2147 Commits

Author SHA1 Message Date
Andreas Huber
6b6ae996b2 Merge "A first shot at proper support for seeking of rtsp streams." into gingerbread 2010-08-24 15:01:14 -07:00
Andreas Huber
e0dd7d3960 A first shot at proper support for seeking of rtsp streams.
Change-Id: I9604f2d09feedc0074c0e715be58e719d4483760
related-to-bug: 2556656
2010-08-24 14:33:58 -07:00
James Dong
05e80b4c1c Make sure that timestamp does not go backward in MP4 file writer
Change-Id: I90745b9df7f19d61f3ab826bf9d2419fe788554e
2010-08-24 12:28:02 -07:00
Eric Laurent
33e0d83431 am b6d71351: Merge "LVM release 1.05 delivery" into gingerbread
Merge commit 'b6d71351c074d5c0bc13a91544d776f1524eaabd' into gingerbread-plus-aosp

* commit 'b6d71351c074d5c0bc13a91544d776f1524eaabd':
  LVM release 1.05 delivery
2010-08-24 09:58:23 -07:00
Eric Laurent
b6d71351c0 Merge "LVM release 1.05 delivery" into gingerbread 2010-08-24 09:50:13 -07:00
Andreas Huber
31e7113104 am 3e22ef1e: Merge "Better handling of rtsp connection and disconnection." into gingerbread
Merge commit '3e22ef1e111966df6ad527632fdc35d105c73916' into gingerbread-plus-aosp

* commit '3e22ef1e111966df6ad527632fdc35d105c73916':
  Better handling of rtsp connection and disconnection.
2010-08-23 12:28:26 -07:00
Andreas Huber
3e22ef1e11 Merge "Better handling of rtsp connection and disconnection." into gingerbread 2010-08-23 12:26:31 -07:00
Andreas Huber
8370be11de Better handling of rtsp connection and disconnection.
Change-Id: Ib126af6c14c5a212a51a5ee3c4a0a7d1860ad167
2010-08-23 11:28:34 -07:00
James Dong
28a92120a7 am 3f51fa78: Runtime dump support for MediaWriter
Merge commit '3f51fa78ada0e064d23db5961337280c267cc2c0' into gingerbread-plus-aosp

* commit '3f51fa78ada0e064d23db5961337280c267cc2c0':
  Runtime dump support for MediaWriter
2010-08-23 10:50:29 -07:00
James Dong
3f51fa78ad Runtime dump support for MediaWriter
Change-Id: I10b2c474de612ee4cef4b7c9eae2ee1dd8c2e895
2010-08-23 10:34:05 -07:00
Chia-chi Yeh
a102871c7c am b80e610b: Merge "Visualizer: replace the FFT implementation with a faster one." into gingerbread
Merge commit 'b80e610b070c2cec98a228a8aec450dc24a5f90a' into gingerbread-plus-aosp

* commit 'b80e610b070c2cec98a228a8aec450dc24a5f90a':
  Visualizer: replace the FFT implementation with a faster one.
2010-08-22 18:38:29 -07:00
Chia-chi Yeh
b80e610b07 Merge "Visualizer: replace the FFT implementation with a faster one." into gingerbread 2010-08-22 18:31:15 -07:00
Nipun Kwatra
701b710c19 am 300b0b7e: Merge "setParamMaxFileDurationUs should allow zero time input as per API of setMaxDuration." into gingerbread
Merge commit '300b0b7e2b8f0ab922e4a83755ae999da191894e' into gingerbread-plus-aosp

* commit '300b0b7e2b8f0ab922e4a83755ae999da191894e':
  setParamMaxFileDurationUs should allow zero time input as per API of setMaxDuration.
2010-08-20 14:18:08 -07:00
Eric Laurent
a1a96f3570 LVM release 1.05 delivery
- Click have been removed from the HP filter activation in the BassBosst Effect.
- SessionId is now stored as a SessionNo
- Effects now stop being called after a delay
- Unix EOL fixed for .java and .xml
- Updated lines limited to 100 characters.
- Removed the remaining warnings from the wrapper code
- Added reverb

Change-Id: I03a2b3b5ee2286958f4901acc8d9b0daf9e2d7c6
2010-08-20 14:17:41 -07:00
Nipun Kwatra
fb45748a1e setParamMaxFileDurationUs should allow zero time input as per API of setMaxDuration.
according to MediaRecorder::setMaxDuration documentation we should disable duration limit
when zero or negative time is passed. Currently setParamMaxFileDurationUs was treating
zero/negative as an error case. Fixed that.

