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Author SHA1 Message Date
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. 
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. 
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 
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. 
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 
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. 
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 
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. 
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 
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. 
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 
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 
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
henrik.lundin@webrtc.org
1e3c5c248a Importing ThreadChecker class from Chromium 
The ThreadChecker class is imported/re-implemented from Chromium.
The implementation is changed to depend on WebRTC primitives.

R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 11:34:44 +00:00
bjornv@webrtc.org
b099a6f9ab Adds aluebs@webrtc.org as owner to audio_processing 
BUG=N/A
TESTED=trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6445 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 10:30:58 +00:00
bjornv@webrtc.org
721f970cba common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 
We should avoid macros in general (see style guide) and the shift ones are particular dangerous since they assume that the user apply a non-negative shift.

Related CL: https://webrtc-codereview.appspot.com/16669004

BUG=3348,3353
TESTED=trybots and manually on linux
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 10:30:14 +00:00
mflodman@webrtc.org
eb16b811fb Implements start bitrate for new video API. 
Added  a new rampup test.

BUG=2879
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6443 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 08:57:39 +00:00