| Commit message (Collapse) | Author | Age | Files | Lines |
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Change-Id: I9bc5573ee07e30b305b5b879023aa9ec69e10b91
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Change-Id: Ib3982a9c960bfdb0cb7e1b174440b141b194cfbe
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o related-to-bug: 6214141
Change-Id: Ic88d1732b3e014af47532a0809e01f6086e8464d
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if media.stagefright.use-nuplayer is set to true.
Change-Id: Ibb217e7d7d5195b7feeea557554fe78e1585744c
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Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
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Change-Id: Ib8cff8abd73723b793f08da99ad59549f219e0e7
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and avoid ambiguous term "channels" where it might be confusing
as to whether it is a channel mask or channel count
Change-Id: I744fa08ccb6001a98c97bd638d2c9d56836c4234
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tracks.
o Newly introduced APIs are (MediaPlayer):
getTrackInfo() / addExternalSource() / enableTrack() / disableTrack().
o Timed text tracks are supported only, for now.
o TODOs:
- Define the audio/video behavior for enableTrack and disableTrack.
- Refactoring AwesomePlayer / TimedTextDriver so that all types of
track index can be managed in the correct order and be ready for
supporting audio/video tracks.
(MediaExtractor and MediaSource for text file might be necessary.)
Change-Id: Idb85e1b3f2ed49a64f377d05472dd6663ce94e07
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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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and deleted the duplicate header files in /frameworks/base
o related-to-bug: 6044887
Change-Id: I17e0692d9a9b5c8796ded36677c833ca8ab36795
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This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
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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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Change-Id: Iec71706cdd4f29c6904993648ce873e83ef9cafe
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ABuffer objects through messages.
Change-Id: I9f8b4e4c4767d0d70a0105e0c0813b754379b49d
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Change-Id: I13e54062d4de584355c5d82bb027a68aeaf2923b
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o also added a check on whether capture rate was set before starting time lapse video recording.
o related-to-bug: 6045507
Change-Id: I8e1fdc8e8931e2684ab3822dc6260db44658e87d
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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>
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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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Change-Id: I963a3b6f79a7292891973cbeeaf3378b38629f08
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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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See https://android-git.corp.google.com/g/157519
Bug: 5449033
Change-Id: I8ceb2dba1b031a0fd68d15d146960d9ced62bbf3
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See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
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See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
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See https://android-git.corp.google.com/g/156801
Bug: 5449033
Change-Id: Ib08fe86d23db91ee153e9f91a99a35c42b9208ea
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See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
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Change-Id: I30f4e5d35d1b24dce2e6152a13fd0fb61414a3e2
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Change-Id: I14104540cfbd2bcdd7af1534db20554b88c4c0ce
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queue them for 200ms" into ics-mr1
* commit 'ebfce84f7fd3e25fa08c36b155db91df90a59b3d':
Don't perform RTSP seeks right away but queue them for 200ms
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ics-mr1
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and only execute the last one.
Change-Id: I9ab342396ec9c9c03624a4b0306d1e180ceca000
related-to-bug: 5732960
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make sure we emulate timestamps" into ics-mr1
* commit 'aa82c39bdb4ad9c1fdcb09f3bea11be5197d3ce6':
Fix Bitreader "putBits" implementation, make sure we emulate timestamps
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if we don't receive npt time mapping from the rtsp server (i.e. live stream)
Change-Id: I5147d665bd90c9a303ad6ffdafbf770f930f917c
related-to-bug: 5660357
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Change-Id: I9eb7b7c1668b5f0de85ec8b1e9257cf1ec4d8b9e
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a) one of the two decoders has a pending discontinuity
b) the renderer holds on to all output buffers for that decoder
c) the renderer is paused
if all three conditions are met the decoder won't ask for more input data
and therefore never see the discontinuity.
To avoid this we briefly resume the renderer just before shutting down.
Change-Id: I9e08af2a1eb4298d1cd00497d6aa33f4ad184e9a
related-to-bug: 5655016
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signal discontinuities" into ics-mr1
* commit '074133b260f8d12e484e93b9aa5a33a4983b7349':
Updated (internal) API for IStreamSource to signal discontinuities
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into ics-mr1
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Change-Id: Idd4b9d8e7cec16b3e3c91c70e75144d42be30f96
related-to-bug: 5553055
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Change-Id: Id5cfbcf20e5ec28782cc88075040cf9edebfdfe8
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