Change-Id: I468c3bcc74cb5a34ee3e172cef5147550d6be096
2010-08-20 14:06:48 -07:00
James Dong
3fc01525ff am b755e325: Merge "Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder" into gingerbread
Merge commit 'b755e3256510ecd325565d6b461d668d224445b1' into gingerbread-plus-aosp

* commit 'b755e3256510ecd325565d6b461d668d224445b1':
  Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder
2010-08-20 11:41:49 -07:00
James Dong
9767dbf923 Only add 4 bytes offset for the output media buffer when SPS is not received for SW AVC encoder
Change-Id: Ia64c2751b6304e5d5891416bf23ff9b8ec54d5ef
2010-08-19 21:13:35 -07:00
James Dong
3540760d1d am 0ea4ed3b: Don\'t drop a late frame which may lead to missing I frames in the MP4 file
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
2010-08-19 18:07:08 -07:00
James Dong
177a7ad825 am 439fe407: Merge "Return error from MPEG4Writer stop() if the check on codec specific data failed" into gingerbread
Merge commit '439fe407ff75b2c0fc21c66b430cd76e9f29ac90' into gingerbread-plus-aosp

* commit '439fe407ff75b2c0fc21c66b430cd76e9f29ac90':
  Return error from MPEG4Writer stop() if the check on codec specific data failed
2010-08-19 18:04:47 -07:00
James Dong
0ea4ed3bbb Don't drop a late frame which may lead to missing I frames in the MP4 file
Change-Id: I8fef1454264230c1369561670236eb0a19ae4e76
2010-08-19 18:04:27 -07:00
James Dong
62948fa4de Return error from MPEG4Writer stop() if the check on codec specific data failed
Change-Id: Icbd08eec9b4201facbad56ff2040f0830cfb0115
2010-08-19 14:04:04 -07:00
James Dong
c8d2fa704a am cbd038fe: Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread
Merge commit 'cbd038fe207f183bc7e0a610973473f7c2e9d118' into gingerbread-plus-aosp

* commit 'cbd038fe207f183bc7e0a610973473f7c2e9d118':
  Make MediaWriter stop and pause return errors if necessary
2010-08-19 14:02:25 -07:00
James Dong
cbd038fe20 Merge "Make MediaWriter stop and pause return errors if necessary" into gingerbread 2010-08-19 13:59:32 -07:00
James Dong
d036662470 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
2010-08-19 13:33:13 -07:00
Andreas Huber
873ebfb825 am 223e4f73: Merge "Support for MP4V-ES packetization format according to RFC3016." into gingerbread
Merge commit '223e4f732a325e456ca6151f132f1d4c3c625631' into gingerbread-plus-aosp

* commit '223e4f732a325e456ca6151f132f1d4c3c625631':
  Support for MP4V-ES packetization format according to RFC3016.
2010-08-19 11:22:14 -07:00
Andreas Huber
a979ad6739 Support for MP4V-ES packetization format according to RFC3016.
Change-Id: I5e182936c52f9eb80cdcf6132ead03705ee32d61
2010-08-19 11:18:35 -07:00
Andreas Huber
b29ebd397e am f0ad5484: Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread
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.
2010-08-19 10:56:40 -07:00
Andreas Huber
f0ad548461 Merge "In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data." into gingerbread 2010-08-19 10:54:21 -07:00
Eric Laurent
5aff90a638 am 1aaba885: Merge "Audio Effects: fixed "strength supported" parameter size." into gingerbread
Merge commit '1aaba885def9a3b59edbfe2a0f8c3899948533ff' into gingerbread-plus-aosp

* commit '1aaba885def9a3b59edbfe2a0f8c3899948533ff':
  Audio Effects: fixed "strength supported" parameter size.
2010-08-19 10:47:01 -07:00
Eric Laurent
1aaba885de Merge "Audio Effects: fixed "strength supported" parameter size." into gingerbread 2010-08-19 10:42:36 -07:00
Andreas Huber
eef3c33e56 In the absence of width/height information in the sdp, extract the dimensions from the avc codec specific data.
Change-Id: I98c4194593c7e6e24f6fc339c862245111800293
2010-08-19 10:39:47 -07:00
Andreas Huber
6bcffcd2dc am 8c192fe9: Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread
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.
2010-08-19 09:11:28 -07:00
Andreas Huber
8c192fe990 Merge "Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description." into gingerbread 2010-08-19 09:09:12 -07:00
Chia-chi Yeh
58d3bd0810 Visualizer: replace the FFT implementation with a faster one.
This implementation uses fixed points instead of floating points. It
is slightly inaccurate compared to the old one but still perfect for
visualization purpose. It runs 40% faster on passion, 5 times faster
on sholes, and of course 14 times faster on sapphire.

