6db6cdc604f9a866991ecf8454eb7f7aa69918ea |
|
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1513303003 Cr-Commit-Position: refs/heads/master@{#11025}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
5c1def8892390a336d1eecd9b61adacece858898 |
|
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
modules/rtp_rtcp/include folder cleared of lint warnings Functions that do not follow lint are marked deprecated, including function in the interface. BUG=webrtc:5308 R=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1493403003 Cr-Commit-Position: refs/heads/master@{#10975}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_impl.h
|
da903eaabbb6c6830efcafc3c2ade1d36f511e43 |
|
02-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify newapi::RtcpMode and RTCPMethod. BUG=webrtc:1695 R=solenberg@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1373903003 Cr-Commit-Position: refs/heads/master@{#10143}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
ac547a653862744d0aae560713f8418ad2852085 |
|
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces. Starts by removing channel/engine id from ViEChannel which propagates down to the RTP/RTCP module as well as the transport class. IncomingVideoStream::RenderFrame() is untouched for now but receives a fake id instead of the previous channel id. Added a TODO to remove it later but the RenderFrame call is implemented in a lot of platform-dependent files and should probably remove the "manager" aspect of renderers, so preferring to do it separately BUG=webrtc:1695 R=henrika@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1335353005 . Cr-Commit-Position: refs/heads/master@{#9978}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
233bd87d45bbeeec50d7687e7d98c1cfc7f65562 |
|
08-Sep-2015 |
sprang <sprang@webrtc.org> |
Add RemoteEstimatorProxy for capturing receive times For use when send-side bandwidth estimation is enabled. Receive times need to be captured, buffered and then sent using TransportFeedback RTCP messaged back to the send side. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1290813008 Cr-Commit-Position: refs/heads/master@{#9898}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
a38233a586dd865c0cd728ce523b3a82ca52ea8b |
|
24-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Removed extended jitter report from RtcpSender. This was never used (value always 0, when sent) BUG=2450 R=asapersson@webrtc.org Review URL: https://codereview.webrtc.org/1208843003 . Cr-Commit-Position: refs/heads/master@{#9631}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
ba8c15b857c0f341d9c1e02d41b6ccd56f9f1030 |
|
14-Jul-2015 |
pbos <pbos@webrtc.org> |
Merge methods for configuring NACK/FEC/hybrid. BUG=webrtc:1695 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1226143013 Cr-Commit-Position: refs/heads/master@{#9580}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
30409b4dca3d9cfdb0e714a5932b135becb0f822 |
|
11-Jul-2015 |
bcornell <bcornell@google.com> |
Add statistics gathering for packet loss. Adds a class used to classify whether packet loss events are a single packet or multiple packets as well as how many packets have been lost. Also exposes a new function in the RtpRtcp interface to retrieve these statistics. BUG= Review URL: https://codereview.webrtc.org/1198853004 Cr-Commit-Position: refs/heads/master@{#9568}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_impl.h
|
0ea42d319e2a18785f5de5fe8d52e0a7a5fd1448 |
|
25-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
Send Sdes using RtcpPacket BUG=2450 R=asapersson@webrtc.org Review URL: https://codereview.webrtc.org/1196863003. Cr-Commit-Position: refs/heads/master@{#9504}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
9ba52f89acd1b9bc88115880dfe2716147bf3b5d |
|
01-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Remove intermediate RTCP CNAME buffers. Sets CNAME using a pointer to only perform a copy inside the RTCP sender. BUG= R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50169005 Cr-Commit-Position: refs/heads/master@{#9346}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
71861a0e2039e1729ad34758474d5e569012fd2f |
|
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove GetSendSideDelay from RtpRtcp. These stats are reported using a callback either way, removing a getter + an old related deadlock suppression. BUG=1695, 2999 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50119004 Cr-Commit-Position: refs/heads/master@{#9314}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
11beccd712dd52ae73c078332122070de3cb5c3d |
|
28-May-2015 |
Erik Språng <sprang@webrtc.org> |
Remove external report blocks from RtcpSender and rtp_rtcp interface. Feature does not seem to be used and complicates other refactoring of the rtcp module. BUG= R=asapersson@webrtc.org, henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/54569004 Cr-Commit-Position: refs/heads/master@{#9304}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
242e22b055940be70b1df3031e2363b0d02397b2 |
|
11-May-2015 |
Erik Språng <sprang@webrtc.org> |
