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Author SHA1 Message Date
wu@webrtc.org
6c75c98964 Propagate capture ntp timestamp from rtp to renderer. 
Mostly the interface changes, the real implementation of ntp timestamp will come in a follow up cl.

TEST=new tests and try bots
BUG=3111
R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 17:46:33 +00:00
buildbot@webrtc.org
1fd5b45a0e (Auto)update libjingle 64956819-> 64982143 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 17:39:43 +00:00
stefan@webrtc.org
2f8d5f3302 Check if a header extension is registered before updating it and fail silently if it's not. 
BUG=
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 12:28:46 +00:00
kjellander@webrtc.org
190b72a350 Make libjingle Android example build without sourcing envsetup.sh 
See https://webrtc-codereview.appspot.com/11799004
for full details (separate to avoid webrtc+talk changes in same CL).

BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 08:35:49 +00:00
kjellander@webrtc.org
6e105ede38 Make WebRTC Android examples build without sourcing envsetup.sh 
The new recipes framework for configuring build explicitly sets the
GYP_DEFINES for Android builds instead of relying on the envsetup.sh script
which probably will be removed at some point in the future.

This causes our build to break since our Android examples relies on the
Android SDK being found using the ANDROID_SDK_ROOT environment variable.
A GYP variable 'android_sdk_root' exists and is set correctly by
common.gypi, which is what I'm using to pass this path correctly to these
tests.

The libjingle example is handled separately in
https://webrtc-codereview.appspot.com/11809004/

BUG=chromium:346198
TEST=Local builds using:
. build/android/envsetup.sh
unset ANDROID_SDK_ROOT
webrtc/build/gyp_webrtc
ninja -C out/Debug
ninja -C out/Release
+ trybots passing: git try --bot=android,android_rel,android_clang

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5907 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 08:35:00 +00:00
mallinath@webrtc.org
ad4440a64e In shared socket mode, use udp port as default receiver even if 
stun server address is not set.

This can happen in a loopback scenarios where clients do not need
to provide any server information.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 01:10:58 +00:00
buildbot@webrtc.org
505f400f27 (Auto)update libjingle 64909599-> 64919255 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 20:33:47 +00:00
fischman@webrtc.org
e98598d3f0 Make everyone an OWNER for .gyp/.gypi add/delete purposes, talk/ edition. 
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of
adding or renaming files. If you're doing\n# structural changes, please get a
review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >>
$d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f;
done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the non-talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5904 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 20:31:16 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition. 
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 20:08:03 +00:00
henrik.lundin@webrtc.org
35ead381f8 Adding a config struct to NetEq 
With this change, the parameters sent to the NetEq::Create method are
collected in one NetEq::Config struct. The benefit is that it is easier
to set, change and override default values, and easier to expand with
more parameters in the future.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5902 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 18:49:17 +00:00
henrik.lundin@webrtc.org
810acbc93e New Packet and PacketSource classes for NetEq tests 
These new classes are intended to replace the old NETEQTEST_RTPpacket
classes. The code in rtp_analyze.cc has been updated to use the new
classes; other test applications will follow.

BUG=2692
R=andrew@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5901 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 18:42:23 +00:00
buildbot@webrtc.org
1da6047132 (Auto)update libjingle 64813990-> 64909599 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5900 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 18:15:15 +00:00
fischman@webrtc.org
cf0b46c762 iosdeviceinfo.cc: remove unnecessary file 
The do-nothing implementation in this file is already present in
mobiledevicemanager.cc (shared with Android) so this isn't adding value, and
causes duplicate-symbol errors under some compilers.

BUG=3201
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5899 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 18:12:32 +00:00
primiano@chromium.org
5cf73962e6 Fix gyp for video_capture/ensure_initialized.cc. 
This is a follow-up to
https://webrtc-codereview.appspot.com/11359004
which introduced an invalid dependency in the
chromium build when building without linker GC.

BUG=2974,3152,chromium:354405
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 17:26:31 +00:00
buildbot@webrtc.org
f875f15afb (Auto)update libjingle 64709629-> 64813990 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 16:06:21 +00:00
henrika@webrtc.org
b9309beea4 Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 14:12:50 +00:00
xians@webrtc.org
5692531f18 Added a new OnMoreData() interface which will not feed the playout data to APM. 
BUG=3147
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11059005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 10:50:37 +00:00
kjellander@webrtc.org
a956ec2019 Add win_drmemory_light trybot to default trybot list. 
BUG=chromium:360054
TEST=None
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-14 08:38:27 +00:00
kjellander@webrtc.org
940894d207 DrMemory: Excluding failing tests for Dr Memory Full 
PortAllocatorTest.TestEnableSharedSocketWithNat
fails in libjingle_p2p_unittest.
Example:
http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/2/steps/memory%20test%3A%20libjingle_p2p_unittest/logs/stdio
Previous experience tells me that disabling only this test
case will make another one in the same test suite fail,
so I'm blanked disabling the whole test.

