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Author SHA1 Message Date
andrew@webrtc.org
be8e8ee6f6 Remove bad *s from filename. 
Appeared to be causing an error on the Windows bots:
svn: Can't check path
'E:\b\build\slave\win\build\src\samples\js\demos\html\****THESE_FILES_ARE_MOVING****':
The filename, directory name, or volume label syntax is incorrect.

TBR=dutton@google.com

Review URL: https://webrtc-codereview.appspot.com/11069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5840 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 20:51:41 +00:00
kjellander@webrtc.org
c7b8b2f2a7 PRESUBMIT.py: use new way to specify default try builders 
In https://codereview.chromium.org/178223016 and
https://codereview.chromium.org/197963003 the way the
PRESUBMIT.py specifies the default try builders for a
try job have changed.

When submitting a try job now, the test filter argument no
longer works unless --bot is also specified.
This CL attempts to resolve this by moving away from the
deprecated approach onto using the new format instead.

This CL also includes two new trybots: win_asan and linux_tsan2
(added in https://codereview.chromium.org/220453004).

BUG=3148
TEST=Successfully fired off a -t compile job where the
test filter worked.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5839 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 20:19:36 +00:00
dutton@google.com
fe165ded46 Added warning for Github move ****THESE_FILES_ARE_MOVING**** 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5837 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 19:57:06 +00:00
bjornv@webrtc.org
240eec3cd4 Delay Estimator: Minor refactoring and added a setter function. 
* Replaced the lookahead input parameter at Create() with a setter. This makes it slightly more user friendly.
* Changed the buffer shifting in SoftReset... to become more readable.

TESTED=trybots, modules_unittests
R=aluebs@webrtc.org, andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-03 08:11:47 +00:00
wu@webrtc.org
148149138d (Auto)update libjingle 64147530-> 64247466 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5835 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 23:25:15 +00:00
wu@webrtc.org
5e760e7b94 Check the return value of the FromString call and return failure when then value is invalid. I.e. uses 
bool FromString(const std::string& s, T* t)
instead of
T FromString(const std::string& str)

Before this change we will silently continue the parsing and take whatever default value returned by FromString.

TEST=new tests
BUG=2507
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5834 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 23:19:09 +00:00
wu@webrtc.org
e387771b98 Remove webrtc_unittest.cc from talk presubmit script. 
BUG=
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5833 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 22:23:16 +00:00
henrik.lundin@webrtc.org
184b913eb5 Rename RTPanalyze to rtp_analyze and remove old version 
The tool RTPanalyze (used to process an input RTP dump into a
text file of RTP header info) was present in both the neteq and
neteq4 folders. This change pulls in changes from the old to the new
and renames the source file and tool to rtp_analyze.

Removing special code for dummy-rtp files (it is supported without
special code), and making the RED payload type settable using flags.

Moving from test/ to tools/ folder.

BUG=2692
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 20:56:17 +00:00
andrew@webrtc.org
c7c432aa9b Remove AudioDevice::{Microphone,Speaker}IsAvailable. 
This was only used for logging, except on Mac, where the methods are
now private.

BUG=3132
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 16:49:26 +00:00
minyue@webrtc.org
7549ff4257 This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets. 
BUG=3140
TEST=trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5830 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 15:03:01 +00:00
henrik.lundin@webrtc.org
1092ea0192 Add format specification to output file names 
This change facilitates running ApmTest.VerifyDebugDumpInt and
ApmTest.VerifyDebugDumpFloat in parallel, since they are not writing
to the same files any longer.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 07:46:49 +00:00
henrika@webrtc.org
620d444c0b Extends max sample rate from 96kHz to 192kHz on the input side. 
TEST=apprtc in Chrome using this WebRTC version and a device on Windows which can capture at 192kHz
BUG=725
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 07:22:34 +00:00
braveyao@webrtc.org
790385fee4 sink_filter_ds.cc: add lock to Receive procedure to Pause(). 
BUG=2233
TEST=AUTO Test
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5827 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-02 02:14:55 +00:00
andrew@webrtc.org
19018ddb17 Make ACM2 the default in voe_cmd_test. 
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 20:58:05 +00:00
wu@webrtc.org
05e7b44b83 (Auto)update libjingle 63948945-> 64147530 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5825 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 17:44:24 +00:00
stefan@webrtc.org
f8f7c8b618 Added simulations of capacity variations and wifi recordings. 
Also changes the packet sizes for the video sender and the trace based filter to match.

R=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 14:00:05 +00:00
kjellander@webrtc.org
7e889b7126 Add /third_party/syzygy/binaries to .gitignore 
This should have been done in
https://webrtc-codereview.appspot.com/2381004

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5823 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 13:46:48 +00:00
kjellander@webrtc.org
d10bdd3f78 Roll chromium_revision 255773:260462 
This disables GN use for the moment (Chromium
has disabled it for now but plan to pick up the
work at a later stage). I'm leaving the rest of
the GN stuff in our DEPS since that's how
the Chromium DEPS currently looks like.

