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bug: 10244066
Change-Id: I317f05b146db962c271893f6208890a5a6c396f1
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This is needed for the MediaTimeProvider java interface, so it does not
continually poll for current media time.
Note: NuPlayer and AwesomePlayer do not correctly handle stop (pause
instead), so for those we will signal PAUSED.
Signed-off-by: Lajos Molnar <lajos@google.com>
Change-Id: I3c61e1bda475f131323f475c18a42e3ec66c9ae1
Bug: 10326117
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Main change is to how recycled tracks are used for gapless
playback. If we are playing offloaded tracks that can't be
recycled we don't open a new offloaded output until we have
closed the previous one. This is because offloaded tracks
are a limited resource so we don't want to spuriously create
unnecessary instances. If the tracks cannot be recycled
this means that the formats are incompatible and so the
hardware most likely will also be unable to use the existing
output channel for the new track. If we already have the
maximum number of hardware offload channels open (which could
be only one) then creation of the next output would fail if
we attempted it while the previous output was still open.
Change-Id: I4f5958074e7ffd2e17108157fee86329506730ea
Signed-off-by: Eric Laurent <elaurent@google.com>
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Also fixes some bugs in ParseMessage and adds "WebSocket" mode to
ANetworkSession, something that's opt-in and should not affect existing clients
of the API.
Change-Id: I86d5748e0d818231d85d3590d86c2b41d4f8b1f1
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Bug: 9805979
Change-Id: I77b19d6a65ff9fb72e7428ce79b117628e4c8658
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Change-Id: Ifbf3a46a4183c8abc0feee1c588953ab10303cc1
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RTSPSource"
* commit '9f0e05978e236352fd2a4a7c6dce909755c36ead':
Prevent double disconnects in RTSPSource
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* commit '7266bc80850dc327e9f18c27b81613f959399630':
Prevent double disconnects in RTSPSource
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* commit 'ca7f160921a4a1a5da32ebe8fab1067bd8998a1f':
Prevent double disconnects in RTSPSource
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Prevent a null pointer from occuring by checking that we are not
already disconnected before performing a disconnect, in case
multiple disconnects have been issued.
Change-Id: Ib88800cd246086fd57f77d10748d41b1aefe2084
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crashes" into jb-mr2-dev
* commit '9a25c012915f2209fff1479b246ae8eba38c3eff':
Part of fix for TEE resource leaks when app crashes
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into jb-mr2-dev
* commit 'eced745af7d2d27a8c45e04eb071d4fb30cd3e48':
Part of fix for TEE resource leaks when app crashes
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bug: 9672606
Change-Id: I98cf65260d5e43d34e3897452f6f26ed2c89294e
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Remove old includes.
Header files only include other header files that they directly need themselves.
Change-Id: Ic471386808d9f42ea19ccbd59cb50a5f83a89dd0
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incorrect format specifier for type \'ssize_t\'."
* commit '903871da7a7c71b8718133e60e2f0835837b5f3f':
Fix compilation warning: incorrect format specifier for type 'ssize_t'.
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type \'ssize_t\'."
* commit 'bdf33c98912090176183d3e9b9a9217dbbdca6da':
Fix compilation warning: incorrect format specifier for type 'ssize_t'.
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warning: format '%ld' expects argument of type 'long int',
but argument 7 has type 'ssize_t {aka int}' [-Wformat]
Change-Id: I823835a26320a3abb82f8dff9fbe82e5132e7556
Signed-off-by: Scott Fan <fancp2007@gmail.com>
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NOTE: this does _not_ include all private member variables added
to classes as part of offload support. Only public/protected functions
and stubs functions/variables needed to make the changes buildable.
- isOffloadSupported() added to audio policy service
A stub implementation is required to build, this always returns false
- setParameters() added to IAudioTrack
A stub implementation is required to build, this always returns
INVALID_OPERATION
- CBlk flag for stream end
- Change AudioSystem::getRenderPosition() to take an audio_output_t
so caller can specify which output to query
- Add AudioSystem::isOffloadSupported()
This is fully implemented down to the AudioFlinger function
AudioPolicyServer::isOffloadSupported() which is just a stub
that always returns false.
- Add EVENT_STREAM_END to AudioTrack interface.
STREAM_END is used to signal when the hardware has actually finished
playing all the data it was sent.