TBR=phoglund
BUG=3158

Review URL: https://webrtc-codereview.appspot.com/11909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5893 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-13 11:45:43 +00:00
kjellander@webrtc.org
ef79fd7965 DrMemory: Excluding failing tests for Dr Memory Full 
BUG=3158
TEST=None
TBR=kjellander,phoglund

Review URL: https://webrtc-codereview.appspot.com/11899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-13 08:31:42 +00:00
kjellander@webrtc.org
41e8774113 DrMemory: Excluding failing tests for Dr Memory Full 
The PortTest.TestLocalToTurn of libjingle_p2p_unittest
and DtmfSenderTest.InsertDtmf of libjingle_peerconnection_unittest
failed in the first run on
http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/
However, I cannot reproduce on my machine, so I'm disabling all
test cases of those tests, assuming the others might fail as well.

BUG=3158
TEST=None
TBR=phoglund

Review URL: https://webrtc-codereview.appspot.com/11889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-12 19:53:06 +00:00
jiayl@webrtc.org
8ce7c72456 Fix the captured screen rect conversion. 
device_mode.dmPosition is already relative to the primary display's top-left, while the expected value of GetScreenRect() is also relative to the primary display's top-left.

TESTED=verified on Windows single monitor capturing and cursor capturing is fixed.

BUG=https://code.google.com/p/chromium/issues/detail?id=362631
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11789006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 22:31:15 +00:00
turaj@webrtc.org
8d1cdaa84e NetEq changes. 
BUG=
R=henrik.lundin@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9859005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 18:47:55 +00:00
kjellander@webrtc.org
ffd242432d DrMemory: Suppress and exclude more tests to green up the full build. 
Exclude failing tests in libjingle tests and suppress
failures in modules_unittests.

Also exclude the following extremely slow tests to make it
possible to get a reasonable execution time for the bot:
* TestScaler.PointScaleTest (81573 ms)
* TestScaler.BiLinearScaleTest (1111554 ms)
* TestScaler.BoxScaleTest (1129625 ms)
* TestVp8Impl.BaseUnitTest (762598 ms)
* VideoProcessorIntegrationTest.ProcessNoLossChangeBitRate (342149 ms)

TBR=phoglund@webrtc.org
BUG=3158, 3183, 3184
TEST=Successful runs on local Windows box:
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test libjingle_p2p_unittest --tool drmemory_full
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test libjingle_peerconnection_unittest --tool drmemory_full
tools\valgrind-webrtc\webrtc_tests.bat --build-dir out --target Debug --test modules_unittests --tool drmemory_full

Review URL: https://webrtc-codereview.appspot.com/11819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 14:16:27 +00:00
stefan@webrtc.org
34c5da6b5e Cleaned up logging in video_coding. 
Converted all calls to WEBRTC_TRACE to LOG(). Also removed a large number of less useful logs.

BUG=3153
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 14:08:35 +00:00
asapersson@webrtc.org
8b2ec15d1e Convert WEBRTC_TRACE to LOG in utility. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-11 07:59:43 +00:00
wu@webrtc.org
b884eb6118 (Auto)update libjingle 64630087-> 64709629 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 16:59:16 +00:00
henrike@webrtc.org
8dce41b3c6 Remove erronuous commit message from auto sync. 
BUG=N/A
TBR=kjellander@webrtc.org

http://webrtc-codereview.appspot.com/11639004/



git-svn-id: http://webrtc.googlecode.com/svn/trunk@5883 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:39:38 +00:00
pbos@webrtc.org
22cf7472a0 Disable UsesTraceCallback 
Ongoing removal of trace code is causing UsesTraceCallback to fail,
disabling it for now.

BUG=3157
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5882 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:39:22 +00:00
andresp@webrtc.org
e6013bb7be Fix loopback test for case where no constraint is given. 
R=stefan@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5881 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 14:21:45 +00:00
asapersson@webrtc.org
2a770828d8 Remove usage of webrtc trace in video processing modules. 
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 11:30:49 +00:00
andresp@webrtc.org
0273fa98e0 Add ability to control peer connection constraints for the loopback test. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 09:40:16 +00:00
buildbot@webrtc.org
15192f909e (Auto)update libjingle 64594651-> 64630087 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5878 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-10 06:34:32 +00:00
fischman@webrtc.org
f93021430d Remove self-assignment hacks that were added to avoid unused variable warnings. 
Instead, appear to use the variables.

