Merge commit '3540760d1d68cc883122d44ab1d38f542fb646e6'
* commit '3540760d1d68cc883122d44ab1d38f542fb646e6':
Don't drop a late frame which may lead to missing I frames in the MP4 file
Merge commit '177a7ad825445acaeea38c48c74ad87db935d054'
* commit '177a7ad825445acaeea38c48c74ad87db935d054':
Return error from MPEG4Writer stop() if the check on codec specific data failed
Merge commit 'c8d2fa704abebdbf0bd8aac185216dc068950217'
* commit 'c8d2fa704abebdbf0bd8aac185216dc068950217':
Make MediaWriter stop and pause return errors if necessary
Merge commit '873ebfb825cb498d9ff3012d1d31b02e31a79980'
* commit '873ebfb825cb498d9ff3012d1d31b02e31a79980':
Support for MP4V-ES packetization format according to RFC3016.
Merge commit 'b29ebd397e25a7176bcc1c81980f17b0190ebe7f'
* commit 'b29ebd397e25a7176bcc1c81980f17b0190ebe7f':
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
Merge commit '6bcffcd2dc410db780c152c70a01b22da6ca58be'
* commit '6bcffcd2dc410db780c152c70a01b22da6ca58be':
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
Added a downsample function which downsamples the source image
starting at an offset and skipping every few pixels. Currently
no low pass filtering is done, but it should be added later.
Change-Id: Iec34092c536bfc661a15521e6a1ef2ef3f815c61
- made width(), height() const member functions.
- added validPixel() which returns true if pixel is in the allowed range.
- now testing validPixel in get/setPixelValue
Change-Id: I1dee5060bd4f8dcbdcd542ec4647ea328f0185c3
Merge commit '0ea4ed3bbb28fb6913392d2bee55621a1290dca8' into gingerbread-plus-aosp
* commit '0ea4ed3bbb28fb6913392d2bee55621a1290dca8':
Don't drop a late frame which may lead to missing I frames in the MP4 file
Merge commit '439fe407ff75b2c0fc21c66b430cd76e9f29ac90' into gingerbread-plus-aosp
* commit '439fe407ff75b2c0fc21c66b430cd76e9f29ac90':
Return error from MPEG4Writer stop() if the check on codec specific data failed
Merge commit 'cbd038fe207f183bc7e0a610973473f7c2e9d118' into gingerbread-plus-aosp
* commit 'cbd038fe207f183bc7e0a610973473f7c2e9d118':
Make MediaWriter stop and pause return errors if necessary
o Make the API consistent with SF framework, which the MediaSource
provides a return status for stop
o Also, helps to convey errors that occurred right when a
premature stop() is called, leading to a potentially
mal-formed output file.
Change-Id: I52a932345f38570fdf8ea04d67d73dd94ccd30ef
Merge commit '223e4f732a325e456ca6151f132f1d4c3c625631' into gingerbread-plus-aosp
* commit '223e4f732a325e456ca6151f132f1d4c3c625631':
Support for MP4V-ES packetization format according to RFC3016.
Merge commit 'f0ad54846168f07fc1fd7f18cde93deea1559f86' into gingerbread-plus-aosp
* commit 'f0ad54846168f07fc1fd7f18cde93deea1559f86':
In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
Merge commit '8c192fe990d7bc7149d2ec1a7c9f4ada3f32e52a' into gingerbread-plus-aosp
* commit '8c192fe990d7bc7149d2ec1a7c9f4ada3f32e52a':
Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
The "strength supported" parameter for bass boost and virtualizer effect was incorrectly using a
short value whereas it should be an int. This is to comply to the definition of boolean type in OpenSL ES
that is uint32.
Change-Id: I74ccb61dcc70fc9d390524a1ca5bbbd8b13ab1af
Merge commit '31eb1ac1db38d0a5cd0b44dd5251941992f74b58'
* commit '31eb1ac1db38d0a5cd0b44dd5251941992f74b58':
Be more lenient when validating ESDS information in mp4 audio tracks. Allow the absence of any codec specific data and assume that the mpeg4 headers are not lying to us.
Merge commit '4dda6ddb25e904c17dcb3012dd229df6ae4692cd' into gingerbread-plus-aosp
* commit '4dda6ddb25e904c17dcb3012dd229df6ae4692cd':
Make the OggExtractor less verbose.
Merge commit '0324ce9a1e21ed66e00d6560c27a6faf6d151f68' into gingerbread-plus-aosp
* commit '0324ce9a1e21ed66e00d6560c27a6faf6d151f68':
Be more lenient when validating ESDS information in mp4 audio tracks. Allow the absence of any codec specific data and assume that the mpeg4 headers are not lying to us.
This should eliminate the spammy logging that my coworkers have noticed.
Change-Id: Ic0e611f5277dd13651490cbe5f7ded5f6e17db4f
Signed-off-by: Mike Lockwood <lockwood@android.com>
Merge commit 'f54da15b7c3fa55268451c485544e831832fdf15'
* commit 'f54da15b7c3fa55268451c485544e831832fdf15':
Change the default time scale for audio/video track during recording
Merge commit '4fc2c9280c5262c835a4eb78961241de105313c1'
* commit '4fc2c9280c5262c835a4eb78961241de105313c1':
Use audio clock as the reference media clock
Merge commit 'eff30e3d1b005fd0696390d1dd47ec4ff0c52784' into gingerbread-plus-aosp
* commit 'eff30e3d1b005fd0696390d1dd47ec4ff0c52784':
Change the default time scale for audio/video track during recording
and reduce rounding errors in calculating the sample duration
- Default time scale for tracks other than audio is set to 90000.
- Audio track by default uses the audio sampling rate as the time scale.
- Default movie time scale remains to be 1000.
- The default time scale values will be overwritten by a user-supplied value if exits.
Change-Id: I81b40ed0626ea45e9fd24a89e21a2c5a4a2c3415
Merge commit 'b72081966da3842e27f88045cfa5a67cef3d4220' into gingerbread-plus-aosp
* commit 'b72081966da3842e27f88045cfa5a67cef3d4220':
Use audio clock as the reference media clock
o Only do this for realtime applications
o Adjust other track clock based on audio clock
o Assume other track uses wall clock as the media clock
o Use some heuristics to reduce the size of stts box by 2/3.
- also
o Remove one unused key from MetaData.h
Change-Id: Ib9432842627b61795b533508158c25258a527332
Merge commit 'e95d192fae5a80ed821c53bfea214a85ea395e90' into gingerbread-plus-aosp
* commit 'e95d192fae5a80ed821c53bfea214a85ea395e90':
Mainly fix two mistakes that I made:
Merge commit '2f02044944d5c526020d4e8cceaae7e77382d56d'
* commit '2f02044944d5c526020d4e8cceaae7e77382d56d':
Support getting codec, width, and height in URL for gtalk playback.
Merge commit '5f96138ba65cecf38d0c752d87ad47d931db8775' into gingerbread-plus-aosp
* commit '5f96138ba65cecf38d0c752d87ad47d931db8775':
Support getting codec, width, and height in URL for gtalk playback.