Change-Id: I1e868417bcffda091becf106a7b941d02813faec
2010-08-19 16:05:32 +08:00
Eric Laurent
ba8da2e61b Audio Effects: fixed "strength supported" parameter size.
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
2010-08-18 14:31:25 -07:00
Andreas Huber
00557baf36 am 4dda6ddb: Merge "Make the OggExtractor less verbose." into gingerbread
Merge commit '4dda6ddb25e904c17dcb3012dd229df6ae4692cd' into gingerbread-plus-aosp

* commit '4dda6ddb25e904c17dcb3012dd229df6ae4692cd':
  Make the OggExtractor less verbose.
2010-08-18 13:38:53 -07:00
Andreas Huber
31eb1ac1db am 0324ce9a: Merge "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." into gingerbread
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.
2010-08-18 13:38:49 -07:00
Andreas Huber
af063a67b2 Better support for rtsp streamed through VLC. Temporarily make the socket blocking to read all of the session description.
Change-Id: Ibe71f5941485660510e24d714da3865b9c6f89a2
2010-08-18 10:17:18 -07:00
Andreas Huber
4dda6ddb25 Merge "Make the OggExtractor less verbose." into gingerbread 2010-08-18 10:10:14 -07:00
Andreas Huber
08c94b265c 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.
Change-Id: Ia29b967cbee9eabb21c6c26508b27b379ff9ba17
2010-08-18 09:58:30 -07:00
Jean-Baptiste Queru
09f672509b resolved conflicts for merge of 0b7bd95d to gingerbread-plus-aosp
Change-Id: I55c1689c7d0737c943efec28d8164d6a5360621c
2010-08-17 15:24:11 -07:00
Rene Bolldorf
0b7bd95d69 Fix compilation errors in libmedia, libstagefright.
(invalid conversion from 'const char*' to 'char*')

Change-Id: Idef85606b7cff629b2778ed8134c79c892af54c2
2010-08-17 23:45:14 +02:00
Andreas Huber
3386c38d59 Make the OggExtractor less verbose.
Change-Id: Ieea5f3fa98d93ca6ad8fa7dcd23054e1cd0b6338
2010-08-16 14:11:40 -07:00
James Dong
f54da15b7c am eff30e3d: Change the default time scale for audio/video track during recording and reduce rounding errors in calculating the sample duration
Merge commit 'eff30e3d1b005fd0696390d1dd47ec4ff0c52784' into gingerbread-plus-aosp

* commit 'eff30e3d1b005fd0696390d1dd47ec4ff0c52784':
  Change the default time scale for audio/video track during recording
2010-08-16 13:20:15 -07:00
James Dong
eff30e3d1b 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
2010-08-16 10:38:35 -07:00
James Dong
4fc2c9280c am b7208196: Use audio clock as the reference media clock
Merge commit 'b72081966da3842e27f88045cfa5a67cef3d4220' into gingerbread-plus-aosp

* commit 'b72081966da3842e27f88045cfa5a67cef3d4220':
  Use audio clock as the reference media clock
2010-08-13 18:28:52 -07:00
James Dong
b72081966d 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
2010-08-13 18:12:48 -07:00
James Dong
38a9f4050a am e95d192f: Mainly fix two mistakes that I made:
Merge commit 'e95d192fae5a80ed821c53bfea214a85ea395e90' into gingerbread-plus-aosp

* commit 'e95d192fae5a80ed821c53bfea214a85ea395e90':
  Mainly fix two mistakes that I made:
2010-08-12 17:02:42 -07:00
Mike Dodd
2f02044944 am 5f96138b: Merge "Support getting codec, width, and height in URL for gtalk playback." into gingerbread
Merge commit '5f96138ba65cecf38d0c752d87ad47d931db8775' into gingerbread-plus-aosp

* commit '5f96138ba65cecf38d0c752d87ad47d931db8775':
  Support getting codec, width, and height in URL for gtalk playback.
2010-08-12 16:47:52 -07:00
James Dong
e95d192fae Mainly fix two mistakes that I made:
1. When the ERROR_END_OF_STREAM is returned from read, the input buffer is not initialized
   release it would lead to crash

2. The mPrevTimestampUs is not initialized and thus fail in the CHECK(mPrevTimestampUs, timeUs)

Change-Id: Id1e51575fb8b3ca48e80547efd3a3a82dfac773b
2010-08-12 16:47:17 -07:00