| Commit message (Collapse) | Author | Age | Files | Lines |
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Set a work source for the playback wakelock, so that playback is
counted against the requesting app instead of the media server.
Cherrypicked from master.
b/9464621
Change-Id: I7329f88a288a95a582a78005a1c3d16a5a611e31
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Current SoundPool implementation allocates the shared memory heap
containing decoded PCM samples in mediaserver process.
When mediaserver process crashes, the shared memory heaps allocated by
AudioCache cannot be mapped anymore in the new instance of mediaserver.
This causes a silent failure to end playback of new sounds because
AudioFlinger believes the new AudioTracks are opened in streaming mode
and not static mode: it sees a NULL shared memory pointer when the track
is created.
The fix consists in allocating the memory heap in the client process. Thus
the heap is not lost when mediaserver restarts. The global memory usage is
the same as this is shared memory.
Also added a way to detect that a shared memory is passed when the track is
created but cannot be mapped on mediaserver side.
Also fix a crash in SoundPool when ALOGV is enabled.
Bug: 10894793.
Change-Id: Ice6c66ec3b2a409d75dc903a508b6c6fbfb2e8a7
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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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Remove old includes.
Header files only include other header files that they directly need themselves.
Change-Id: Ic471386808d9f42ea19ccbd59cb50a5f83a89dd0
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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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Change-Id: Ie0dd00045aba668d8b49da73224e7a7c9c04f69b
related-to-bug: 8873723
(cherry picked from commit 2704965b8a1ff3b7450ff58ccecf86d8ec688c40)
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Change-Id: I9ff8eeb7d0c383b5c0c68cd54eb54ce7d2d22fe6
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support of _decryption_ modules in addition to what we already supported.
Change-Id: Ic37b87dc170ba8def3817991d25df798f21e950b
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Just get the parameter on server side
Change-Id: I433a63104dbb257e0d862be2ab61847cb36d1c15
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The C++ class names don't match what the classes do, so rename
ISurfaceTexture to IGraphicBufferProducer, and SurfaceTexture to
GLConsumer.
Bug 7736700
Change-Id: I64520a55f8c09fe6215382ea361c539a9940cba5
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Change-Id: Ia010e7a00534f9356b3247369d0ffd65591d91aa
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Change-Id: I866768b1e3f3b232f1934a35b65f66befc12f3f6
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Change-Id: I582ed000026bba6d116db8304e15a3c52f8a9a01
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interface.
Change-Id: I7d3c44cb79cd40e73499f2d7ccf35c69b628e6d7
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Change-Id: I08f17efa0c7d007e17408feb7d4fbef0a19f531a
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Add the ability to dynamically register low level MediaPlayer
factories which will be probed at setDataSource time to determine the
proper MediaPlayerBase to instantiate.
This change is in preparation for moving libaah_rtp out of
frameworks/base and into phantasm platform directory.
Change-Id: Icf8904db3ab9e3c85df6e780d5546d9988cb9076
Signed-off-by: John Grossman <johngro@google.com>
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Allow AudioSink to use deep audio buffering when the
source is audio only and its duration is more than
a certain threshold.
This helps improve battery life but implies higher
audio latency.
Change-Id: Ie79915b61c370292f05aabda9779356570e03cbb
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This makes NuPlayer use a SkipCutBuffer when needed, and adds a new
AudioSink method to retrieve the number of frames written so far, so
NuPlayerRenderer can calculate how much data it can write without blocking.
Also make some more methods const.
Change-Id: Id7d253ad8a7b85e9a84ca2baafbe32817b16c744
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Change-Id: Ib3982a9c960bfdb0cb7e1b174440b141b194cfbe
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Currently able to play Ogg Vorbis, PCM WAV and other lossless files seamlessly
by reusing the initial AudioTrack for subsequent players.
Change-Id: Ie7cf6b9076bdf4f9211574456d192c02c04fecc7
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The AudioSink latency is currently cached when the associated AudioTrack
is created. However, the AudioTrack latency can change if the AudioTrack is moved
from one output stream to another.
