| Commit message (Collapse) | Author | Age | Files | Lines |
... | |
|
|
|
| |
Change-Id: I27c46bd1d1b2b5f96b87af7d05b951fef18a1312
|
|
|
|
|
|
|
|
|
|
| |
When creating a new AudioTrack (not inheriting one from a previous play),
the AudioSink should take the AudioTrack's position as the initial starting
point for mBytesWritten, since otherwise NuPlayer's calculations will be off.
Normally this position will be 0, but if the test code for 32 bit wraparound
in AudioFlinger.cpp is enabled, it might be (much) larger.
Change-Id: I1e4f906d529861c3dea996de8afc6dbd491589af
|
|
|
|
|
| |
Change-Id: I69ed31e7a8b4d69d1209d2d516f94d258f072566
related-to-bug: 6275919
|
|
|
|
|
|
|
|
|
| |
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
|
|
|
|
|
|
|
| |
o plus a few file relocation: ActivityManager.cpp/h, SoundPool.h, etc
o remove some runtime dependencies to libandroid, libandroid_runtime, etc
Change-Id: I047a47c5fb361dd5cf85cd98798c39f629a75d10
|
|
|
|
| |
Change-Id: Ib3982a9c960bfdb0cb7e1b174440b141b194cfbe
|
|
|
|
|
|
| |
if media.stagefright.use-nuplayer is set to true.
Change-Id: Ibb217e7d7d5195b7feeea557554fe78e1585744c
|
|
|
|
| |
Change-Id: I327663a020670d0a72ff57bd0b682e2ce0528650
|
|
|
|
|
|
|
| |
and avoid ambiguous term "channels" where it might be confusing
as to whether it is a channel mask or channel count
Change-Id: I744fa08ccb6001a98c97bd638d2c9d56836c4234
|
|\ |
|
| |
| |
| |
| |
| |
| |
| | |
Currently able to play Ogg Vorbis, PCM WAV and other lossless files seamlessly
by reusing the initial AudioTrack for subsequent players.
Change-Id: Ie7cf6b9076bdf4f9211574456d192c02c04fecc7
|
|/
|
|
|
|
|
|
|
|
| |
This affects:
- IAudioFlinger::openOutput
- AudioTrack::AudioTrack
- AudioTrack::set
- apps that call these
Change-Id: I26fb281bac6cb87593d17697bc9cb37a835af205
|
|\ |
|
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| |
| | |
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
|
|/
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
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
|
| |
| |
| |
| |
| |
| |
| | |
See https://android-git.corp.google.com/g/#/c/157220
Bug: 5449033
Change-Id: Ic9c19d30693bd56755f55906127cd6bd7126096c
|
|/
|
|
|
|
|
| |
See https://android-git.corp.google.com/g/157065
Bug: 5449033
Change-Id: I00a4b904f9449e6f93b7fd35eac28640d7929e69
|
|
|
|
|
|
|
| |
See https://android-git.corp.google.com/g/156016
Bug: 5449033
Change-Id: I4c4e33bb9df3e39e11cd985e193e6fbab4635298
|
|\
| |
| |
| | |
Change-Id: Ib1536b1a4c9eeff80e0726b3e61cee12057cd120
|
| |
| |
| |
| |
| |
| | |
All surfaces are now supported through surface textures.
Change-Id: I95dd823e7099c0c32a48a1121624149dcc29d9c6
|
|\ \
| |/
| |
| |
| |
| |
| | |
MediaPlayerService" into ics-mr0
* commit 'fc9592f8a5f2f75207e5e532655ac294eb2b334b':
Stagefright: ANW::connect in MediaPlayerService
|
| |\
| | |
| | |
| | |
| | |
| | |
| | | |
ics-mr0
* commit '08479ceeba56c460fb52f60a24df27776f1936c3':
Stagefright: ANW::connect in MediaPlayerService
|
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | | |
This change moves the ANativeWindow connect and disconnect logic from
MediaPlayer to MediaPlayerService::Client.
