1227e8b3451b1a2f2a765bf6101cb0862f625568 |
|
21-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] time helper functions RTP timestams helper functions moved from rtp_utility added functions to deal with CompactNtp timestamps R=åsapersson BUG=webrtc:5260 Review URL: https://codereview.webrtc.org/1535113002 Cr-Commit-Position: refs/heads/master@{#11106}
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
b8b6fbb7a5d2f5a14f7f6f81c253747aa28e4c7f |
|
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint build/include errors fixed in rtp_rtcp module BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1505993003 Cr-Commit-Position: refs/heads/master@{#10971}
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include BUG=webrtc:5095 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1413333002 . Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
d436298332c7a7ecb51241f3a66588539c2ece83 |
|
07-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove ResetStatistics from RTP feedback. BUG= R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1213603002 Cr-Commit-Position: refs/heads/master@{#9548}
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36229004 Cr-Commit-Position: refs/heads/master@{#8517} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
273fbbb921e61273c3d83eb494d0a68db7834d3d |
|
27-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Update StreamDataCounter with FEC bytes. Add histograms stats for send/receive FEC bitrate: - "WebRTC.Video.FecBitrateReceivedInKbps" - "WebRTC.Video.FecBitrateSentInKbps" Correct media payload bytes in StreamDataCounter to not include FEC bytes. Fix stats for rtcp packets sent/received per minute (regression from r7910). BUG=crbug/419657 R=holmer@google.com, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values. Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
ce4e9a356200170abcdd44ff2af95f87a6781b8e |
|
18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor some receive-side stats. Removes polling of CName as well as receive codec statistics in favor of internal callbacks keeping a statistics struct up to date. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/28259005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
|
15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
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
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
97d0489058ae7a983f7306f32cfd49a2356c6488 |
|
09-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add video send bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateSentInKbps") - media bitrate ("WebRTC.Video.MediaBitrateSentInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps") - retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps") Add retransmitted bytes to StreamDataCounters. Change in UpdateRtpStats to also update counters for retransmitted packet. BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
d952c40c7e31c1603988c1f09ebfba9f17c6a866 |
|
27-Nov-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add receive bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateReceivedInKbps") - media bitrate ("WebRTC.Video.MediaBitrateReceivedInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateReceivedInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateReceivedInKbps") BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27189005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7756 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths. See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information. This CL was reviewed and approved in pieces in the following CLs: https://webrtc-codereview.appspot.com/24209004/ https://webrtc-codereview.appspot.com/24229004/ https://webrtc-codereview.appspot.com/24259004/ https://webrtc-codereview.appspot.com/25109004/ https://webrtc-codereview.appspot.com/26099004/ https://webrtc-codereview.appspot.com/27069004/ https://webrtc-codereview.appspot.com/27969004/ https://webrtc-codereview.appspot.com/27989004/ https://webrtc-codereview.appspot.com/29009004/ https://webrtc-codereview.appspot.com/30929004/ https://webrtc-codereview.appspot.com/30939004/ https://webrtc-codereview.appspot.com/31999004/ Committing as TBR to the original reviewers. BUG=chromium:81439 TEST=none TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom Review URL: https://webrtc-codereview.appspot.com/23129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
c30e9e230065ddde4cc439d9ba430273413e70d7 |
|
08-Sep-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Ignore FEC packet in stats, if it is first packet on ssrc. BUG=chrome:410456 R=mflodman@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22309004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
a45cac0fb79782fd4bfe9c6ef1e1a74074a33aee |
|
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Avoid potential dead lock in StreamStatisticianImpl Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to ReceiveStatisticsImpl, into separate methods with guards agains having incorrect lock order. BUG=2856 R=andresp@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7649004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
c00adbed7388c7c3a2e6214e6ab06242997e1825 |
|
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should be cached while holding lock to avoid race condition. Also, rtp_callback_ do not need to be called in GetStatistics() at all BUG=2853 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
0e93257cee79c0d19ddaef1f14ba750bf469a084 |
|
23-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for receive channel RTP statistics This allows a listener to receive new statistics (byte/packet counts, etc) as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable. The change is primarily targeted at the new video engine API. TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up. BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6259004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
7dba27c740048c92692d4e1cf6fee1fee7827901 |
|
21-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Potential dead lock in receive statistics A dead lock could occur if the following to code paths are called concurrently: ReceiveStatisticsImpl::IncomingPacket() -> StreamStatisticianImpl::IncomingPacket() StreamStatisticianImpl::GetStatistics() -> ReceiveStatisticsImpl::StatisticsUpdated() Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket(). BUG=2818 R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
54ae4ffb9e235a9742e2b11298327e02d870571c |
|
19-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for receive channel RTCP statistics. This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable. The change is primarily targeted at the new video engine API. TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up. BUG=2235 R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
|
06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for combining RTX and FEC/RED. This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
286fe0b04d97205ac84688bbe613d5749192b2d1 |
|
21-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ...""" ...and fixes the RTCP bug. BUG=2277 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2089004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
a0218a84d17a727111e2e24cf5af915b1b91c06e |
|
21-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 4582 "Reverts a second set of reverts caused by a bug in ..." > Reverts a second set of reverts caused by a bug in a dependency. > > Revert "Revert r4328" > > Revert "Revert r4322 "Support sending multiple report blocks and keeping track > of statistics on" > > BUG=1811 > R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/2072004 TBR=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2087004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
1a65d6c36b6a25f9f734176c697c684c3b43ac4b |
|
21-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reverts a second set of reverts caused by a bug in a dependency. Revert "Revert r4328" Revert "Revert r4322 "Support sending multiple report blocks and keeping track of statistics on" BUG=1811 R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2072004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
|
16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584. Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
aa4d96a134a03f998d52fb9699845d9c644eb24b |
|
16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4301 R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
b7eda43810125cd01b29671a6beab61ddb48ebdb |
|
15-Jul-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4322 "Support sending multiple report blocks and keeping track of statistics on several SSRCs" R=pwestin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1774006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
717d147ebb168ed498fa4777ffaf8646a1dc6d7a |
|
10-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Support sending multiple report blocks and keeping track of statistics on several SSRCs. BUG=1811 TEST=vie_auto_test --automated, voe_auto_test --automated, trybots R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1768004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
|
66b2e5c05a3f2a93d634d1dbbcbb283fb218ca4f |
|
05-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
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