OSDN Git Service
Andreas Huber [Wed, 27 Feb 2013 16:53:21 +0000 (16:53 +0000)]
Merge "MPEG4Writer: Write a zero-length compressor string"
Martin Storsjo [Wed, 30 Jan 2013 16:45:30 +0000 (18:45 +0200)]
MPEG4Writer: Write a zero-length compressor string
The first byte is the number of bytes in the string to display,
the rest is the actual string (see ISO/IEC 14496-12:2008 page 29,
compressorname in VisualSampleEntry) to display. Currently, all 31
space characters are displayed in certain players (such as Apple
QuickTime Player) instead of the plain codec name (e.g. "H.264").
By writing the string length properly, the actual content of the
remaining 31 bytes is ignored instead of displayed.
Change-Id: I7f93a8966d74a4e86abc46c3dcb70f71780d9cd6
Eino-Ville Talvala [Thu, 31 Jan 2013 02:57:21 +0000 (18:57 -0800)]
am
9b7e8fc7: Camera2: Fix JPEG quality metadata type, new warnings.: DO NOT MERGE
# Via Igor Murashkin
* commit '
9b7e8fc7c564b24a1c099dddae4e9c1126bd7c4e':
Camera2: Fix JPEG quality metadata type, new warnings.: DO NOT MERGE
Igor Murashkin [Thu, 31 Jan 2013 02:57:20 +0000 (18:57 -0800)]
am
abeee0ec: Camera2: fix metadata symbols: DO NOT MERGE
# Via Igor Murashkin
* commit '
abeee0ec92b5c77147036bf346de9309092a5ca1':
Camera2: fix metadata symbols: DO NOT MERGE
Eino-Ville Talvala [Fri, 4 Jan 2013 20:05:56 +0000 (12:05 -0800)]
Camera2: Fix JPEG quality metadata type, new warnings.: DO NOT MERGE
- JPEG quality tag is now a uint8_t, not an int. Update parameter code
accordingly.
- Fix new warnings about narrowing conversions.
Bug:
7944244
Change-Id: Ie081c57c9e9323148614b170b132ffb98c0a0b9f
Igor Murashkin [Wed, 28 Nov 2012 00:25:46 +0000 (16:25 -0800)]
Camera2: fix metadata symbols: DO NOT MERGE
Change-Id: I59e457824782de26b7b489cd92eb33d48e6ee2d9
James Dong [Fri, 21 Dec 2012 22:42:34 +0000 (14:42 -0800)]
am
12834fe2: Merge "libvideoeditor: fix code bugs when enabling all-level debugging macros"
* commit '
12834fe289e8173e94f31603fa6d313abe56ad70':
libvideoeditor: fix code bugs when enabling all-level debugging macros
James Dong [Fri, 21 Dec 2012 22:09:14 +0000 (14:09 -0800)]
Merge "libvideoeditor: fix code bugs when enabling all-level debugging macros"
James Dong [Fri, 21 Dec 2012 18:05:52 +0000 (10:05 -0800)]
am
b54cedab: Merge "Use default values when MPEG4 audio config parsing fails."
* commit '
b54cedabdf0261211241e2f3af09c75cffd911ed':
Use default values when MPEG4 audio config parsing fails.
James Dong [Fri, 21 Dec 2012 02:25:12 +0000 (18:25 -0800)]
Merge "Use default values when MPEG4 audio config parsing fails."
James Dong [Thu, 20 Dec 2012 22:08:05 +0000 (14:08 -0800)]
am
2a55b596: Merge "libvideoeditor: fix cannot show some thumbnails in storyboard or no frame display in PreviewRenderer surface after importing video"
* commit '
2a55b5964c6ae167cc1bc57eca5ca7a1c4aa452c':
libvideoeditor: fix cannot show some thumbnails in storyboard or no frame display in PreviewRenderer surface after importing video
James Dong [Thu, 20 Dec 2012 21:15:28 +0000 (13:15 -0800)]
Merge "libvideoeditor: fix cannot show some thumbnails in storyboard or no frame display in PreviewRenderer surface after importing video"
Glenn Kasten [Thu, 20 Dec 2012 19:03:48 +0000 (11:03 -0800)]
am
df9cce64: Merge "Allow multichannel FLAC files"
* commit '
df9cce649d9ac1f4446f613c54191e7c4d646f07':
Allow multichannel FLAC files
Glenn Kasten [Thu, 20 Dec 2012 15:30:41 +0000 (07:30 -0800)]
Merge "Allow multichannel FLAC files"
Glenn Kasten [Mon, 17 Dec 2012 21:18:08 +0000 (13:18 -0800)]
am
f38259cf: Merge "Fix bug in WAVExtractor for 24-bit per sample wav"
* commit '
f38259cfe5566914a49f5b0f8b11aa6a951d667a':
Fix bug in WAVExtractor for 24-bit per sample wav
Glenn Kasten [Mon, 17 Dec 2012 20:29:06 +0000 (12:29 -0800)]
Merge "Fix bug in WAVExtractor for 24-bit per sample wav"
Jean-Baptiste Queru [Fri, 14 Dec 2012 23:25:22 +0000 (15:25 -0800)]
am
0cd75629: Merge "Match new paths and organization in external/libvpx"
* commit '
0cd75629a8f7a6bd47b2abc6dd8d9f66e70d76a4':
Match new paths and organization in external/libvpx
Marco Nelissen [Fri, 14 Dec 2012 23:25:21 +0000 (15:25 -0800)]
am
409542cd: Merge "FileSource: FILE is not closed"
* commit '
409542cdaf72d8983043f2746d9d916ee1566a80':
FileSource: FILE is not closed
Jean-Baptiste Queru [Fri, 14 Dec 2012 20:48:41 +0000 (12:48 -0800)]
Merge "Match new paths and organization in external/libvpx"
Marco Nelissen [Fri, 14 Dec 2012 20:05:02 +0000 (12:05 -0800)]
Merge "FileSource: FILE is not closed"
You Kim [Fri, 14 Dec 2012 13:22:36 +0000 (22:22 +0900)]
FileSource: FILE is not closed
Add destructor to call fclose.