Bug: 5502654
Change-Id: Ifc43b98b01ad8f35d62d7ece43110724ec7fda3d
|
|/ /
| |
| |
| |
| |
| |
| | |
See https://android-git.corp.google.com/g/#/c/143865
Bug: 5449033
Change-Id: I0122812ed6ff6f5b59fe4a43ab8bff0577adde0a
|
|\ \
| |/
| |
| |
| |
| |
| | |
dumping." into ics-mr0
* commit 'b3cdadb639027f62c7c1637ca962a70d2d1f3b4d':
Check whether media recorder client exists before dumping.
|
| |
| |
| |
| |
| | |
Change-Id: I1f3a644a958975e4cf6c02099c53e30cc4d2fd82
related-to-bug: 5477177
|
|/
|
|
| |
Change-Id: Ie204db8810807f1e7981959e34dc0149e5d9563a
|
|
|
|
|
|
|
|
| |
When decoding a file for the SoundPool, do not
reject the entire file in case of error but
return what was decoded so far instead.
Change-Id: Iff199a1b6a4c8e064e42a0dfe0704e0ae36a27fd
|
|
|
|
|
|
| |
Bug #1870981
Change-Id: Ia3ad166390c4d60cea19c3783895b078a2c4c15f
|
|
|
|
| |
Change-Id: Id56bd0c16104e46d8dc71f13d8a44aefe251fad4
|
|
|
|
|
|
|
| |
- Atribute network activity to uid calling the mediaplayer
- Enables logging of chromium network stack in logcat
Change-Id: I2d28c8392248a056b3cee305dd4d4475ebba4337
|
|
|
|
|
| |
Change-Id: I2bcb54b8232afd3fc7ee16289f37c7a7b3f23067
related-to-bug: 4517282
|
|
|
|
| |
Change-Id: I12ba7d542331a8293d67a0d47378b8be4f777759
|
|
|
|
|
|
| |
The gettid system call is always available now.
Change-Id: Ib78b41781eda182dc8605daf456bbea7ff7c2dc0
|
|
|
| |
This reverts commit 2225e4b7049fa3fb9d39a068b8268b63c952d7c1
|
|
|
|
| |
Change-Id: Ifefc708d46874e04fd0d01cb6e2d43b987ee796c
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| |
Record and playback objects (resp AudioRecord and AudioTrack)
are created using a channel mask, but this information is lost
in the mixer because only the channel count is known to
AudioFlinger. A channel count can always be derived from a
channel mask.
The change consists in:
- disambiguiting variable names for channel masks and counts
- passing the mask information from the client to AudioFlinger
and the mixer.
- when using the DIRECT ouput, only verifying the format of
the track is compatible with the output's for PCM.
Change-Id: I50d87bfb7d7afcabdf5f12d4ab75ef3a54132c0e
|
|
|
|
|
| |
Change-Id: Ic4c62c4037800802427eb7d3c7f5eb8b25d18876
Signed-off-by: Dima Zavin <dima@android.com>
|
|
|
|
|
|
| |
for bug 1982947
Change-Id: If3f40e4f18cbba155af29944af38bdc627f8cd53
|
|
|
|
|
| |
Change-Id: Ibc637918637329e4f2b62f4ac7781102fbc269f5
Signed-off-by: Dima Zavin <dima@android.com>
|
|
|
|
|
|
|
|
| |
through listener during video playback.
- Add OnTimedTextListener in the MediaPlayer
For feature request 800939.
Change-Id: I65072c27acb4c0037109a72be38c73e9f667420f
|
|
|
|
|
|
|
|
|
|
| |
Instead of returning 0-filled buffers after EOS from AudioTrack we do this
work in AudioOutput instead. That way the EOS signal (0 frames returned)
is preserved in AudioCache which otherwise would lead to a heap size overflow
filling everything with zeroes.
Change-Id: I7e07429ba887957a4340dd4b21eef4bba76248cd
related-to-bug: 3514073
|
|
|
|
|
| |
Change-Id: I2284e1d62babde7f739fba6a3cb4e2619f0e62f9
related-to-bug: 4148291
|
|
|
|
|
|
| |
bug - 4099038
Change-Id: I6c048eaf3d7f34bc144b8daaa5fdef1ed474af66
|