buildbot@webrtc.org
8563ef448a
(Auto)update libjingle 69587333-> 69588608
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6501 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-20 00:13:01 +00:00
buildbot@webrtc.org
1ef789d455
(Auto)update libjingle 69568113-> 69587333
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6500 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 23:54:12 +00:00
jiayl@webrtc.org
594aefa807
Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver.
...
It's wasted work and affects frame rate adaptation in Chrome.
BUG=
R=sergeyu@chromium.org
Review URL: https://webrtc-codereview.appspot.com/19789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6499 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 22:04:41 +00:00
buildbot@webrtc.org
df9bbbee56
(Auto)update libjingle 69567902-> 69568113
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6498 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 19:54:33 +00:00
buildbot@webrtc.org
fbd13286dc
(Auto)update libjingle 69555283-> 69567902
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6497 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 19:50:55 +00:00
buildbot@webrtc.org
21794f9862
(Auto)update libjingle 69543894-> 69555283
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6496 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 17:14:19 +00:00
fgalligan@google.com
304ca76be1
Revert 6481 and 6482
...
Revert 6482 "Update webrtc to fix unpack_lib expansion."
Revert 6481 "Update generated asm offsets scripts."
The roll has not been successful. Reverted based on the request of the
committer.
TBR=turaj
Review URL: https://webrtc-codereview.appspot.com/17759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6495 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 17:08:46 +00:00
turaj@webrtc.org
8de8c9155e
Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow.
...
To save memory in iSAC-fix, decoder operated directly on the recieved bitstream. However, this breaks constantness of input when decoder performed in-place big to little Endian conversion. Furthermore, for bit-streams with odd lengths, this meant writing outside the memory. That is because the last byte will be shifted to the Most Significat Byte which might be outside the allocated memory.
If we care about memory, the solution is to do a big-to-little Endian conversion everytime we read a Word16 from the bitstream.
BUG=845,chrome:379458
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6494 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 15:47:09 +00:00
henrik.lundin@webrtc.org
9158df2aa4
Adding an empty constructor implementation to the AudioSink class
...
Turns out it was needed.
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6493 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 12:34:31 +00:00
bjornv@webrtc.org
84f8ec1f9c
Changes to tests and tools in audio_processing.
...
- Disables ApmTest.EchoCancellationReportsCorrectDelays
This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored.
- Adds flag to turn off reported_delay in audioproc
Now it is feasible to turn on and off the use of reported system delays.
BUG=N/A
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6492 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 12:14:33 +00:00
stefan@webrtc.org
077593b805
Ensure that the start bitrate can be set multiple times.
...
If the start bitrate is set twice, it will be set to the sum of the start
bitrates of the currently registered bitrate observers, or left unchanged
if the current estimate actually is greater than the sum.
BUG=3503
R=henrik.lundin@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6491 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 12:13:00 +00:00
henrik.lundin@webrtc.org
496a98463b
Adding test::AudioSink interface and derived classes
...
The AudioSink interface is supposed to be used by tests that produce
audio output. Two implementation classes are also provided:
OutputAudioFile: Writes the audio to a pcm file.
AudioChecksum: Calculates the MD5 checksum of the audio.
These will both be used in future changes.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 10:02:11 +00:00
bjornv@webrtc.org
5c3f4e3b0f
Fixes and re-enables tests disabled on Android
...
Several tests were disabled in r6325 and r6326. Also, see issue 3445. This CL fixes the remaining four of the audio_processing related ones. Affects the tests:
- SystemDelayTest.CorrectDelayAfterStableBufferBuildUp
- SystemDelayTest.CorrectDelayDuringDrift
- SystemDelayTest.ShouldRecoverAfterGlitch
- ApmTest.EchoCancellationReportsCorrectDelays
The tests assumes reported delays are used, which now is explicitly set.
BUG=3445
TESTED=trybots
R=aluebs@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6489 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 09:51:29 +00:00
buildbot@webrtc.org
d27d9ae644
(Auto)update libjingle 69506154-> 69515138
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6488 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 01:56:46 +00:00
jiayl@webrtc.org
6ce1d58613
Exclude flaky test PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate on memcheck.
...
TBR=wu@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/17739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6487 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-19 00:06:36 +00:00
jiayl@webrtc.org
acede34aea
Fix a memory leak in SctpDataMediaChannelTest.
...
BUG=3492
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 23:36:16 +00:00
jiayl@webrtc.org
85b19a1a12
Exclude SctpDataMediaChannelTest on Win DrMemory for third_party/usrsctp issues.
...
TBR=wu@webrtc.org
BUG=3492
Review URL: https://webrtc-codereview.appspot.com/14719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6485 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 23:34:18 +00:00
jiayl@webrtc.org
f8063d34de
Properly shut down the SCTP stack.
...
TBR phoglund@webrtc.org for the tsan_v2/suppressions.txt change.