Change-Id: I765c56236f8df58af998439492b0b955ab57606f
Johann [Mon, 29 Oct 2012 23:48:23 +0000 (16:48 -0700)]
Match new paths and organization in external/libvpx
See I2c0a87b0fc049e855e21f455b806287b578091de
Change-Id: Iec8566bce0a4763a9a17b4a46f15aa39b5bcd141
Glenn Kasten [Mon, 10 Dec 2012 23:34:56 +0000 (15:34 -0800)]
am
f0ff908d: Logging to investigate a crash
* commit '
f0ff908da019a44115109f1b4d1b6864b35a8a29':
Logging to investigate a crash
Glenn Kasten [Fri, 7 Dec 2012 22:13:35 +0000 (14:13 -0800)]
Logging to investigate a crash
Bug:
6490974
Change-Id: Ib926a9258bde4ee05ed42eea662dff68e426a997
Andreas Huber [Thu, 6 Dec 2012 18:39:00 +0000 (10:39 -0800)]
am
03425d9c: Merge "Fix bad checks that causes crash when streaming H.263 content."
* commit '
03425d9cf7f22cf400c13cda60d3e91f0fd48d6e':
Fix bad checks that causes crash when streaming H.263 content.
Andreas Huber [Wed, 5 Dec 2012 16:58:56 +0000 (08:58 -0800)]
Merge "Fix bad checks that causes crash when streaming H.263 content."
Andreas Huber [Mon, 3 Dec 2012 17:17:51 +0000 (09:17 -0800)]
am
f00d40c7: Merge "Memory leak solved in MPEG4Extractor::parseChunk for cover art"
* commit '
f00d40c716fd4741fa56dd2b59813227658e50f2':
Memory leak solved in MPEG4Extractor::parseChunk for cover art
Andreas Huber [Mon, 3 Dec 2012 17:00:29 +0000 (09:00 -0800)]
Merge "Memory leak solved in MPEG4Extractor::parseChunk for cover art"
Roger1 Jonsson [Tue, 21 Dec 2010 08:57:41 +0000 (09:57 +0100)]
Fix bad checks that causes crash when streaming H.263 content.
Remove checks that causes crash for rtsp streamed h.263 content
with certain values in the RTP payload header:
Remove zero check for the five reserved bits in the payload header.
According to RFC 4629 these bits MUST be ignored by receivers.
Remove zero-check for the VRC (Video Redundancy Coding) bit,
skip packet instead.
Remove zero-check for the PLEN bits (extra picture header),
skip packet instead.
Remove zero-check for the PEBIT bits (extra picture header),
skip packet instead.
Remove corresponding zero check for the four resreved bits in the
AMR payload header. According to RFC 4867 these bits MUST be
ignored by receivers.
Change-Id: I7fc21d69a19d23da24f9267623c338d415ef1387
cong.zhou [Wed, 28 Nov 2012 23:52:28 +0000 (15:52 -0800)]
Fix bug in WAVExtractor for 24-bit per sample wav
support stereo/multichannel 24-bit format
kMaxFrameSize is fixed to 32768. When converting 24-bit to 16-bit,
number of samlpes is maxBytesToRead /3.
In this case, if the maxBytesToRead is not multiple of 3,
pcm data is messed when converting.
Bug:
7630939
Change-Id: I0ea1b53eb1272a8d83b63815fc0a05b73cef75f1
Andreas Huber [Fri, 30 Nov 2012 16:50:09 +0000 (08:50 -0800)]
am
e7550eb2: Merge "Initialize and copy mRotationAngle in VideoFrame"
* commit '
e7550eb2805d864253f88ea2ced57014d6140e55':
Initialize and copy mRotationAngle in VideoFrame
Andreas Huber [Fri, 30 Nov 2012 16:30:40 +0000 (08:30 -0800)]
Merge "Initialize and copy mRotationAngle in VideoFrame"
Christer Fletcher [Thu, 13 Sep 2012 08:28:39 +0000 (10:28 +0200)]
Initialize and copy mRotationAngle in VideoFrame
When thumbnails were generated they could be generated at random
angles as the mRotationAngle variable was not initialized to any
value. This variable would have to be explicitly overwritten to not
cause random rotation. Changed the implementation to initialize the
value to 0 (no rotation). mRotationAngle was also missing in the
copy constructor.
Change-Id: I67a5340fdd807c6ab3a3da5eecb09b5b9d5f4666
Dylan Powers [Thu, 29 Nov 2012 23:20:31 +0000 (15:20 -0800)]
am
47c1a5f7: Bug fix for the MediaPlayer::prepare() api.
* commit '
47c1a5f7c13d82aa8834fd4543bd1d713b97808e':
Bug fix for the MediaPlayer::prepare() api.
Dylan Powers [Wed, 28 Nov 2012 00:06:38 +0000 (16:06 -0800)]
Bug fix for the MediaPlayer::prepare() api.
For an MP3 source, within the prepare command, ID3 tags are checked in search of
gapless playback info. This causes problems for streamed sources. If ID3v2 tags
aren't present, then a check is done for ID3v1 tags. This results in a read
command that asks the cache to jump to the end of the file, and subsequently
make an extra http call to request those bytes. For a streamed source, this
causes the file to not be downloaded automatically when MediaPlayer::prepare()
is called, and causes stuttering and extra buffering time to be needed when
start() is finally called.
The solution is to ignore the ID3v1 tags as the gapless info would never exist
there, and only check for ID3v2 tags.
Cherrypicked from external contribution
ffd6ffc5429c45577fd8e9f8fa90e79bb91b8a84
b/
7638165
Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
Glenn Kasten [Thu, 29 Nov 2012 22:28:51 +0000 (14:28 -0800)]
Andreas Huber [Thu, 29 Nov 2012 22:27:59 +0000 (14:27 -0800)]
am
b64def9a: Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into jb-mr1.1-dev
* commit '
b64def9a555bfbf533a2da41ba0189b9842a76fb':
[wfd] Support a low(er) power state by triggering PAUSE/RESUME.