- Add event type enumeration to media player interface AudioSink callbacks
so that the same callback can be used to handle multiple types of
event. For offloaded tracks we also have to handle STREAM_END and
TEAR_DOWN events
- Pass audio_offload_info_t to various functions used for opening outputs,
tracks and audio players. This passes additional information about the
compressed stream down to the HAL when using offload.
For publicly-available APIs this is an optional parameter (for some of
the internal and low-level APIs around the HAL interface it is mandatory)
- Add getParameters() and setParameters() API to AudioTrack
Currently dummy implementations.
- Change AudioPlayer contructor so that it takes a set of bitflags defining what
options are required. This replaces the original bool which only specified
whether to use deep buffering.
- Changes to StageFright class definition related to handling tearing-down of
an offloaded track when we need to switch back to software decode
- Define new StageFright utility functions used for offloaded tracks
Currently dummy implementations.
- AudioFlinger changes to use extended audio_config_t.
Fills in audio_offload_info_t member if this info is passed in when
opening an output.
- libvideoeditor changes required to add the new event type parameter
to AudioSink callback functions
- libmediaplayerservice changes required to add the new event type parameter
to AudioSink callback functions
Change-Id: I3ab41138aa1083d81fe83b886a9b1021ec7320f1
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Eric Laurent <elaurent@google.com>
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This change prepares for the new implementation of AudioTrack client, which
will require clients to use only sp<AudioTrack>, not raw AudioTrack *.
A raw delete will cause a race condition during AudioTrack destruction.
AudioTrack was made a RefBase by commit b68a91a70bc8d0d18e7404e14443d4e4020b3635
on 2011/11/15, when it was needed by OpenSL ES (for the callback protector).
At that time, the only other client that was also converted from
AudioTrack * to sp<AudioTrack> was android.media.AudioTrack JNI in
project frameworks/base (file android_media_AudioTrack.cpp).
Details:
* Use .clear() instead of delete followed by = NULL.
* ALOG %p need .get().
* sp<> don't need to be listed in constructor initializer, if initially 0.
* Use == 0 for sp<> vs == NULL for raw pointers.
* Use if (sp != 0) instead of if (raw).
Change-Id: Ic7cad25795d6e862e112abdc227b6d33afdfce17
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playback inside stagefright.
* commit 'f9bb8358e454be6dc88403171db7e0e182407f17':
Update HTTP proxy configuration for all media playback inside stagefright.
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stagefright.
* commit 'b7319a7eb0a06ef4fd3a0c9157ee63e637ad7aa1':
Update HTTP proxy configuration for all media playback inside stagefright.
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Change-Id: Ie0dd00045aba668d8b49da73224e7a7c9c04f69b
related-to-bug: 8873723
(cherry picked from commit 2704965b8a1ff3b7450ff58ccecf86d8ec688c40)
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initiated and then" into jb-mr2-dev
* commit 'dd309b97f3e30cb8ab30892b4bf6b5d824853a86':
Handle the case where an asynchronous prepare was initiated and then
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the mediaplayer was reset.
Change-Id: Ib241747c5dc002b88a3854569c1f8340b2a8ef41
related-to-bug: 8688986
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* commit '628a8c0ed91531468f5c615fea8745f63cc812aa':
fix a race condition of MediaPlayerServce::Client::mClient
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some cases.
related-to-bug: 8630032
Change-Id: I8e94b53b34e137e827e9630c65f3252ea91e4ebd
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Change-Id: I01aa0e47b55d0dffe34525edf9f055a5cb4dc70f
related-to-bug: 8620223
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if no duration information was available. This prevents us from
entering ERROR state, effectively rendering the player instance useless.
Change-Id: I602d2661ae8b8633360306c0ea9208fb11e2bf17
related-to-bug: 8596285
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Bug: 8580410
Change-Id: If493d87d60d71be664ad75b140c62acadb75b0d0
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jb-mr2-dev
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Change-Id: I007f147d693664e777b8758be2bb8a4c7ec0236b
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Clarify offline usage of sessions and keys and implement
implement CryptoSession to support additional crypto use
cases.
Change-Id: I5d8000ce7e1dd7eba08969fc50296c9e1456c4fc
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Change-Id: I0e52352845398a4db074e939487f6f6de94bd523
related-to-bug: 8225122
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The constructor calls reset(), which in turn calls stop(), which then
accesses mCaptureTimeLapse before it has been initialized.
Change-Id: Ia94ac740b9bd1a0389c72647a5639dd25320d92c
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Currently crypto plugins are expected to be in libdrmdecrypt.so.