R=ldixon@webrtc.org , pthatcher@webrtc.org
TBR=phoglund@webrtc.org
BUG=2749
Review URL: https://webrtc-codereview.appspot.com/12739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6484 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 21:30:40 +00:00
fgalligan@google.com
a19b930b5b
Update webrtc to fix unpack_lib expansion.
...
Add on fix for:https://webrtc-codereview.appspot.com/12789004/
*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.
BUG=377062
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6482 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 19:20:45 +00:00
fgalligan@google.com
8f06a8aeb0
Update generated asm offsets scripts.
...
Libvpx updated the unpack scripts to fix building dependencies.
Roll libvpx 269083:278063
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
https://codereview.chromium.org/325313007/
https://codereview.chromium.org/346563002/
for the libvpx changes.
See https://codereview.chromium.org/313243004/
for the WebView changes.
*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.
BUG=377062
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6481 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 17:38:08 +00:00
bjornv@webrtc.org
b947d954a5
Neon version of FilterAdaptation()
...
The performance gain on a Nexus 7 reported by audioproc is ~5.2%.
The output is bit exact.
Measured total of 15% speed gain on N7 compared to C.
R=bjornv@webrtc.org , cd@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17699004
Patch from Scott LaVarnway <slavarnw@gmail.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6480 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 14:55:49 +00:00
henrik.lundin@webrtc.org
12396aba42
Update PacketSource and RtpFileSource
...
The NextPacket method should now return NULL when the end of the
source was reached. In the RtpFileSource, this means that when
the end of file is reached, NULL is returned. Also, when an RTCP
packet is encountered, the next packet will be read from file
immediately.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6479 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 12:20:31 +00:00
henrik.lundin@webrtc.org
d8de0669c9
Revert "Restore ptypes.txt file"
...
This reverts r6460. It turns out the file was no longer needed after
all.
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6478 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-18 11:09:53 +00:00
turaj@webrtc.org
ec869bf781
Revert 6473 "Update generated asm offsets scripts."
...
The roll has not been successful. Reverted based on the request of the committer.
> Update generated asm offsets scripts.
>
> Libvpx updated the unpack scripts to fix building dependencies.
>
> Roll libvpx 269083:277778
> See https://codereview.chromium.org/295313002/
> https://codereview.chromium.org/298063002/
> https://codereview.chromium.org/305533008/
> https://codereview.chromium.org/305703002/
> https://codereview.chromium.org/298383003/
> https://codereview.chromium.org/302863004/
> https://codereview.chromium.org/320923003/
> https://codereview.chromium.org/325313007/
> for the libvpx changes.
>
> See https://codereview.chromium.org/313243004/
> for the WebView changes.
>
> *NOTE* This CL will break the Android bots as they are built in a
> Chromium checkout, which will pull in old libvpx DEPS. They will
> cycle to green when we roll libvpx into Chromium. We must do the
> rolls in this order because we have to land webrtc and libvpx at
> the same time into Chromium.
>
> BUG=377062
> TBR=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/15809004
TBR=fgalligan@google.com
Review URL: https://webrtc-codereview.appspot.com/18589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6475 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 19:07:56 +00:00
jiayl@webrtc.org
e398954658
Update usrsctp to r8875
...
TBR=pthatcher@webrt.org
BUG=2749
Review URL: https://webrtc-codereview.appspot.com/16739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6474 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 18:16:08 +00:00
fgalligan@google.com
32196decd6
Update generated asm offsets scripts.
...
Libvpx updated the unpack scripts to fix building dependencies.
Roll libvpx 269083:277778
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
https://codereview.chromium.org/325313007/
for the libvpx changes.
See https://codereview.chromium.org/313243004/
for the WebView changes.
*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.
BUG=377062
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6473 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 17:55:23 +00:00
stefan@webrtc.org
a15fbfdcde
Add round-robin selection of send stream to pad on.
...
BUG=1812
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 17:32:05 +00:00
niklas.enbom@webrtc.org
9c09e6ee2b
Add high perf mode to VP8
...
R=marpan@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 16:32:08 +00:00
andrew@webrtc.org
26eaf7c7f7
Add a check to all.gyp to respect the include_tests variable.
...
When include_tests==0, tests should be excluded from the build. This
ensures libjingle_tests.gyp is excluded appropriately.
BUG=b/15673188
R=tnakamura@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6469 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 16:10:20 +00:00
jiayl@webrtc.org
2eaac188bb
Makes the sid of a closed DataChannel available to reuse per the spec.
...
BUG=2646
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6468 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 16:02:46 +00:00
henrike@webrtc.org
a685c9df62
base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/
...
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 14:48:44 +00:00
henrike@webrtc.org
5654b305e5
Rebase webrtc/base with r6464 version of talk/base:
...
cd webrtc/base
svn diff -r 6463:6464 http://webrtc.googlecode.com/svn/trunk/talk/base >
6464.diff
patch -p0 -i 6464.diff
BUG=3379
TBR=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12749005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6466 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 14:37:05 +00:00
tina.legrand@webrtc.org
d469443959
Rolling new version of opus.gyp
...