The AudioPlayer must also periodically update its view of the latency
as it is needed to compensate the real audio time used for A/V sync.
This fixes an A/V sync problem seen when switching A2DP on and off while
playing a video.
Change-Id: I28b24049ca114e1af3e24791dcc900f463536ba4
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Add support for specifying a channel mask when opening an AudioSink.
This parameter does not replace the channel count parameter in order
to not have to duplicate the logic to derive a mask from the
channel count everywhere an AudioSink is used without a known mask.
A mask of 0 (CHANNEL_MASK_USE_CHANNEL_ORDER) means a mask will
be automatically derived from the number of channels.
Update existing AudioSink implementations to use the channel mask,
and users of AudioSink to specify the mask if available, and
CHANNEL_MASK_USE_CHANNEL_ORDER otherwise.
Change-Id: Ifa9bd259874816dbc25ead2b03ea52e873cff474
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This is a cherry-pick of I6ab07d89b2eeb0650e634b8c3b7a0b36aba4e7dd
with merge conflicts addressed by hand and additional changes made in
response to code review feedback.
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: I3b46c5227bbf69acb2f3cc4f93cfccad9777be98
Signed-off-by: John Grossman <johngro@google.com>
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Add support for modifying the playback rate of a MediaPlayer
by altering the sample rate of its AudioTrack.
The playback rate is expressed in permille, where 1000 is the
playback at normal speed.
Change-Id: I981d060ab32f7bae7a767e82c60c88ae635dceed
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Remove unnecessary includes of AudioTrack.h.
Use forward declaration of class names in preference to #include when possible.
Change-Id: I12982811fa75c2c7695d8bbfa595a7aaec047dc0
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At native level it was a mixture of audio_stream_type_t, int, uint32_t,
and uint8_t. Java is still int. Also fixed a couple of hard-coded -1
instead of AUDIO_STREAM_DEFAULT, and in startToneCommand a hard-coded 0
instead of AUDIO_STREAM_VOICE_CALL.
Change-Id: Ia33bfd70edca8c2daec9052984b369cd8eee2a83
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Was int, uint32_t, uint16_t, and uint8_t with 2-bit bitfield.
Also replace 0 by AUDIO_FORMAT_DEFAULT and replace 1 by
AUDIO_FORMAT_PCM_16_BIT.
Change-Id: Ia8804f53f1725669e368857d5bb2044917e17975
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All surfaces are now supported through surface textures.
Change-Id: I95dd823e7099c0c32a48a1121624149dcc29d9c6
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This change moves the ANativeWindow connect and disconnect logic from
MediaPlayer to MediaPlayerService::Client.
Bug: 5502654
Change-Id: Ifc43b98b01ad8f35d62d7ece43110724ec7fda3d
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Bug #1870981
Change-Id: Ia3ad166390c4d60cea19c3783895b078a2c4c15f
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Change-Id: I2bcb54b8232afd3fc7ee16289f37c7a7b3f23067
related-to-bug: 4517282
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This reverts commit 2225e4b7049fa3fb9d39a068b8268b63c952d7c1
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Change-Id: Ifefc708d46874e04fd0d01cb6e2d43b987ee796c
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Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
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for bug 1982947
Change-Id: If3f40e4f18cbba155af29944af38bdc627f8cd53
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Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
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through listener during video playback.
- Add OnTimedTextListener in the MediaPlayer
For feature request 800939.
Change-Id: I65072c27acb4c0037109a72be38c73e9f667420f
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- To track the currently used audio device
- The devices are separated as speaker and other audio devices
- Provide the collected data to battery application through pullBatteryData()
Change-Id: I374c755266b5ac6b1c6c630400f4daf901ea8acc
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This change enables the use of a SurfaceTexture in place of a Surface
as the video sink for an android.media.MediaPlayer. The new API
MediaPlayer.setTexture is currently hidden.