When there are multiple plugins supporting different schemes,
this approach requires source code integration across
vendors which is unmanagable. Also, for integration with
MediaDrm where the crypto keys are obtained from a drm server,
the MediaCrypto plugin needs to interoperate with the MediaDrm
plugin. This change allows {MediaCrypto, MediaDrm} pairs that
are logically related to be implemented in a common shared
library.
Change-Id: I7f6638f29171f91609fc2d944396365568630b56
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Change-Id: I9ff8eeb7d0c383b5c0c68cd54eb54ce7d2d22fe6
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jb-mr2-dev
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Change-Id: I13d7a9553aa335bca790a3a59d389d7533c83d57
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commit f4edf442741886cdbe071e2d15f6e6247269f7c5
Author: Andreas Huber <andih@google.com>
Date: Tue Mar 12 09:09:18 2013 -0700
Pass additional flags to the sink, use TCP by default in wolfiecast mode.
Change-Id: I41e11a2375d4199656e45c4f149d8441d0016092
commit 6302602ed280a38287f507159abfb40a1da38c5a
Author: Andreas Huber <andih@google.com>
Date: Tue Mar 12 08:51:58 2013 -0700
tweaks
Change-Id: Ie29e422d7258be522f4bb1f6c5afcf74c937e547
commit a38a860e4979ba563cadbaafa21b084439449d26
Author: Andreas Huber <andih@google.com>
Date: Mon Mar 11 16:57:43 2013 -0700
Report average lateness all the way from NuPlayerRenderer...
Change-Id: I2e7700703ae656515e44b9c25610d26c75778111
commit a7d49b11675ea88be4029dd8451d1649db94571d
Author: Andreas Huber <andih@google.com>
Date: Mon Mar 11 14:54:19 2013 -0700
Make TimeSyncer smarter, enable TunnelRenderer
Change-Id: I27377a60cd8feb01589da456967fddd34532c20e
commit 0f214c8ef68179f7b61512c37040939554013151
Author: Andreas Huber <andih@google.com>
Date: Thu Mar 7 15:57:56 2013 -0800
convert source timestamps to sink timestamps, report lateness.
Change-Id: I051a60fbbceca2f7b508ae3dac6e01e402bae39e
commit 04a4f8e16bad09157b5615a5fa45310438955832
Author: Andreas Huber <andih@google.com>
Date: Thu Mar 7 09:00:28 2013 -0800
Sync time between sink and source.
Change-Id: Ie8b4d75c957aa48310e7c81d1279761b9f821efe
commit aebe20e6184e3636a99082f8ece08e708015cb8d
Author: Andreas Huber <andih@google.com>
Date: Wed Mar 6 09:03:12 2013 -0800
play with back pressure
Change-Id: I51eb69257e6a79e76f5f9c75ff99d8adbd083947
Change-Id: Ifdf57228667fed7fc71c5090a2c3f7cea1037c5c
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Camera:
- Signal to AppOpsService when camera usage starts and stops
- Listen to permissions revocations and act on them
- Currently just kill camera connection when permissions lost
Stagefright:
- Pass on client name, UID to camera as needed
Bug: 8181262
Change-Id: I9e33c9d05e9daa77dbb2d795045d08eb887ec8f0
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Change-Id: I4e8a8b20914cb64edc37abe68233fbc9f2b5d830
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When a stream is paused, RTSP Pause is also sent to the server.
Otherwise the buffering might continue until the memory runs out.
When the stream is resumed, RTSP Play will be sent in order to
resume the buffering.
Change-Id: I5dc1761140827c532451638c3fd3f34271e5b9ab
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Added buffering start and end notifications for RTSP.
MEDIA_INFO_BUFFERING_START is sent when buffering is started
and MEDIA_INFO_BUFFERING_END is sent when the buffer has
filled up.
This patch also adds RTSP end of stream handling.
EOS is signalled when BYE is received OR when
detecting end of stream even if no actual EOS is received.
Change-Id: I5cccb6845060ae6afd66d9f735b89da81476cd13
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has successfully completed and a sufficient amount of data fetched,
and only then signals that preparation is completed.
Change-Id: I7684a14238b826909f518f2af506966e522dfcfc
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has successfully completed and only then signals that preparation is
complete.
Change-Id: I1a60f718e673fe1462c69369c40eafbed6a14326
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