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
Merge commit '7ed7668b30e70ca8e3f0f183364433326ed29f39' into gingerbread-plus-aosp
* commit '7ed7668b30e70ca8e3f0f183364433326ed29f39':
Calculate audio media drift time from AudioSource
Added support for the "device friendly name" and "synchonization partner"
properties, which are required by Microsoft.
Change-Id: Ic0443333d75f7d98a2d902a790b9d505a56d4eef
Signed-off-by: Mike Lockwood <lockwood@android.com>
The problem was that the time to receive an output buffer
from an audio encoder is different because the encoder does not
need to read from the source for all output buffers. This leads
to large fluctuation in terms of wall clock duration between two
neighboring audio sample outputs from the audio encoder. As a
result, the media time for the video track after adjustment using
the drifting changes wildly sometimes.
This patch addresses this issue by only updating the media drift
time when an audio source input buffer is read. the wall clock
for the audio track is also calculated at the same time when
the input audio buffer is read at AudioSource.
bug - 2959800
Change-Id: I3174aa182f744784b540f0a7198524d4eee8bd7b
Merge commit 'fd0eed007d99178092ede56ec2c4799046615f70'
* commit 'fd0eed007d99178092ede56ec2c4799046615f70':
Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data.
Merge commit '3fd01c4da9b8fb7796d64096b9bbd6fcdee280e6'
* commit '3fd01c4da9b8fb7796d64096b9bbd6fcdee280e6':
Make sure that if initialization fails, AudioSource still behaves well.
Merge commit 'a2511da9d65b11be7f59ed3f525f77e85aeb4bef' into gingerbread-plus-aosp
* commit 'a2511da9d65b11be7f59ed3f525f77e85aeb4bef':
Better support for buffered streaming of rtsp content, if buffer drops below a certain threshold we will temporarily pause playback until we have sufficient data.
Merge commit 'd3c1bae4eb78404bd1e17b7acf67087a18c83ef3' into gingerbread-plus-aosp
* commit 'd3c1bae4eb78404bd1e17b7acf67087a18c83ef3':
Make sure that if initialization fails, AudioSource still behaves well.
Merge commit '6c33904ad948cb64245fbc5950c839e4d9e56de3' into gingerbread-plus-aosp
* commit '6c33904ad948cb64245fbc5950c839e4d9e56de3':
Now that AmrInputStream no longer relies on opencore, make sure it's registered in non-opencore builds.
Added support for passing width, height and video bitrate
for the auxiliary video.
Also setting encoder level depending on the video size and bitrate.
Change-Id: I4a90046853f67287c3e7e6babc75b4827f0c3e73
Merge commit '412fc7cdb6a1c4b6afe85b58fcc794fd67271942' into gingerbread-plus-aosp
* commit '412fc7cdb6a1c4b6afe85b58fcc794fd67271942':
Keep gtalk video chat specific code consistent with rtsp changes.
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.
Change-Id: Ibfbf893dc3513db0b3d3221bec5708c77287cddc
Merge commit 'de2b1615d27881d98f483fc9158497fbe1fc5f8d' into gingerbread-plus-aosp
* commit 'de2b1615d27881d98f483fc9158497fbe1fc5f8d':
Properly extract all raw_data_blocks from an ADSP mpeg4 audio buffer.
- Fixed click when re-enabling effect during the turn off phase:
make sure the effect states where effect is processed are the same
where volume control is delegated to effect.
- Fixed click when effect is deleted while still active: do not apply
volume ramp if an effect having volume control was just removed from the
effect chain.
Also fixed a crash when PCM dump is enabled in effect bundle wrapper.
Change-Id: Ib562f5cf75c69af75df0e862536262e2514493e4
Merge commit '48ac68e1b117b6b55f06daced7d9d5d550853306' into gingerbread-plus-aosp
* commit '48ac68e1b117b6b55f06daced7d9d5d550853306':
Support for RFC3640 - mpeg4-generic RTP packet type, AAC-lbr and AAC-hbr.
Merge commit 'e1a3cddd94749a42457a8f32cf21f663f07e4edf'
* commit 'e1a3cddd94749a42457a8f32cf21f663f07e4edf':
Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)
Merge commit '99fa510e67cb973b45fc216c75bdc817421e14ae' into gingerbread-plus-aosp
* commit '99fa510e67cb973b45fc216c75bdc817421e14ae':
Better detection of connection problems - timeout if no rtcp packets arrive within a certain time, not a final frame (which may take longer)
This can be used as a compatibility workaround for host operating systems
without MTP support.
Change-Id: If4f1856206056ca8e40c3ffbfa382f185c413598
Signed-off-by: Mike Lockwood <lockwood@android.com>
