Merge commit '2b4f1f4cb814f7a7df8d2cb9fcb5210bfe7999c7'
* commit '2b4f1f4cb814f7a7df8d2cb9fcb5210bfe7999c7':
Disable 10secs forward/backward seeking for rtsp as seek is a very expensive operation there. Decouple the 10sec forward/backward button functionality from seekbar functionality.
Merge commit 'bb70837397e3fb437b7b4443b37d7a83c11e6e43'
* commit 'bb70837397e3fb437b7b4443b37d7a83c11e6e43':
Work to support switching transport streams mid-stream and signalling discontinuities to the decoder.
Merge commit '45bd1159fa34b51ba077e0cde760d171ca092552'
* commit '45bd1159fa34b51ba077e0cde760d171ca092552':
On this particular device the hardware video decoder spits out buffers that don't actually contain our video data, so we cannot use them to restore the video frame after suspend/resume.
Part III: Move startRecording() call earlier, asking camera hal
to allocate video buffers before CameraSource.start() is called.
Change-Id: I3f1d7d5636ca2644fe52af61f297d48c6b1ce89d
o updated comments and streamlined the logic in
checkVideoSize() and checkFrameRate() as suggested
Change-Id: I49d04ac7998d4a215997aa63555dfb6e814e38d3
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
Merge commit '56ee1080f004110bff622e5b60c243d9cabfe120'
* commit '56ee1080f004110bff622e5b60c243d9cabfe120':
Make sure to call AudioTrack::stop() instead of AudioTrack::pause() after submitting all samples to AudioTrack to make sure those remaining samples are actually played out.
Merge commit 'a86a6c4e326bfdfc351dacca95b23bb78f78efbe'
* commit 'a86a6c4e326bfdfc351dacca95b23bb78f78efbe':
Fixed an issue where the reserved free space in the file writer was larger than intended
The problem was that even though user does not explicitly request the max file size
limit via MediaRecorder.setMaxFileSize(), the file writer sets an implicit file
size limit if 32-bit file offset is used on user's behalf. The reserved free space
is estimated based on the file size, if the file size limit is set by the user.
The fix is to add an extra bool to tell the difference between an
explit requested file size and an implicit file limit and use that
to set the estimated moov box size accordingly.
Change-Id: I731aca6c7833aa764ed7b905edb77721577471b3
Merge commit 'd6c30e8c1521bc584f33500b8ee897dafdfec023'
* commit 'd6c30e8c1521bc584f33500b8ee897dafdfec023':
Instead of constantly polling the AudioPlayer to see if it reached EOS or finished seeking, initiate the notification from the AudioPlayer when the event happens.
Merge commit 'c889bbfa965f4ba90636f561c5e1353289d4cb06'
* commit 'c889bbfa965f4ba90636f561c5e1353289d4cb06':
Vorbis files may have more samples encoded that should be used, i.e. we have to trim samples at the end of the stream. This is crucial for proper looping of some audio files.
commit 29a4d3effb05a2e074cb0693316ab1977baeb0b6
Author: Andreas Huber <andih@google.com>
Date: Mon Sep 27 12:01:32 2010 -0700
Fully working implementation of MPEG2TSWriter (for AAC and AVC sources).
Change-Id: I8a32a47565b647bf6c078c520e39565e08ea0d84
commit f4dec4c3899f3be393508e180d6c07e249d3335e
Author: Andreas Huber <andih@google.com>
Date: Mon Sep 27 10:36:31 2010 -0700
More reliable identification of MPEG2 transport streams. Don't keep scanning forever in case the stream does not have both audio and video tracks.
Change-Id: Icc5b4e8be145b2805e8776559546a6818342aea7
commit 4fe3cc942f9b3d3cf54138b828c41214aa916dd2
Author: Andreas Huber <andih@google.com>
Date: Mon Sep 27 08:23:39 2010 -0700
test code
Change-Id: I16560a17661407d06497f99ff88230724bb898af
commit 64d988b24f49f179a90fa677be11c823959e734b
Author: Andreas Huber <andih@google.com>
Date: Thu Sep 23 14:42:52 2010 -0700
First shot at supporting writing to an MPEG2 transport stream.
Change-Id: Ie537939a99fa3ddc0c7661c47c18277584817c74
Change-Id: If78fd034af8f6e8ceac8dbeff96d5ecb3f6b96dc
The low and high profiles should each match one of the
specific profiles. So we need to add the specific profiles
corresponding to the low/high profiles. This makes the
default profile compliant to documentation + cts.