BUG=3152
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5877 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 21:19:55 +00:00
andrew@webrtc.org
0569d93db7 Move a chatty creation log in neteq to LS_VERBOSE. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 17:48:48 +00:00
henrike@webrtc.org
8f89497949 Remove erronuous commit message. 
BUG=N/A
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 14:43:43 +00:00
solenberg@webrtc.org
f4357f3530 Make Android-APK compile in release again. 
BUG=3152
R=kjellander@webrtc.org
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 14:21:37 +00:00
kjellander@webrtc.org
52fd65b16a Partial revert of "Removing samples directory following move to Github" 
Reason: Unfortunately we depend on AppRTC being in this location
for the bots in our Chromium WebRTC waterfalls so I'm reverting
this until we've solved that dependency.

This reverts apprtc and adapter.js from being removed in r5871.

R=phoglund@webrtc.org
TBR=dutton@google.com
BUG=

Review URL: https://webrtc-codereview.appspot.com/11529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 13:52:24 +00:00
henrika@webrtc.org
8883a0f47f (landing) Exclude VoiceEngine::SetAndroidObjects in WebRTC chrome builds 
Landing https://webrtc-codereview.appspot.com/11419004/ manually.

TBR=niklase
BUG=none

Review URL: https://webrtc-codereview.appspot.com/11439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5872 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 13:04:12 +00:00
dutton@google.com
7ecc142d6b Removing samples directory following move to Github 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5871 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 09:55:54 +00:00
buildbot@webrtc.org
61c1b8ea32 (Auto)update libjingle 64585415-> 64594651 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 06:06:38 +00:00
fischman@webrtc.org
2e9d89cf77 Unbreak android APK buildbots by emptying the video_capture_tests_apk target. 
Needed until the bots start to specify include_internal_video_capture=1.

TBR=henrike@webrtc.org
BUG=3152

Review URL: https://webrtc-codereview.appspot.com/11479006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 02:34:50 +00:00
fischman@webrtc.org
b0b135e4c2 VideoCaptureAndroid: support multiple frame-rates per resolution. 
Also enables running video_capture_tests_apk on the WebRTC/Chromium APK bots,
assuming GYP_DEFINES includes include_tests=1 and
include_internal_video_capture=1.
This required running VideoCaptureAndroid's camera capture on a dedicated thread, matching other platform's video_capture impls.

BUG=2974,3152
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5868 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 01:18:32 +00:00
sergeyu@chromium.org
74f6074ec1 Fix DesktopSize::is_empty() for the case when only width or only height is 0. 
BUG=crbug.com/358909
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/11479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-09 01:04:22 +00:00
andrew@webrtc.org
a78a41f985 Move output_mixer_unittest.cc to utility_unittest.cc. 
This reflects a move of the tested code in:
https://webrtc-codereview.appspot.com/11019005/

TBR=xians

Review URL: https://webrtc-codereview.appspot.com/11449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5866 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 23:09:28 +00:00
fischman@webrtc.org
f4c9444c65 VideoCaptureAndroid: stop referencing ViERenderer 
To facilitate building video_capture's java code without video_render's java
code this reorganizes the local-preview hack to be driven by MediaEngine.
This is the "first step" in the linked bug.

BUG=3175
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5865 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 22:55:07 +00:00
henrike@webrtc.org
f824fde36f (Auto)update libjingle 64326665-> 64585415 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5864 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 22:13:01 +00:00
fischman@webrtc.org
984e4fbaaa video_capture(iOS): move stopCapture to background thread 
Also suspend frame delivery on stopCapture() to avoid pause+onVideoError
during hangup.

BUG=3162
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/11389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 21:06:52 +00:00
pbos@webrtc.org
2a03498825 Implement FEC support in VideoReceiveStream. 
Added an FEC end-to-end test. NACK+FEC is probably working but not yet tested
as the test for it must introduce packet delays as the underlying API prefers
NACK over FEC if RTT is low.

BUG=3174
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5862 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 11:21:45 +00:00
andresp@webrtc.org
dc80bae2a6 Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. 
Clean some logs and add asserts in the way.

BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-08 11:06:12 +00:00