asapersson@webrtc.org
d08d389ce8
Add field to counters for when first rtp/rtcp packet is sent/received.
...
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).
BUG=crbug/419657
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 12:03:11 +00:00
bjornv@webrtc.org
b395a5ea65
audio_processing: Moved legacy AGC code to webrtc/modules/audio_processing/agc/legacy/
...
include/ is renamed to legacy/ and analog_agc.* and digital_agc.* moved into the directory.
BUG=
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7909 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 10:38:10 +00:00
guoweis@webrtc.org
55360ae402
Revert "Add adapter_type into Candidate object."
...
This reverts commit aaf02cc2d4
.
BUG=
TBR=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 05:28:10 +00:00
marpan@webrtc.org
d021bbbc9e
Fix vp9 setting in vie loopback test.
...
If vp9 codec was selected then videoCodec.codecSpecific.VP8.numberOfTemporalLayers was being set.
TBR=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/37389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7907 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-16 00:21:47 +00:00
guoweis@webrtc.org
aaf02cc2d4
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7885
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 23:03:10 +00:00
pkasting@chromium.org
0b1534c52e
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
...
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00
aluebs@webrtc.org
96a626262a
Remove 20ms support in AGC
...
Today, 10ms is the standard chunk length used in whole AudioProcessing, so this was only adding unnecessary complexity and maintainance.
Removing it doesn't change the bahavior in any use case of today.
R=andrew@webrtc.org , bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7904 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 21:54:50 +00:00
guoweis@webrtc.org
1f05c45976
Reenable test case P2PTransportChannelTest.TestIPv6Connections
...
BUG=3317
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 21:25:54 +00:00
pbos@webrtc.org
a7f77720cb
Merge in AGC manager and AGC tools.
...
R=bjornv@webrtc.org
BUG=4098
Review URL: https://webrtc-codereview.appspot.com/37379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7902 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 16:33:16 +00:00
bjornv@webrtc.org
903b4ae603
Removes unused test files by audio_processing/transient
...
BUG=
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7901 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 16:13:05 +00:00
bjornv@webrtc.org
dd322136fe
resources/audio_processing: Removed unused test files
...
Two files not used by any tests are removed.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32829004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 15:57:11 +00:00
minyue@webrtc.org
6fd9308420
Suppressing warnings in GetRTT() in VoE.
...
GetRTT() was separated from GetRTPStatistics() but the warnings were not updated.
Now GetRTT() is only only used by GetRTPStatistics() and the warning pops up pointlessly and too often.
This CL is to suppress these warnings and maintain a proper warning for GetRTPStatistics().
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7899 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 14:56:44 +00:00
tommi@webrtc.org
e2e199b894
Clean up StatsObserver's OnComplete methods (address TODOs).
...
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 13:22:54 +00:00
pbos@webrtc.org
3440fe1bc5
Use webrtc_root instead of DEPTH for iSAC.
...
Un-breaks chromium.webrtc.fyi. Broken as Chromium doesn't have webrtc/
checked out in root.
TBR=bjornv@webrtc.org ,tommi@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/28289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7897 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 10:56:50 +00:00
buildbot@webrtc.org
032b802a8c
(Auto)update libjingle 82121498-> 82126219
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:48:07 +00:00
tommi@webrtc.org
dd0601fbcf
Remove unneeded ctor and add a more practical one
...
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:47:49 +00:00
tommi@webrtc.org
69bc5a300f
Add thread asserts to StatsCollector.
...
Also adding a "ForTest" postfix to a test-only method.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:44:48 +00:00
pbos@webrtc.org
788acd17ad
Merge audio_processing changes.
...
R=aluebs@webrtc.org , bjornv@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/32769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 09:41:24 +00:00
pbos@webrtc.org
fb108b5a28
Revert r7885.
...
Breaks compile step of other code where network name of
cricket::Candidate is used.
TBR=guoweis@webrtc.org ,juberti@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/31229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 08:04:50 +00:00
andrew@webrtc.org
b413a30097
Add WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
...