Andreas Huber [Thu, 29 Nov 2012 22:27:53 +0000 (14:27 -0800)]
am
e5aed03d: Enable retransmission of UDP packets in case we want to use it
* commit '
e5aed03d30ea0ce49728873c5b74f89ba05a9541':
Enable retransmission of UDP packets in case we want to use it
James Dong [Thu, 29 Nov 2012 22:25:49 +0000 (14:25 -0800)]
am
79c56d3f: Merge "Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display" into jb-mr1.1-dev
* commit '
79c56d3f17d3193a0a86eb3c9bfdea90b89ae3f9':
Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display
James Dong [Thu, 29 Nov 2012 22:25:47 +0000 (14:25 -0800)]
am
0dbe5a93: Added optional intra macroblock refresh support for encoding
* commit '
0dbe5a9321b24b6883fbb2fe97cd9d525128b0b5':
Added optional intra macroblock refresh support for encoding
Glenn Kasten [Thu, 29 Nov 2012 22:09:41 +0000 (14:09 -0800)]
Fix log spam
Change-Id: Ie6c982af906dcfd3cdea4b771dfab1f7e47745ca
Marco Nelissen [Thu, 29 Nov 2012 19:33:11 +0000 (11:33 -0800)]
am
031c93df: Merge "Bug fix for the MediaPlayer::prepare() api."
* commit '
031c93df74621dc2149876dc377aedee8930547f':
Bug fix for the MediaPlayer::prepare() api.
Marco Nelissen [Thu, 29 Nov 2012 19:08:16 +0000 (11:08 -0800)]
Merge "Bug fix for the MediaPlayer::prepare() api."
Andreas Huber [Thu, 29 Nov 2012 18:57:06 +0000 (10:57 -0800)]
Merge "[wfd] Support a low(er) power state by triggering PAUSE/RESUME." into jb-mr1.1-dev
Andreas Huber [Thu, 29 Nov 2012 17:54:42 +0000 (09:54 -0800)]
Enable retransmission of UDP packets in case we want to use it
in our upcoming wfd _sink_ implementation.
Change-Id: I4509c30d5a7b992bc841b73d63db902bbcf8f76a
related-to-bug:
7638155
Andreas Huber [Fri, 16 Nov 2012 18:38:11 +0000 (10:38 -0800)]
[wfd] Support a low(er) power state by triggering PAUSE/RESUME.
Change-Id: Ibe42bfa73816bbfeb7e652d435254d0171b89727
related-to-bug:
7638150
Andreas Huber [Thu, 29 Nov 2012 17:14:30 +0000 (09:14 -0800)]
am
251c04b1: Merge "Unsolicited server responses cause RTSP streaming to crash"
* commit '
251c04b1f3d048f541832c93347c6604b314e0ea':
Unsolicited server responses cause RTSP streaming to crash
Andreas Huber [Thu, 29 Nov 2012 16:51:21 +0000 (08:51 -0800)]
Merge "Unsolicited server responses cause RTSP streaming to crash"
Lena Magnusson [Fri, 20 Jan 2012 08:39:38 +0000 (09:39 +0100)]
Unsolicited server responses cause RTSP streaming to crash
If the set up of the RTSP stream contains an incorrect or otherwise
problematic URL, some servers will send an unsolicited server response
that contains a negative number in the sequence number (CSeq) field.
This negative value is not returned from the function findPendingRequest(),
so the check in notifyResponseListener() will not work. Instead there will
be a crash when 0 is used as the index to find a matching request/response
pair that doesn’t exist.
The fix is to return the received sequence number also when it is an
unsolicited server-client message.
Change-Id: Iedaba8a63dece7b43bce007069baefbfd10970b8
James Dong [Wed, 28 Nov 2012 23:40:54 +0000 (15:40 -0800)]
Merge "Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display" into jb-mr1.1-dev
James Dong [Fri, 16 Nov 2012 02:31:50 +0000 (18:31 -0800)]
Reduce the frequency of IDR frames and add intra-fresh mode support for WiFi display
The time interval between periodic neighboring IDR frames is increased from 1 second to 15 seconds.
o related-to-bug:
7524791
Change-Id: Ic32f37448f952f329549eda5e73637ee3b02f046
James Dong [Thu, 15 Nov 2012 22:00:26 +0000 (14:00 -0800)]
Added optional intra macroblock refresh support for encoding
o related-to-bug:
7524791
Change-Id: I95ac4ee925e2dbeb00b3cfb2e29c611698c5cc9f
Andreas Huber [Wed, 28 Nov 2012 17:42:18 +0000 (09:42 -0800)]
am
b7c8e918: Add support for HLS playlists of type \'event\'.
* commit '
b7c8e91880463ff4981e3e53e98e45d68e2fe374':
Add support for HLS playlists of type 'event'.
Dylan Powers [Wed, 28 Nov 2012 00:06:38 +0000 (16:06 -0800)]
Bug fix for the MediaPlayer::prepare() api.
For an MP3 source, within the prepare command, ID3 tags are checked in search of
gapless playback info. This causes problems for streamed sources. If ID3v2 tags
aren't present, then a check is done for ID3v1 tags. This results in a read
command that asks the cache to jump to the end of the file, and subsequently
make an extra http call to request those bytes. For a streamed source, this
causes the file to not be downloaded automatically when MediaPlayer::prepare()
is called, and causes stuttering and extra buffering time to be needed when
start() is finally called.
The solution is to ignore the ID3v1 tags as the gapless info would never exist
there, and only check for ID3v2 tags.
Change-Id: I7d1b94cffbfe7c38ca094834dedbc92a58855e20
Andreas Huber [Tue, 27 Nov 2012 23:02:53 +0000 (15:02 -0800)]
Add support for HLS playlists of type 'event'.
related-to-bug:
6870049
Squashed commit of the following:
commit
eee2f3ba6bb7335f4e285632726db85645669929
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 15:02:01 2012 -0800
Make everything a lot less verbose by default.
Change-Id: I884d7a7901aa1e7d4ff590f065ca57a79d2af8b3
commit
6bbdb837ed5bd88008e45efb8faf595e4051ba26
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 14:34:46 2012 -0800
HLS now properly signals media time changes at discontinuities including
the start of playback (which may not necessarily be at time 0 if the playlist
is of type 'event' and hasn't completed yet).
