afb76a06d
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Change the ICommonClock marshallers to return an explicit error with
each transaction. This makes it easier to disambiguate between binder
errors (dead object due to the service crashing) vs. runtime errors
(cannot return common time due to lack of sync)
Change-Id: I2182cf0cfb2e11b4669f392554f104ef7c4bca8b
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Move the State enum up to the ICommonClock interface so it can be
returned for status/debugging up to clients.
Change-Id: I81fef5b96ffc69a4f2e9801b3744feea099ccd47
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Bulk name change to remove references to Android@Home from the common time
service in preparation for cleanup and up-integration into the master
branch. Basically, aah_timesrv is now common_time.
Change-Id: I3d3db212f96e8ba171aa36b9c58e27e4a336cb0a
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