This intrinsics version gives bit-exact result as the current assembly
neon code. And the performance is 38% better than current assembly
neon version, 5.92 times faster than current C version. The test runs
under Cortex-a53 aarch32 mode, other cpu should give similar performance
result.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I257e33ef6d634a519fd71adc4f52b06dd655bd9d
Review URL: https://webrtc-codereview.appspot.com/32749004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7891 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:23:49 +00:00
pbos@webrtc.org
18a3896bd2
Revert r7886:7887.
...
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.
TBR=tommi@webrtc.org ,pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/36439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 07:03:04 +00:00
andrew@webrtc.org
40e4767f2b
Add NEON intrinsics version for min_max_operations_neon.c
...
WebRtcSpl_MinValueW32Neon, WebRtcSpl_MaxValueW32Neon, WebRtcSpl_MaxValueW16Neon
and WebRtcSpl_MaxAbsValueW32Neon are added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: Id461d64c3313f56147edadd2231e8845574ead2a
Review URL: https://webrtc-codereview.appspot.com/28259004
Patch from Yang Zhang <yang.zhang@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7889 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 06:07:47 +00:00
magjed@webrtc.org
e575e9c40f
Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h
...
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-14 11:09:23 +00:00
pthatcher@webrtc.org
e9db7fe80c
Put pseudotcp back because remoting uses it.
...
R=andrew@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-13 01:56:39 +00:00
pthatcher@webrtc.org
dee76f3b89
Move the obvious/easy Jingle-specific code into webrtc/libjingle.
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
guoweis@webrtc.org
8c9d79a29d
Add adapter_type into Candidate object.
...
Expose adapter_type from Candidate such that we could add jmidata on top of this.
Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.
This is migrated from issue 32599004
BUG=
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 19:21:14 +00:00
tommi@webrtc.org
c57310b982
Switch kStatsValueName* constants to be enums instead of char*.
...
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 17:41:28 +00:00
henrik.lundin@webrtc.org
3b79daff14
Moving encoded_bytes into EncodedInfo
...
BUG=3926
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7883 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 13:31:24 +00:00
kjellander@webrtc.org
c8bc717905
Fix webrtc gn windows build.
...
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7882 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 12:10:46 +00:00
jansson@webrtc.org
f68faa542a
Removing manual test pages because they have been moved to github.
...
BUG=https://github.com/GoogleChrome/webrtc/issues/203
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7881 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 09:30:41 +00:00
pthatcher@webrtc.org
40b276ea7b
Cleanup little things found when refactoring.
...
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/33519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 02:44:30 +00:00
aluebs@webrtc.org
27d106bcf7
Move the downmixing out of AudioBuffer
...
This provides more flexibility if some component in AudioProcessing wants to operate before downmixing.
Now the AudioProcessing does only track the processing rate, but not the processing number of channels. This is tracked by the AudioBuffer itself and can be changed at any time to one smaller or equal the input number of channels. For each chunk it is reset to input number of channels and the end it should be equal to the output number of channels.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7879 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 17:09:21 +00:00
minyue@webrtc.org
0ca768b131
Adding DTX to WebRTC Opus wrapper (relanding).
...
This is relanding of r7846, which failed since the unit test depended on whether Opus is in fixed-point or float-point.
See the review of r7846 here:
https://webrtc-codereview.appspot.com/13219004/
Patch set 1 is the same as r7846. Further fixes are found in patch set 2 and later.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 16:09:35 +00:00
pbos@webrtc.org
5f162c8509
Merge AEC changes.
...
R=bjornv@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/34459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7877 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:46:59 +00:00
pbos@webrtc.org
2b19f06312
Wire up RTT statistics to webrtc::Call.
...
R=mflodman@webrtc.org , stefan@webrtc.org
BUG=1667,1788
Review URL: https://webrtc-codereview.appspot.com/32249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:26:09 +00:00
pbos@webrtc.org
13518951e3
Remove old_factory from WebRtcVideoEngine.
...
Minor pending cleanup.
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/28239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 13:14:30 +00:00
perkj@webrtc.org
128fabaf7b
Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin...""
...
Original cl description:
Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
This also add a new build target to build java PeerConnection using Chromes build macros.
BUG=4031
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 12:25:57 +00:00
kjellander@webrtc.org
626c09f6a3
Move isolate path into webrtc/build/android/test_runner.py
...
This will make it easier to execute tests and allows
for more cleanup in the buildbot recipes.