Change-Id: I5ab747d024f9b8d0df72a4e06a12ebb29f62802e
commit
1555589832b1878a144a976a643e1af4d61f877c
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 14:32:28 2012 -0800
As part of a time discontinuity, clients of IStreamListener can now
signal the corresponding media time after the discontinuity, i.e. the first PTS
timestamp following the discontinuity will be considered equivalent to the
specified media time and media buffers timestamped accordingly.
Change-Id: Id7db7679b7faa6efd6270620ff52e34e884f3e92
commit
5c24c605c073a11c426d025b1e7478fc1ad8365a
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 13:00:56 2012 -0800
NuPlayer sources now expose flags() and can announce
that duration may change (increase) dynamically, in which case duration
will be polled at 1 second intervals and communicated to the upper layers.
Change-Id: I45102909b7a19eed0dda576747e3814d742a0eea
commit
ecb71de8e281e61971a2cd73e7161a97540bc357
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 12:57:47 2012 -0800
Stop caching duration in MediaPlayer, duration could increase dynamically.
Change-Id: I7bb2f16c0abe49debdf45c776d2266aa069d7791
commit
544aec5823e6d7a3e97e15b6b23546616bcd343e
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 27 08:46:28 2012 -0800
An attempt to add support for "event" style HLS playlists.
Change-Id: I3dfb2e801ecaff8f5d8bdb3a4fca1b18aeeb2c60
Change-Id: I48cf7f65a654d33f2f49ded74f8be22aed9e3b98
Glenn Kasten [Tue, 27 Nov 2012 23:05:48 +0000 (15:05 -0800)]
resolved conflicts for merge of
41829f30 to jb-mr1-dev-plus-aosp
Change-Id: I952d333c2e88b6b28663793046a136822d1b6838
Glenn Kasten [Tue, 27 Nov 2012 22:51:19 +0000 (14:51 -0800)]
Merge "New VHQ resampler" into jb-mr1.1-dev
Glenn Kasten [Mon, 19 Nov 2012 17:00:47 +0000 (09:00 -0800)]
New VHQ resampler
Squashed commit of the following:
commit
12b6952da9f25e94d06dd7185bce255924e7e791
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 19 15:27:26 2012 -0800
fix a typo in SINC resampler that prevented tracks to be mixed
we were always erasing the current mix instead of mixing into it.
Change-Id: Ib229245f9e5a0d384f1727640a59e9f0469211a2
commit
0019ce082df430278f14ab922e900ce33b64897d
Author: Dave Bort <dbort@google.com>
Date: Tue Nov 13 01:30:32 2007 -0800
Rename "TARGET" to "MODULE" in the build system.
Part one of the grand renaming.
API_CHANGE: Third parties may need to update their makefiles.
Any variables with "LOCAL" and "TARGET" in their names
should now use "MODULE" instead of "TARGET"; e.g., LOCAL_MODULE,
LOCAL_MODULE_TAGS.
PRESUBMIT=passed
OCL=39840
Change-Id: Ica9a7937d3d9552ab84db46ac6eea8a290e404fe
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit
f01adc0cef0e39e75c76d9195ac26a94cac0a100
Author: Glenn Kasten <gkasten@google.com>
Date: Wed Nov 14 08:32:08 2012 -0800
Fix build warnings
Change-Id: Ic43bcca166a529a6431711b05a7fa21849b6a38b
commit
9bb031a565c753a03d9c9397edea318947d80528
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 04:44:30 2012 -0800
more optimizations...
calculate the offsets from the phase differently, this happens
to reduce the register pressure in the main loop, which in turns
allows the compiler to generate much better code (doesn't need
to spill a lot of stuff on the stack).
this gives another 15% performance increase
Change-Id: I2ce3479dd48b9e6941adb80e6d443d6e14d64d96
commit
5a951598f31217b8cd2babd0720c9608ee17291a
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 10 03:26:39 2012 -0800
refactor code to improve neon code
we want to make sure we don't transfer data from the
neon unit to the arm register file, as this can be quite
slow. instead we do all the calculation on the neon side
and write the result directly to main memory.
Change-Id: Ibb56664d3ab03098ae2798b75e2b6927ac900187
commit
b381ee9e83bc9fd18986e79c7809841514ed590e
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 15:16:13 2012 -0800
NEON optimized SINC resampler
this currently gives us a 60% to 80% boost depending
on the quality level selected.
Change-Id: I7db385007e811ed7bffe5fd3403b44e300894f5b
commit
bea077354210242ea193a50b0dbab0fedab25df3
Author: Mathias Agopian <mathias@google.com>
Date: Mon Nov 5 01:51:37 2012 -0800
minor cleanups
Change-Id: Ia12ee4fb59e90221761bec85e6450db29197591f
commit
8f4ed7decbe161a5ff38200b218f5216d80aba46
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 18:49:14 2012 -0800
improve resample test
- handle stereo input
- input file can now be ommited, in this case
a linear chirp will be used automatically
- better usage information
Change-Id: I5d62a6c26a9054a1c1a517a065b4df5a2cdcda22
commit
5fcd634ea6cb4df27c495abe20f5f9b8ff55d128
Author: Mathias Agopian <mathias@google.com>
Date: Sun Nov 4 02:03:49 2012 -0800
change how we store the FIR coefficients
The coefficient table is now transposed and shows
much better its polyphase nature: we now have a FIR
per line, each line corresponding to a phase.
This doesn't change at all the results produced by
the filter, but allows us to make slightly better
use of the data cache and improves performance a bit
(although not as much as I thought it would).
The main benefit is that it is the first step
before we can make much larger optimizations
(like using NEON).
Change-Id: Iebf7695825dcbd41f25861efcaefbaa3365ecb43
commit
d652231abf4c7e2ea1fc89caae730cec1f7259a1
Author: Mathias Agopian <mathias@google.com>
Date: Sat Nov 3 23:37:53 2012 -0700
improve SINC resampler performance
The improvement is about 60% by just tweaking a few
things to help the compiler generate better code.
It turns out that inlining too much stuff manually was hurting us.