This roll includes changes that enables FIXED_POINT and -O3 for Opus when building for ARM, for higher speed.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6465 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 13:24:48 +00:00
phoglund@webrtc.org
ed3e0d8f0d
Increasing tolerances quite a bit to fight flakes.
...
From these errors:
[----------] 3 tests from ProfilerTest
[ RUN ] ProfilerTest.TestFunction
../../talk/base/profiler_unittest.cc:56: Failure
The difference between kWaitSec and event->mean() is 0.13612610600000002, which exceeds kTolerance, where
kWaitSec evaluates to 0.25,
event->mean() evaluates to 0.38612610600000002, and
kTolerance evaluates to 0.10000000000000001.
[ FAILED ] ProfilerTest.TestFunction (655 ms)
[ RUN ] ProfilerTest.TestScopedEvents
../../talk/base/profiler_unittest.cc:98: Failure
The difference between kEvent2WaitSec and event2->mean() is 0.33170768900000003, which exceeds kTolerance, where
kEvent2WaitSec evaluates to 0.14999999999999999,
event2->mean() evaluates to 0.48170768899999999, and
kTolerance evaluates to 0.10000000000000001.
I didn't spend time understanding why; I reckon the test had too tight
tolerances to start with so I'm just adjusting them a bit. That's
probably better than disabling the test, now it still has some value.
R=aluebs@webrtc.org
TBR=aluebs@webrtc.org
BUG=None
Review URL: https://webrtc-codereview.appspot.com/13729005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6464 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 11:09:00 +00:00
buildbot@webrtc.org
ae740dd94c
(Auto)update libjingle 69359922-> 69365993
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6463 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 10:56:41 +00:00
minyue@webrtc.org
d42da54768
Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..."
...
> Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
>
> TEST=passed_all_trybots
> R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/16619005
TBR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6462 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 09:50:12 +00:00
kjellander@webrtc.org
851a09e71a
Initial GN work for WebRTC
...
This CL makes it possible to build the 'webrtc_base'
target using GN.
The majority of our GYP stuff in webrtc/build/common.gypi has been
translated into the configs of webrtc/BUILD.gn.
The webrtc/base/base.gyp file is translated into webrtc/base/BUILD.gn.
This CL depends on https://codereview.chromium.org/322373002/ for the
jsoncpp BUILD.gn file and the ssl config.
To build inside Chromium, https://codereview.chromium.org/321313006/
needs to be landed first.
BUG=webrtc:3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true" && ninja -C out/Default
I also ran:
gn gen out/Default --args="build_with_chromium=false have_dbus_glib=true"
but it fails to compile: something is probably wrong with with pkg-config for that.
For Chromium, I symlinked src/third_party/webrtc to the webrtc subfolder of the
WebRTC checkout and applied the following patches:
https://codereview.chromium.org/322373002 (for jsoncpp and ssl config)
https://codereview.chromium.org/321313006 (enable building WebRTC)
Then I built successfully using:
gn gen out/Default && ninja -C out/Default webrtc_base
R=brettw@chromium.org
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6461 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:54:03 +00:00
henrik.lundin@webrtc.org
2ca2188906
Restore ptypes.txt file
...
The file was lost when the neteq folders where moved and renamed.
BUG=2996
TBR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6460 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:51:01 +00:00
phoglund@webrtc.org
6b061425c2
Updated W3C getusermedia tests to the latest version of the spec.
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BUG=webrtc:3455
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:46:58 +00:00
minyue@webrtc.org
8f8503d947
Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling.
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TEST=passed_all_trybots
R=henrik.lundin@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6458 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 08:02:05 +00:00
buildbot@webrtc.org
44a317a698
(Auto)update libjingle 69337301-> 69359922
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6457 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 07:49:15 +00:00
henrike@webrtc.org
9f36c087f1
Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries.
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BUG=3379
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17699005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 21:35:20 +00:00
buildbot@webrtc.org
53f57936c1
(Auto)update libjingle 69306183-> 69323802
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6454 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 21:08:51 +00:00
pbos@webrtc.org
587ef60056
Implement RTP extension support in WebRtcVideoEngine2.
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BUG=1788
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6453 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 17:32:02 +00:00
buildbot@webrtc.org
d054bff3b9
(Auto)update libjingle 69292418-> 69293749
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6452 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:37:41 +00:00
asapersson@webrtc.org
d980307197
Add max limit of number for overuses. When limit is reached always apply the rampup delay.
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BUG=1577
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6451 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:27:19 +00:00
buildbot@webrtc.org
88d9fa63df
(Auto)update libjingle 69291002-> 69292418
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6450 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:11:32 +00:00
asapersson@webrtc.org
4b12d40008
Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
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BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 14:09:28 +00:00
buildbot@webrtc.org
27626a6256
(Auto)update libjingle 69278008-> 69291002
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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6448 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 13:39:40 +00:00
kjellander@webrtc.org
d6e2213edd
Remove ivinnichenko from webrtc/test/OWNERS
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Apparently, We're doing a poor job of cleaning out
really old OWNERS.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6447 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 11:42:27 +00:00