Now tests can be listed using:
webrtc/build/android/test_runner.py gtest --help
and executed like
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests
TESTED=
Ran:
webrtc/build/android/test_runner.py gtest --help
and verified the tests were listed.
I wiped /sdcard/resources on my device, executed:
webrtc/build/android/test_runner.py gtest -s audio_decoder_unittests
and verified it passed and that resources/audio_coding/testfile32kHz.pcm
was copied to the device.
BUG=
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7873 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 11:59:46 +00:00
henrik.lundin@webrtc.org
817e50dd7d
Make an AudioEncoder subclass for PCM16B
...
The implementation depends on AudioEncoderPcm to reduce code
duplication.
BUG=3926
R=kjellander@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7872 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:47:19 +00:00
kwiberg@webrtc.org
b3ad8cf6ca
Make an AudioEncoder subclass for iSAC
...
BUG=3926
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7675
and reverted: https://code.google.com/p/webrtc/source/detail?r=7676
R=henrik.lundin@webrtc.org , kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7871 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 10:08:19 +00:00
minyue@webrtc.org
abe3f1879c
Checking whether ACM uses codec internal or WebRTC DTX.
...
It was not clear how one could know if ACM is using DTX from WebRTC or codec internal DTX.
This CL makes better use of IsInternalDTXReplacedWithWebRtc() which was designed for G.729 to export such information.
Before
IsInternalDTXReplacedWithWebRtc() gives true only if codec == G729 and G729's internal DTX is replaced with WebRTC DTX.
Now
IsInternalDTXReplacedWithWebRtc() gives true also when codec does not have internal DTX, i.e., must use WebRTC DTX, which is much more logical.
BUG=
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7870 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:53:21 +00:00
kwiberg@webrtc.org
55d42c32a4
DCHECK: Reference condition parameter in release builds
...
So that caller's won't get warnings about unused variables for
variables that are only used in calls to DCHECK, such as
int x = ...
DCHECK_EQ(x, 17);
R=andrew@webrtc.org
Previously committed: https://code.google.com/p/webrtc/source/detail?r=7858
and reverted: https://code.google.com/p/webrtc/source/detail?r=7859
Review URL: https://webrtc-codereview.appspot.com/31169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7869 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 08:32:30 +00:00
phoglund@webrtc.org
cd5b209d68
Deleting quality dashboard code.
...
This code has served its purpose and can now be deleted. The dashboard
was mostly defunct, so I've turned it off in AppEngine.
R=kjellander@webrtc.org
BUG=None
Review URL: https://webrtc-codereview.appspot.com/33599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7868 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 07:57:22 +00:00
andrew@webrtc.org
3c31e6e2f9
Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon
...
WebRtcSpl_MinValueW16Neon is added. SplTest in common_audio_unittests
is passed on ARM32/ARM64 platforms.
BUG=4002
R=andrew@webrtc.org , jridges@masque.com
Change-Id: I33c3853766d7594ed121166288e5325a03b3d6fe
Review URL: https://webrtc-codereview.appspot.com/32639004
Patch from Yang Zhang <yang.zhang@arm.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7867 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-11 00:24:13 +00:00
mflodman@webrtc.org
f4c19480fc
Remove jitter_estimate_test.h
...
BUG=2156
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7866 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 21:08:39 +00:00
aluebs@webrtc.org
c5ebbd98f5
Support 48kHz in Noise Suppression
...
Doing the same for the 16-24kHz band than was done in the 8-16kHz.
Results look and sound as nice.
BUG=webrtc:3146
R=andrew@webrtc.org , bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7865 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 19:30:57 +00:00
pbos@webrtc.org
d8ca723de7
Remove CELT support from audio_coding.
...
R=henrik.lundin@webrtc.org , juberti@webrtc.org
TBR=kjellander@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/33579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7864 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 11:49:13 +00:00
asapersson@webrtc.org
8084f9500f
Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval.
...
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7863 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 11:04:13 +00:00
pbos@webrtc.org
85bd53e7c9
Add AbsSendTime unittests to rampup_tests.cc.
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R=stefan@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/28229004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7862 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 10:36:20 +00:00
asapersson@webrtc.org
0df371549f
Cast payload type to int in logs.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7861 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-10 10:30:45 +00:00