This includes:
- New Java and C++ interfaces
- C++ plumbing and implementation (JNI, Binder)
- Stagefright AwesomePlayer and NuPlayer use ANativeWindow
(either Surface or SurfaceTextureClient)
Change-Id: I2b568bee143d9eaf3dfc6cc4533c1bebbd5afc51
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codec usage (duration) for the battery app
- Collect MediaPlayer decoding usage data
Change-Id: I0ef4e32b6a041ba1fe73c19f9c67185c61d03965
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Change-Id: Ifbac61406dcb81343765f99ccba08bd90f9274cc
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Change-Id: I17b8e0dbf53fca37c96830c41131b4bc0c24ca6d
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commit 0d5694ba2d399dd0869532a4d6256448185a1be0
Author: Andreas Huber <andih@google.com>
Date: Fri Oct 29 11:59:23 2010 -0700
suspend() and resume() methods on VideoView are back but don't do anything.
They need to be back because they were public before.
Change-Id: Iddfd1021ffcf341f26e8d55ba761fd33701e2425
commit 16192891ed7d349ee97e47d1729d20a2d0d247b8
Author: Andreas Huber <andih@google.com>
Date: Fri Oct 29 11:47:05 2010 -0700
Revert "New API on VideoView widget to suspend/resume a session. Do not release the MediaPlayer client for video suspending/resuming."
This reverts commit 2e1818a4d16c3309660f27286c77d8d1eee95a25.
Conflicts:
api/current.xml
Change-Id: I68dd1d05871044faf3f832d0838aa40bc7f890e5
commit 8f934dc1a3ae4e60f0790fcf97671e063fa20fad
Author: Andreas Huber <andih@google.com>
Date: Fri Oct 29 11:44:16 2010 -0700
Revert "Release mediaplayer if the current state is not suspending. Fix for bug 2480093."
This reverts commit efb882cf75eef39ecaf9f8920ed302a019fa629f.
commit f2ed03550887986f39d36b5dabcd9e919949c7cf
Author: Andreas Huber <andih@google.com>
Date: Fri Oct 29 11:44:08 2010 -0700
Revert "Release MediaPlayer if suspend() returns false."
This reverts commit 047212fd4ea360675e94d3ce83c7f5544f65b268.
commit 441ecce678bd24e9660a72c8627b5bd94433ff8b
Author: Andreas Huber <andih@google.com>
Date: Fri Oct 29 11:40:46 2010 -0700
manually.
Change-Id: I4fdd43c9f7c8b3eedddb31a196da4984e1c58e87
Change-Id: I60d4b10e7a9e4ed8d9a796f1711618f557eb6e89
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into gingerbread
Merge commit '4f21e517d09b9d793d20d64547df330fba705b3c'
* commit '4f21e517d09b9d793d20d64547df330fba705b3c':
Added getter for session Id to AudioSink
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Added a method to expose the audio session id at AudioSink interface
so that the AudioPlayer in stagefright can retrieve it.
Also:
- Fixed audio effect send level not being initialized in mediaplayer.
- Fixed compilation error when LOGV is enabled in mediaplayer JNI
Change-Id: I4bb55454fd63d646e0e677692d737c4843fb05fb
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commit 35cc68814a9537c31fde146e171e7b0bbdfe211e
Author: Andreas Huber <andih@google.com>
Date: Mon Aug 16 08:48:42 2010 -0700
Only enable support for yuv to yuv conversion on passion, where it's available, use the slower yuv->rgb565 path everywhere else.
commit d8ac5a8814103e60d11d2acf61997fc31a1dc58d
Author: Andreas Huber <andih@google.com>
Date: Fri Aug 13 13:56:44 2010 -0700
The software renderer takes over all rendering, converting from yuv to yuv if possible and rgb565 otherwise.
commit 684972074b74318bdcb826ed9b5b0864d2d2e273
Author: Andreas Huber <andih@google.com>
Date: Fri Aug 13 09:34:35 2010 -0700
A first shot at supporting the new rendering APIs.
Change-Id: Iea9b32856da46950501f1a700f616b5feac710fd
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Added methods to AudioTrack and MediaPlayer java classes to enable use of
auxiliary audio effects. The effect can be attached and detached by specifying its
ID and the send level controlled.
Change-Id: Ie74ff54a453096a742688476f612ce355543b6f3
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