Change-Id: I8068f0f75051f95ac600e50ce552572dd1e8c304
commit
9dc68ef5b94c700c4ee68790e8cbb334c90a538d
Author: Mathias Agopian <mathias@google.com>
Date: Thu Nov 1 21:03:46 2012 -0700
new coefficients for the vhq resampler
previous coefficients were provided by a 3rd party and didn't have a
way to re-generate them. we're now using the 'fir' utility.
the performance of the filter is virtually identical, except for
the down-sampling case which seems slightly better now:
It looks like both the previous and new coefficients are generating
some sort of clipping for full-scale signals in the down-sampling case
(although the new ones seem better), the reason for that is
unknown (see bug:
7453062)
Also updated the HQ coefficients for the down-samplers, previous ones
were a little bit too conservative -- the new ones push the cut-off
frequency up by about 1 KHz.
Change-Id: I54a827b5c707c7cc41268ed01283758dce1d7647
commit
38e0b8560a6fc1b7124e22e0e09a84a285182f8e
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 13:51:44 2012 -0700
fix SINC resampler on non ARM architectures
make sure the C version of the code generates the same
output than the ARM assemply version.
Change-Id: Ide218785c35d02598b2d7278e646b1b178148698
commit
a1878128b182696ba508569b4d211d0dfae92463
Author: Mathias Agopian <mathias@google.com>
Date: Tue Oct 30 12:49:07 2012 -0700
fix another issue with generating FIR coefficients
the impulse response of a low-pass is 2*f*sinc(2*pi*f*k), we were
missing the 2*f scale factor. This explains why we were seeing
clipping and had to manually scale the filter down.
Change-Id: I86d0bb82ecdd99681c8ba5a8112a8257bf6f0186
commit
1a0fb993430acc9f601e6c538305bc407c20ac5d
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:20 2012 -0700
fir a typo that caused up-sampling coefficiens to be wrong
up-sample coefficient were generated with a cut-off frequency of 24KHz
intead of ~20KHz, which caused more aliasing in the audible band.
also increased the attenuation to 1.3 dB on both up and down
sampling coefficient to avoid clipping.
Change-Id: Ie8aeecf1429190541b656810c6716b6aae5ece2e
commit
9520ad6862bd682ad075a9d9e3e94ada9f6e58b6
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 17:13:16 2012 -0700
test-resample: clip instead of overflowing
Change-Id: I550e5a59e51c11e1095ca338222b094f92b96878
commit
ba36656300f250f7f1fdeb75149749344260e6cb
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 01:01:38 2012 -0700
a test app for the resamplers
Change-Id: I66852d90d384f1d9e77b51ad1a1ebdbaf61d0607
commit
056a08b9bfd33cf27228c992adc8293a56b01be8
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 14:11:01 2012 -0700
reenable the cubic resampler
cubic resampler was disabled because it hadn't been qualified,
however after I did some tests, it does improve significantly
the sound quality over the order-1 resampler, even if it is
still quite bad.
also HIGH_QUALITY resampler was partially disabled, it's now
fully enabled. It's a big improvement over the cubic resampler
in terms of aliasing noise (it's not as good in the pass-band).
Change-Id: I70e3658c255896588642697be9eb594ff4ec0f8b
commit
8c0241d3ff50ae85167f69b3bd369244894cfa44
Author: Mathias Agopian <mathias@google.com>
Date: Fri Oct 26 13:48:42 2012 -0700
improve SINC resampler coefficients
- we increase the interpolation precision from 4 to 7 bits
this doesn't increase CPU power required, it only increases the
size of the filter table but significantly reduces the noise
introduced by the quantization of the impulse response.
- the parameters of the filter are set such that aliasing is
rejected at 80 dB below 20 KHz. Because we don't use a lot of
coefficient (to save compute power), there are quite a bit of
attenuation in the pass-band: starting at 9KHz for the
down-sampler (48 to 44.1), and starting at 13 KHz for the
up-sampler (44.1 to 48) -- the transition band is about 15 KHz.
Change-Id: I855548d2aab8a0fb0d2a2da3a364b6842d7d3838
commit
69e7dab2192adc1f780464146810629ebd01b145
Author: Pixelflinger <mathias.agopian@gmail.com>
Date: Thu Oct 25 19:43:49 2012 -0700
improve fir tool: cleanup, better default, bug fixes
- all parameters can be changed on the command-line
- added float output
- added debug option
- added an option to generate a polyphase filter coefficients
- added an attenuation option in dBFS
- added a lot of comments and references
- fixed kaiser window parameter
also the default should generate a filter with 80 dB rejection
(of the 24 KHz aliasing) above 20 KHz and a 15 KHz transition
band around ~20 KHz (for 48 KHz sampling rate).
It's not very good but corresponds to the current code.
commit
8347499d105a50257c18e9dac652e750b06428b1
Author: Glenn Kasten <gkasten@google.com>
Date: Mon Oct 22 17:09:27 2012 -0700
Increase allowed number of VHQ resamplers to 3
Bug:
7378660
Change-Id: I69e33ca2eb4bb9bd38e2c63df62cd1130d68baf6
commit
f91cf3cad7f5c4d52614271c89ab468741c5d24c
Author: Mathias Agopian <mathias@google.com>
Date: Sun Oct 21 03:04:05 2012 -0700
Fix a typo that caused the high quality resampler to produce garbage
the problem is that if libaudio_resampler is present, it is those
coefficients that will always be selected, but the correct
meta-data.
Bug:
7385994
Change-Id: Ieebeb37b4dfb62a1a051bc29fae2ce056dbc6621
commit
e158a9e4262a174c59469a205658bc3ca4078234
Author: Dan Bornstein <danfuzz@google.com>
Date: Fri Oct 3 10:34:57 2008 -0700
Manually merge change #111620 from tc3 to mainline, to keep the
automerger from choking on it.
p4 sync
p4 integrate -r -b android_to_tc3 //...@111620,111620
p4 resolve -a
p4 resolve # resolve a couple merge travesties
PRESUBMIT=passed
BUG=
1399648
TBR=edheyl
OCL=111902
Change-Id: I854b01553dd92bbf9c864f5a9bd51a3d665f0ac2
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit
b9f3c26032be7a6ea01a10d93d94826f449e68ab
Author: Dave Bort <dbort@google.com>
Date: Fri Jan 18 14:51:05 2008 -0800
Rename "Makefile" to "Android.mk" throughout the tree.