Overview of changes in Chrome DEPS:
$ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 255773:260462

which can be compared with the output of:
$ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq

in a WebRTC checkout, gives the following relevant changes:
* third_party/android_tools 0582bd:ca3567
* third_party/icu 249466:259309
* third_party/libjpeg_turbo 251747:259851
* third_party/libyuv 979:986
* third_party/nss 254867:259440
* tools/gyp 1860:1880

The following also shows that Clang is upgraded from r198389 to r202554:
$ svn diff http://src.chromium.org/chrome/trunk/src/tools/clang/scripts/update.sh -r 255773:260462

TEST=trybots
BUG=None
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 10:40:03 +00:00
andrew@webrtc.org
ca9d038ac8 Fix ARM64 detection. 
Use only __aarch64__ and don't look for __arm64__ at all.
It turns out that clang defines both and GCC only the former.
Hence, looking only for __aarch64__ should be safe.

BUG=chromium:354405,chromium:358092
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10939004

Patch from Primiano Tucci <primiano@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5821 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 01:19:08 +00:00
fischman@webrtc.org
a789f3720a VoiceEngine(iOS & Android): removed NOT_SUPPORTED 
Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
  bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds

BUG=2050,3132
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-01 00:16:35 +00:00
fbarchard@google.com
8f8119409d Roll libyuv to 994 for arm64 initial support using C versions of code. 
BUG=chromium:354539
TESTED=GYP_DEFINES="OS=ios target_arch=armv7 target_subarch=64" GYP_CROSSCOMPILE=1 GYP_GENERATOR_FLAGS="output_dir=out_ios" ./build/gyp_chromium -f ninja --depth=. libyuv_test.gyp  && ninja -j7 -C out_ios/Debug-iphoneos
R=andrew@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/10929005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 21:35:01 +00:00
fischman@webrtc.org
49c5ba32bb AppRTCDemo(iOS): now works in the iOS Simulator! 
...which has no camera device emulation or pass-through, so no local video
view.

R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 20:22:19 +00:00
fischman@webrtc.org
61e78fca6c AppRTCDemo(iOS): remote-video reliability fixes 
Previously GAE Channel callbacks would be handled by JS string-encoding the
payload into a URL.  Unfortunately this is limited to the (undocumented,
silently problematic) maximum URL length UIWebView supports.  Replaced this
scheme by a notification from JS to ObjC and a getter from ObjC to JS (which
happens out-of-line to avoid worrying about UIWebView's re-entrancy, or lack
thereof).  Part of this change also moved from a combination of: JSON,
URL-escaping, and ad-hoc :-separated values to simply JSON.

Also incidentally:
- Removed outdated TODO about onRenegotiationNeeded, which is unneeded
- Move handling of PeerConnection callbacks to the main queue to avoid having
  to think about concurrency too hard.
- Replaced a bunch of NSOrderedSame with isEqualToString for clearer code and
  not having to worry about the fact that [nil compare:@"foo"]==NSOrderedSame
  is always true (yay ObjC!).
- Auto-scroll messages view.

BUG=3117
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/10899006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 20:16:49 +00:00
fbarchard@google.com
30cd5b5278 libyuv roll to r986 for c89 fix to cpu_id. 
BUG=none
TESTED=cl cpu_id.cc
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 17:28:46 +00:00
solenberg@webrtc.org
caeae4680c Add tests for the RBE RemoveStream() API. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 13:33:39 +00:00
henrik.lundin@webrtc.org
d0a81d91ff VoE Channel: Don't register codecs when stopping receiver 
VoiceEngine's Channel::StopReceiving() would call
RegisterReceiveCodecsToRTPModule(), which caused some errors
with RED and ULP-FEC. In particular, an error message would be
printed when hanging up a call in voe_cmd_test application.

BUG=3085
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-31 07:31:53 +00:00
fischman@webrtc.org
fe16488184 AppRTCDemo(android): specify DtlsSrtpKeyAgreement:true in CreatePeerConnection's constraints. 
This is required to interop with Chrome now that SDES is disabled in Chrome (as of r5640).

BUG=2774
R=jiayl@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 19:58:03 +00:00
fischman@webrtc.org
4f2bd68744 Silence pointless LS_WARNING about port 0 for active-only candidates. 
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 18:13:34 +00:00
wu@webrtc.org
987f2c9aae (Auto)update libjingle 63913264-> 63948945 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5807 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 16:22:19 +00:00
kjellander@webrtc.org
0aa04f9f24 Restore support for code coverage in WebRTC 
In https://codereview.chromium.org/68193002
Chromium dropped the support for the coverage=1 flag.
This restores it for WebRTC purposes for the Linux platform.