Also fixed javadoc to account time lapse profiles.
Change-Id: I34e7307d00ce261c69dc10ead2900025c7f6d428
If the frame capture interval is large, read will block for a long time.
Due to the way the mediaRecorder framework works, a stop() call from
mediaRecorder waits until the read returns, causing a long wait for
stop() to return. To avoid this, we return a copy of the last read
frame with the same time stamp if a frame is not available quickly.
This keeps the read() call from blocking too long. This method is
triggered when startQuickReadReturns() is called on
CameraSourceTimeLapse.
In the still camera case, also using waitRelative on Condition
instaed of sleeping, so that we can wake it up.
Also for the idle check instead of sleeping, we now wait on a
condition variable, which is woken up when the last takePicture
callback gets called.
Change-Id: Ia74386e175536aee0f44ae2f8b114c353d3d72f5
The native media scanner no longer filters files based on file extension.
Audio, video, image and playlist files are handled as before, but non-media
files are now inserted into the "files" table, which was originally added
to support MTP.
Change-Id: I9053218fb6d2671a3bb181405c34442b94678afc
Signed-off-by: Mike Lockwood <lockwood@android.com>
- Exposing the specific resolution profile levels
QUALITY_{QCIF,CIF,480P,720P,1080P} and the new time lapse profiles
QUALITY_TIME_LAPSE_{LOW,HIGH,QCIF,CIF,480P,720P,1080P}
- Unhiding the hasProfile() function used to test if a given profile exists.
Change-Id: I5d8b9e1ba61718f304235e76d85244e428e68643
- Added hasProfile to CamcorderProfile and JNI.
- Added hasCamcorderProfile to MediaProfiles.
- using android.hardware.Camera.CAMERA_ID_DEFAULT for default camera
in get().
Change-Id: Ib57bb49ae79492d7cbc0ec6c7b6efcbf74f80013
- Added enums QUALITY_{QCIF,480P,720P,1080P}
QUALITY_TIME_LAPSE_{LOW,HIGH,QCIF,480P,720P,1080P} in CamcorderProfile
and corresponding ones in MediaProfiles.
- Added functions createDefaultCamcorderTimeLapseLowProfile,
createDefaultCamcorderTimeLapseHighProfile to set default values.
- Moved javadoc for constants to the get() function.
Change-Id: Ib8b3f8d29395dff77a397d1e6b44cfaf8c481d4d
Merge commit 'e126119c3a406bc564f2549aeb1416aff112689d'
* commit 'e126119c3a406bc564f2549aeb1416aff112689d':
Modify type of some environmental reverb parameters
Changed type of decay time, reverb delay and reflections delay parameters
from signed to unsigned int to match OpenSL ES interface definition.
Also fixed some type casts in lvm reverb wrapper.
Change-Id: I5ca5e76a87c2590f01f031f3168355586ef22556
CameraSourceTimeLapse now decides whether to use still or video
camera automatically. It checks if the passed in size is a valid
preview size and if it is, then uses the video camera else uses
the still camera.
Removed from StagefrightRecorder the support to set parameter
useStillCameraForTimeLapse.
Change-Id: I71f5b0fc7080ca524792381efe918d22e41a7f36
Merge commit '8e11c82247151085fa165c76bfbc157bc6091ca4'
* commit '8e11c82247151085fa165c76bfbc157bc6091ca4':
Ogg files can be tagged to be automatically looping, this setting always overrides the MediaPlayer's setLooping setting.
This change defines the two OMX_SetParameter calls that enable OMX codecs to
interact with ANativeWindows. It also adds the plumbing to the IOMX, OMX, and
OMXNodeInstance classes to use these new APIs.
This is try 2 for this change, after reverting the first one because it broke
the build.
Change-Id: I94249b72bdb5d5719360f03d7935fcca4ece5028
When the recorded file becomes large, the metadata size can
no longer be ignored. This makes it possible to save the
recorded file when the storage becomes almost full at the
end of the recording session.
Change-Id: Ief038080f825c9946ce550949c03e914aec1e31a
Merge commit 'bb64e554d9a28fcf8eebf579e91ff71b8ffef1e3'
* commit 'bb64e554d9a28fcf8eebf579e91ff71b8ffef1e3':
Calculate audio media drift time from AudioSource