For <http://b/issue?id=960416>.
I've tested this as much as I can, but 1500 open files =
easy to mess things up. Please let me know if there's
a problem rather than rolling back this change.
PRESUBMIT=passed
BUG=960416
TBR=joeo
OCL=46537
Change-Id: I5a404caf0f398a7afa7ae7abaf2f2a1c6ab490eb
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit
0c22a9a44c4103483fba1d944acf1354c5eb1617
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 23:44:25 2007 -0700
Tweak the SINC resampler parameters and double the performance. It's using about 10% CPU in the worse case now.
Change-Id: I50ac7e6c6702a427fa36ab6d976c507155057507
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit
b85e41487983ad085b859acf8251e7e54480308a
Author: Mathias Agopian <mathias@google.com>
Date: Mon Oct 29 04:34:36 2007 -0700
A sinc resampler for Audioflinger. It's not enabled yet, but fully functional and apparently working. It need more "quality" tests. In the 48->44 KHz, it takes about 25% of the CPU time.
Change-Id: I80eb5185e13ebdb907e0b85c49ba1272c23d60ec
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit
ba3949ef17cac2ba71cc3096c413782a49c922e5
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 21:01:28 2007 -0700
fix a few small typos in the FIR computation
Change-Id: I6e56b514fe520f30f7487f85c64ea5d2a7c19b40
Signed-off-by: Glenn Kasten <gkasten@google.com>
commit
7474bfa7de2604021963794dddfe44985648db6a
Author: Mathias Agopian <mathias@google.com>
Date: Thu Aug 23 03:16:02 2007 -0700
This is a tool to compute the the reconstruction filter coefficients for a sinc audio resampler.
Change-Id: I99be2505139b8e0e7647200e1647509d4f7e6067
Signed-off-by: Glenn Kasten <gkasten@google.com>
Bug:
7577965
Change-Id: I2c84a9283a1668723bad83e1a119c849c88c3e6b
Igor Murashkin [Tue, 27 Nov 2012 19:39:22 +0000 (11:39 -0800)]
am
aa9e3e01: Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
* commit '
aa9e3e01b86bd9bfb5ac36c0f360d5fe478cbb2d':
Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
Andreas Huber [Tue, 27 Nov 2012 17:39:12 +0000 (09:39 -0800)]
am
5768fa03: Merge "Crash in android::MyHandler::parsePlayResponse"
* commit '
5768fa034ede834656697d3612c525595ff85ef9':
Crash in android::MyHandler::parsePlayResponse
Andreas Huber [Tue, 27 Nov 2012 17:39:11 +0000 (09:39 -0800)]
am
3eb46d17: Merge "Fix of uninitialized mIsDrm variable."
* commit '
3eb46d179b1f62cde21077fde466925d4c5c79ad':
Fix of uninitialized mIsDrm variable.
Andreas Huber [Tue, 27 Nov 2012 16:50:52 +0000 (08:50 -0800)]
Merge "Crash in android::MyHandler::parsePlayResponse"
Andreas Huber [Tue, 27 Nov 2012 16:49:34 +0000 (08:49 -0800)]
Merge "Fix of uninitialized mIsDrm variable."
Patric Frederiksen [Mon, 26 Sep 2011 08:51:35 +0000 (10:51 +0200)]
Crash in android::MyHandler::parsePlayResponse
This fix handles problems with several asynchronous calls
within streaming. This case is when the phone has sent a
request to the server and while the response is being sent
back by the server the request is aborted by the user.
The fix is an if case that checks if we have aborted while
waiting for a response from the server. If we have aborted
we should ignore the late response instead of continuing.
Change-Id: I1264bb992f6abcaee1f10a89479e08b54a95e3c2
Henrik B Andersson [Wed, 31 Oct 2012 12:02:47 +0000 (13:02 +0100)]
Fix of uninitialized mIsDrm variable.
The mIsDrm is a bool that isn't initialized.
This causes it to be true in most default cases.
Change-Id: I41b534514bf6a3ca88a9f0994b814d55fcd7453b
Igor Murashkin [Mon, 26 Nov 2012 18:50:55 +0000 (10:50 -0800)]
Camera: Play shutter sound iff enableShutterSound(true) && ShutterCallback !null
Bug:
7564718
Change-Id: Ie7821cdee57966d88af048759578439a3e6ecb2e
Andreas Huber [Mon, 26 Nov 2012 22:37:22 +0000 (14:37 -0800)]
am
79fd6853: Merge "MediaCodec: Add a method for getting the component name"
* commit '
79fd685323e34e0fde22d17fd6848d33f171f4ae':
MediaCodec: Add a method for getting the component name
Andreas Huber [Mon, 26 Nov 2012 21:48:41 +0000 (13:48 -0800)]
Merge "MediaCodec: Add a method for getting the component name"
Marco Nelissen [Mon, 26 Nov 2012 18:00:09 +0000 (10:00 -0800)]
am
5865ddf7: Merge "AudioTrack::dump null mCblk check test"
* commit '
5865ddf769d368d714af630aba18392ea1387bc7':
AudioTrack::dump null mCblk check test
Marco Nelissen [Mon, 26 Nov 2012 16:59:50 +0000 (08:59 -0800)]
Merge "AudioTrack::dump null mCblk check test"
Zbigniew Mazur [Thu, 11 Oct 2012 11:56:41 +0000 (13:56 +0200)]
AudioTrack::dump null mCblk check test
This fix is protecting AudioTrack::dump from SIGSEGEV
when calling dumpsys shell command.