TEST=Manually ran the coverage steps on my machine, verified
that .gcno files are generated.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-28 13:14:00 +00:00
wu@webrtc.org
f7d501d48a (Auto)update libjingle 63884381-> 63913264 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5805 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 23:48:25 +00:00
andrew@webrtc.org
a5586b50e5 Protect ENABLE_PROFILING to fix profiling=1. 
Chromium defines ENABLE_PROFILING under the gyp flag profiling=1. This
corrects the resulting mulitple defintion error:
../../talk/base/profiler.h:61:9: error: 'ENABLE_PROFILING' macro redefined [-Werror]
#define ENABLE_PROFILING

and allows us to use profiling=1 in standalone builds.

TESTED=build passes with profiling=1
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5804 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 22:44:13 +00:00
fischman@webrtc.org
dd0b99debb Roll libvpx 258445:259973. 
- 259973: unbreak iOS simulator build (-mssse3)
- 259953: add a missing file (follow-up to r259946)
- 259946: Disable assembly optimizations in MemorySanitizer builds.
- 259324: disable function level linking when building vp8_asm_enc_offsets.c

BUG=3126
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10829005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5803 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 22:27:54 +00:00
andrew@webrtc.org
fff3fd35a6 Add arm64 to typedefs.h 
This is the first step to get a buildable chrome_shell_apk for arm64.

BUG=chromium:354405
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10819004

Patch from Primiano Tucci <primiano@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 19:48:53 +00:00
andresp@webrtc.org
5a0218c794 Allow loopback tests to do TURN when served from webrtc.googlecode.com. 
BUG=3037
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 19:24:45 +00:00
wu@webrtc.org
cfe5e9c894 (Auto)update libjingle 63837929-> 63884381 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5800 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 17:03:58 +00:00
andresp@webrtc.org
6b17be0bf8 Add svn mime-type properties to loopback_test files so they can be served from: 
https://webrtc.googlecode.com/svn/trunk/webrtc/tools/loopback_test/loopback_test.html


git-svn-id: http://webrtc.googlecode.com/svn/trunk@5799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 10:52:09 +00:00
andrew@webrtc.org
b13a7d5b1c Don't disable experimental AGC in audioproc. 
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5798 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-27 00:11:11 +00:00
henrike@webrtc.org
b0ecc1c6fb (Auto)update libjingle 63777286-> 63837929 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 22:44:28 +00:00
andrew@webrtc.org
b6dfbed1dc Exclude TwoStreamsSendAndFailUnsignalledRecvInOneToOne from TSAN. 
Example failure:
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan/builds/1458

TBR=wu@webrtc.org
BUG=2380

Review URL: https://webrtc-codereview.appspot.com/10759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5796 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 22:22:46 +00:00
fischman@webrtc.org
b25576a75b talk/: enable _DEBUG in Debug for all posix 
Chromium's build/common.gypi defines _DEBUG for Debug builds _except_ on
(OS=="mac" OS=="ios").  But libjingle uses _DEBUG on all platforms so define it on all posix (chromium covers non-posix separately and fine).

BUG=webrtc:3101
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/10699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 21:53:47 +00:00
andresp@webrtc.org
44caf01c34 Re-submit: rev5775 
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.

Additionally:
 Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.

 Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).

 Did not touch decrease logic, however since it can be triggered more often it
 may decrease much faster and closer to the original written cap of once every
 300ms + rtt.

Note:
 rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
 bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.

BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5794 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 21:00:21 +00:00
henrike@webrtc.org
1ca08f65e3 Fix after auto update in r5787. APPRTCVideoView.h/m was removed incorrectly. 
BUG=3121
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5793 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 16:42:14 +00:00
jiayl@webrtc.org
7ee0c16edd Makes ScreenCapturerMac exclude the window specified in DesktopCapturer::SetExcludedWindow. 
No behavior change for now since Chromium has not been updated to call SetExcludedWindow.

BUG=2789
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/10299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 15:57:43 +00:00
solenberg@webrtc.org
4e65602886 Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:32:47 +00:00
andresp@webrtc.org
d09d074827 Protect write of send_target_bitrate. 
This issue was catch by tsan bot.

BUG=3065
R=stefan@webrtc.org, andrew

Review URL: https://webrtc-codereview.appspot.com/10619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:27:34 +00:00
henrike@webrtc.org
5fb7428496 (Auto)update libjingle 63775799-> 63776369 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 02:00:10 +00:00
henrike@webrtc.org
a92fd74f40 (Auto)update libjingle 63773382-> 63775799 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 01:46:18 +00:00
henrike@webrtc.org
dce3feb0b0 (Auto)update libjingle 63738002-> 63773382 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 01:17:30 +00:00
solenberg@webrtc.org
440fa23553 Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out. 
BUG=2954
R=mflodman@webrtc.org, stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 19:57:07 +00:00