include/camera -> camera/include/camera
include/media/audiohal -> media/libaudiohal/include
include/media/AudioResampler*.h -> media/libaudioprocessing/include
include/media/Audio*.h,IAudio*.h,IEffect*.h,ToneGenerator.h -> media/libaudioclient/include
include/media/EffectsFactoryApi.h -> media/libeffects/include
include/media/stagefright -> media/libstagefright/include
include/media/nbaio -> media/libnbaio/include
include/media/<rest of files> -> media/libmedia/include
include/cpustats -> media/libcpustats/include/cpustats
Added symlinks from old location to new ones
Bug: 33241851
Test: VNDK linked modules will need to add explicit lib dep.
All other modules should compile the same
Change-Id: I0ecf754a2132640ae781a3cc31428fb8c0bd1669
The VolumeShaper is used to apply a volume
envelope to an AudioTrack or a MediaPlayer.
Test: CTS
Bug: 30920125
Bug: 31015569
Change-Id: I42e2f13bd6879299dc780e60d143c2d465483a44
Property media.stagefright.audio.sink (in milliseconds)
Also change the default buffer size for PCM playback to 500 ms.
Bug: 21198655
Change-Id: I5781288f59bf08fbecd9263a26c919570b58be0f
This allows apps to implement MediaDataSource, which is modeled on
stagefright's DataSource, to supply media data to the framework. This
was already implemented for MediaExtractor, but it was renamed from
DataSource.
MediaExtractor, MediaPlayer and MediaMetadataRetriever each have a new
overload: #setDataSource(android.media.MediaDataSource)
Only NuPlayer supports this new data source.
The change introduces:
* IDataSource: The binder interface for DataSource.
* JMediaDataSource: The native counterpart to the java interface. It
implements IDataSource.
* CallbackDataSource: A stagefright DataSource that wraps an
IDataSource.
Change-Id: Ib3c944b49cc8a792c8eb9c85e5015c07f298ebc1
This gets rids of a bunch of special midi handling and replaces it
with an extractor that works with NuPlayer and MediaMetadataRetriever.
Change-Id: I8d0f5bbdde2ca24267cf4d62ab26afe9630e0217
since we started to use java's HTTPConnection instead of the native
implementation. Also remove other remnants of the previous http implementation,
such as accounting for the http user's uid.
Change-Id: I60bfd31381ea40d2220db587ec5c433093b60034
AudioPlayer must read the sampling rate from offloaded audio sinks
whenever a new time position is computed as the decoder can update
the sampling rate on the fly.
Change-Id: I997e5248cfd4017aeceb4e11689324ded2a5bc88
canOffloadStream() function in stagefright utils forces the
stream type to AUDIO_STREAM_MUSIC when querying the audio policy
manager if a particular track is offloadable or not.
This causes MP3 ringtones to be offloaded which is not a validated use case.
The fix consists in using the actual stream type read from the AudioSink.
Bug: 11410937.
Change-Id: I44b8e033a8e785a79cdc291b142f80b5580bdc4d
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>
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>
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
Relocate the AAH RTP code from framework/av into
vendor/google_devices/phantasm. This change is the deletion, there
will be a separate CL which re-introduces on the vendor side of
things.
Change-Id: Ibe7e6d4b633a3886b87a615691a2692f2382af6c
Signed-off-by: John Grossman <johngro@google.com>
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>
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
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
Currently able to play Ogg Vorbis, PCM WAV and other lossless files seamlessly
by reusing the initial AudioTrack for subsequent players.
Change-Id: Ie7cf6b9076bdf4f9211574456d192c02c04fecc7
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
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>
Upintegrate the android at home TX and RX players developed in the
ICS_AAH branch.
Change-Id: I8247d3702e30d8b0e215b31a92675d8ab28dccbb
Signed-off-by: John Grossman <johngro@google.com>
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
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
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
through listener during video playback.
- Add OnTimedTextListener in the MediaPlayer
For feature request 800939.
Change-Id: I65072c27acb4c0037109a72be38c73e9f667420f
Vijay Venkatraman
Hassan Shojania
Andy Hung
Wei Jia
Glenn Kasten
Ronghua Wu
Lajos Molnar
Chris Watkins
Marco Nelissen
Chong Zhang
Andreas Huber
Eric Laurent
Richard Fitzgerald
Andy McFadden
John Grossman
Jean-Michel Trivi
Gloria Wang
Dima Zavin