Change-Id: I30d136e195a12d6fdad41e14f557b0ad9e59b8a2
Marco Nelissen [Tue, 20 Nov 2012 20:47:56 +0000 (12:47 -0800)]
am
2da6e4ae: Merge "Avoid memory leaks when handling metadata strings"
* commit '
2da6e4ae488896df10b22166b0aa0b2cc15492f1':
Avoid memory leaks when handling metadata strings
Marco Nelissen [Tue, 20 Nov 2012 20:28:27 +0000 (12:28 -0800)]
Merge "Avoid memory leaks when handling metadata strings"
David Williams [Mon, 19 Nov 2012 08:52:16 +0000 (09:52 +0100)]
Avoid memory leaks when handling metadata strings
Don't duplicate strings when retrieveing metadata from media
files. As any requests for metadata strings would pass through
the binder, this would cause the reference to the duplicate string
to be lost, causing a memory leak as the duplicate would not be
freed.
Change-Id: I2593733472b1bb589bc502b2c11080f581015bb5
Andreas Huber [Tue, 20 Nov 2012 17:24:25 +0000 (09:24 -0800)]
am
7013209c: Merge "Handle large AVCC chunks"
* commit '
7013209cdf393b3d958ddd46ed50394349378826':
Handle large AVCC chunks
Marco Nelissen [Tue, 20 Nov 2012 17:24:24 +0000 (09:24 -0800)]
am
dca2b5d7: Merge "Changed parsing of trkn and disk from 8 bits to 16 bits"
* commit '
dca2b5d7c29ee06f3c82527dd7264fcc21cac9a6':
Changed parsing of trkn and disk from 8 bits to 16 bits
Andreas Huber [Tue, 20 Nov 2012 16:52:44 +0000 (08:52 -0800)]
Merge "Handle large AVCC chunks"
Marco Nelissen [Tue, 20 Nov 2012 16:18:38 +0000 (08:18 -0800)]
Merge "Changed parsing of trkn and disk from 8 bits to 16 bits"
Martin Storsjo [Tue, 25 Sep 2012 08:43:02 +0000 (11:43 +0300)]
MediaCodec: Add a method for getting the component name
If the codec was chosen based on mime type, the caller does
not know what component actually was chosen. This allows
getting essential information (such as supported color formats,
for a video encoder) for this component.
Change-Id: Ie471f40f8104b37d27ced3dba5a54facc6504b1b
Andreas Lillvik [Wed, 13 Oct 2010 13:37:01 +0000 (15:37 +0200)]
Changed parsing of trkn and disk from 8 bits to 16 bits
The MPEG4Extractor was parsing 8 bits instead of 16 bits when parsing
'trkn' and 'disk'. Also added support for 16 bytes size 'disk'.
Change-Id: I22b4de2ac800881884d5759776cb380917522a87
Jan Olof Svensson [Wed, 26 Sep 2012 07:08:11 +0000 (09:08 +0200)]
Handle large AVCC chunks
If enabling seq_scaling_matrix_present_flag = 1 the AVCC chunk can
be larger than the original buffer size. Changed to using ABuffer
instead.
Change-Id: Idacc14b45ea2634c5e608919f3ce567f23363135
Jens Gulin [Mon, 10 Sep 2012 13:54:40 +0000 (15:54 +0200)]
Memory leak solved in MPEG4Extractor::parseChunk for cover art
Repeated leaks detected in mediaserver context. Most chunks small but
some bigger. Adding up it was not uncommon to see 200Mb footprint.
libc memdebug allocation call stack showed the root cause in "covr"
parsing - that explains why content often was empty but sometimes big.
Change-Id: I0de6c1eeef174d2529973ff9c6a020ec3dd44c75
Erik Rydgren [Mon, 19 Sep 2011 08:38:10 +0000 (10:38 +0200)]
Use default values when MPEG4 audio config parsing fails.
MPEG4 audio packets may be multiplexed using the so called
LATM (Low Overhead Audio Transport Multiplex) scheme.
LATM parsing was recently introduced in Stagefright and it
has caused issues in cases when the LATM config element
cannot be parsed correctly. The main problem occurrs when
the AudioSpecificConfig part of the config element contains
more information than what is expected, causing the
frameLengthType parameter to get the wrong value. This fix
introduces default values of some config parameters that are
used in case config parsing fails.
Change-Id: I3cb35df76826f95ca0831dc08c2a1e7c6c2c586d
Glenn Kasten [Fri, 16 Nov 2012 22:44:05 +0000 (14:44 -0800)]
am
d983364b: Static AudioTrack plays twice initially
* commit '
d983364b3655a547b55bb11dbe148103198c011d':
Static AudioTrack plays twice initially
Glenn Kasten [Thu, 15 Nov 2012 22:13:16 +0000 (14:13 -0800)]
Static AudioTrack plays twice initially
Bug:
7528721
Change-Id: I10bc16a26f33dba6572b730a170cb3bf00e68e30
Marco Nelissen [Fri, 16 Nov 2012 17:34:19 +0000 (09:34 -0800)]
am
da33d66e: Merge "Add .mpga to acceptable file name extensions list."
* commit '
da33d66e68791d0bfeccebc8253a59467b5ef670':
Add .mpga to acceptable file name extensions list.
Marco Nelissen [Fri, 16 Nov 2012 16:59:14 +0000 (08:59 -0800)]
Merge "Add .mpga to acceptable file name extensions list."
Jan Bjernler [Fri, 16 Nov 2012 15:40:42 +0000 (16:40 +0100)]
Add .mpga to acceptable file name extensions list.
The *.mpga files are playable, but are not correctly scanned.
This is because they are prevented from being scanned in
StagefrightMediaScanner.cpp.
What this fix does is to add the extension to the list of valid
file extensions so that the scanner handles the filetype properly.
We have previously added the .mpga extension to the framework to
make it playable, but not added it so that the scanner scans it.
Change-Id: I02a44d770adb80d38e8bed77d0d76efa1b28a861
Patrik2 Carlsson [Fri, 16 Nov 2012 15:34:33 +0000 (16:34 +0100)]
Allow multichannel FLAC files
FLACExtractor had artificial limits which did not allow multichannel
audio, 88.2 or 96 kHz samplerate, these were removed since it is up
to the audio subsystem to handle.
Change-Id: Iadc20cdf4724814b5c980827a528a1e1b4bd6a82
beanz [Fri, 16 Nov 2012 05:47:50 +0000 (13:47 +0800)]
libvideoeditor: fix cannot show some thumbnails in storyboard or no frame display in PreviewRenderer surface after importing video
1. this is caused mFrameIntervalMs isn't assigned for use-software-codec case
2. without the assingment, some decoded frames will not be copied to the buffer queue of VideoEditorVideoDecoder
Change-Id: I578954e38958ebac8f258300e60b8ddf60e50521
Signed-off-by: beanz <beanz@marvell.com>
Andreas Huber [Wed, 14 Nov 2012 19:49:55 +0000 (11:49 -0800)]
am
f0937247: Merge "Properly signal an error if codec configuration goes wrong." into jb-mr1.1-dev
* commit '
f0937247b6d92b7d9457c64e36fe4c10927685ac':
Properly signal an error if codec configuration goes wrong.
Andreas Huber [Wed, 14 Nov 2012 19:49:54 +0000 (11:49 -0800)]
am
0224bf17: Various improvements of wifi display code
* commit '
0224bf170a3904576bba81593eaab113c5d3a4e7':
Various improvements of wifi display code
Andreas Huber [Wed, 14 Nov 2012 19:45:51 +0000 (11:45 -0800)]
Merge "Properly signal an error if codec configuration goes wrong." into jb-mr1.1-dev
Andreas Huber [Wed, 14 Nov 2012 17:06:33 +0000 (09:06 -0800)]
Properly signal an error if codec configuration goes wrong.
previously any error signaled by setupXXX inside ACodec::configureCodec
would be overwritten with the result of setMinBufferSize at the end
of the function.
Change-Id: Id4beb571ca52ea4646239d0af006e09ce1130268
related-to-bug:
7542181
Andreas Huber [Mon, 12 Nov 2012 21:08:44 +0000 (13:08 -0800)]
Various improvements of wifi display code
- manually prepend SPS/PPS if encoder doesn't support it
- latency improvements
- support for "our" method of optional RTP retransmission
- improvements to the wfd commandline tool for testing
- make it easier to turn on/off suspension of the video pipeline on idle
- fixes an issue where an error during encryption would cause a SEGV
- add HDCP descriptor if necessary
Squashed commit of the following:
commit
1115be0ebb3b885b4f1b7dba56761ca013d0ec4a
Author: Andreas Huber <andih@google.com>
Date: Fri Nov 9 11:32:23 2012 -0800
Better shutdown of wfd -l sessions.
Change-Id: Id898a14ae21efd3b065b00a729830063d39195a7
commit
0e7d106dfe4eb6e2640b0b66c65deaba265f7ff0
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 16:38:55 2012 -0800
No more sending delay, create rtp packets upfront.
Change-Id: I809a225f664fdb485c7d9a49a27886601a6a26b2
commit
d399e8571b77353d59afb57508dfd2a82c1ef93a
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 14:19:43 2012 -0800
Restore AudioSource buffer size, factor out TimeSeries, make
suspending video optional.
Change-Id: Ifdfe4d447b901e714abf52856b4641d1d55a5d41
commit
f8b649f0b8f917d59f4b8a2e8e6d7db61a684a78
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 09:34:06 2012 -0800
Pull 480 frames at a time from AudioSource/AudioRecord
Change-Id: I1e215abd329faec3da026631122c0f4c800c0ac4
commit
1bc13452eb35eebbba00f5da93fa86535be5db59
Author: Andreas Huber <andih@google.com>
Date: Thu Nov 8 08:50:30 2012 -0800
fixed bitrate traffic simulation
Change-Id: Ic5efb7cbb0b5d3b4917bc77b8ba73d447249e695
commit
016cdff18e74bdc631a5679e97192645ed095aa2
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 7 14:00:03 2012 -0800
resurrected "our" style of retransmission.
Change-Id: I34d757aba67428437cb39b8293a9651750ad20d9
commit
384cf1a3c8fb4ec410bdf8fa5722c298e6028f3e
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 6 09:38:55 2012 -0800
Changes to make wfd work on manta.
Change-Id: I7a4e00cf16581fe2146edd1b359af195774090e4
commit
9628f24b22b35f28630d99dda3614babf51bc07e
Author: Andreas Huber <andih@google.com>
Date: Wed Nov 7 09:15:44 2012 -0800
Patch up rtp timestamps to more accurately measure network jitter.
Change-Id: I9502a4615575f97f98a215a13131a89a6ae93c6d
commit
7c891a1a24f08bbd50f55be13f7d05f43e807eb8
Author: Andreas Huber <andih@google.com>
Date: Tue Nov 6 09:37:24 2012 -0800
Additions to the "wfd" tool to create a local wfd source.
Change-Id: I99558653a70fdc703f9d13990b3ce1c4d3ae227a
Change-Id: Ia94c63fc390f597014531073485f0cfc53b3418a
beanz [Wed, 7 Nov 2012 02:58:59 +0000 (10:58 +0800)]
libvideoeditor: fix code bugs when enabling all-level debugging macros
Change-Id: Ic8aedfa821d79698d31cf7ee3bf382aae85981f6
Signed-off-by: beanz <beanz@marvell.com>
The Android Open Source Project [Tue, 6 Nov 2012 20:24:38 +0000 (12:24 -0800)]
am
78b012f5: Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
* commit '
78b012f5bc05706d1ce412811ccfaa9385dae0ab':
The Android Open Source Project [Tue, 6 Nov 2012 20:16:54 +0000 (12:16 -0800)]
Reconcile with jb-mr1-factory-release jb-mr1-release - do not merge
Change-Id: Idf4c25115d89397ba668fc290344b2e7c1ea1993
Glenn Kasten [Tue, 6 Nov 2012 19:06:20 +0000 (11:06 -0800)]
am
1c6833af: Ignore SIGPIPE during write() to broken pipe
* commit '
1c6833afd6cfdbc6c696ed0719ceedc37ec0b534':
Ignore SIGPIPE during write() to broken pipe
Glenn Kasten [Wed, 17 Oct 2012 19:16:50 +0000 (12:16 -0700)]
Ignore SIGPIPE during write() to broken pipe
This happens occasionally when taking a bugreport.
Bug:
6447319
Change-Id: Ia6531a4a3658461f8fd3f7106e7996da7cc5933a