Refactor RTCP sender The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but it has quite a few ramifications. Notable changes: * Removed the rtcpPacketTypeFlags bit vector and don't assume RTCPPacketType values have a single unique bit set. This will allow making this an enum class once rtcp_receiver has been overhauled. * Flags are now stored in a map that is a member of the class. This meant we could remove some bool flags (eg send_remb_) which was previously masked into rtcpPacketTypeFlags and then masked out again when testing if a remb packet should be sent. * Make all build methods, eg. BuildREMB(), have the same signature. An RtcpContext struct was introduced for this purpose. This allowed the use of a map from RTCPPacketType to method pointer. Instead of 18 consecutive if-statements, there is now a single loop. The context class also allowed some simplifications in the build methods themselves. * A few minor simplifications and cleanups. The next step is to gradually replace the builder methods with the builders from the new RtcpPacket classes. BUG=2450 R=asapersson@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48329004 Cr-Commit-Position: refs/heads/master@{#9166}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
61be2a401635eed1d13c169dc104b9ff4a2f477b |
|
27-Apr-2015 |
Erik Språng <sprang@google.com> |
Clean up RTCPSender. Reformat to current code style, remove non-const references, use scoped_ptr, remove empty comments and dead code, etc.. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49019004 Cr-Commit-Position: refs/heads/master@{#9086}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
e62202fedf57b74cc263246c0586ee353978caf8 |
|
21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
Support handling multiple RTX but only generate SDP with RTX associated with VP8. This implementation registers RTX-APT map inside RTP sender and receiver. While it only generates SDP with RTX associated with VP8 to make it compatible with previous Chrome versions. Should add following changes after reaches stable, * Use RTX-APT map for building and restoring RTP packets. * Add RTX support for RED or VP9 in Video engine. * Set RTX payload type for RED inside FecConfig in EndToEndTest. BUG=4024 R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36889004 Cr-Commit-Position: refs/heads/master@{#9040}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
6ae2572fa6dc29349b946e3cfd926289e54d9371 |
|
13-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add missing configuration of rtx payload type for rtp/rtcp module. BUG=4528 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/51639004 Cr-Commit-Position: refs/heads/master@{#8989}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_impl.h
|
9dd0ebc379d449d91c61884997e451d2c52a350f |
|
26-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove the default RTP module. This CL removes the default module owned by ViEEncoder, functionality in the module to register default modules and the final changes in rtp_rtcp_impl using default/child modules. BUG=769 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42509004 Cr-Commit-Position: refs/heads/master@{#8514} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8514 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
50e28166afcdf4b2fcc6e331b70e77a284c3a560 |
|
23-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Move SetTargetSendBitrates logic from default module to payload router. This cl just moves the logic form the default module SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch in size between trate the vector and rtp modules. This was the same in the default module and is quite hard to protect from before we have the new video API. I also removed some test form rtp_rtcp_impl_unittest that were affected by this change. The test tests code that isn't implemented, hence the DISABLED_, and this will never be implemented in the RTP module, rather the payload router in the future. BUG=769 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42419004 Cr-Commit-Position: refs/heads/master@{#8453} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
1d0fa5d352fe12092201fade249905c7e1ff974b |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add RtcpPacketTypeCounter stats to new API. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667,1788 Review URL: https://webrtc-codereview.appspot.com/37489004 Cr-Commit-Position: refs/heads/master@{#8429} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
0abc6011b968dab31635841cec64195441732992 |
|
17-Feb-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Remove SetCaptureDelay from the RTP module. This is a small step in getting rid of the default module, but also to eventually delete FrameProviderBase completely. BUG=769 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34229004 Cr-Commit-Position: refs/heads/master@{#8396} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
7c4d20fd6c95f76cf909669b94effdbef05ecb54 |
|
12-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove potential deadlock in RTPSenderAudio. Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls by doing a lot shorter locking which fetches a current state of RTPSenderAudio variables before sending. Thread annotates locked variables and removes one lock in RTPSenderAudio, bonus fixes data races reported in voe_auto_test --automated under TSan (DTMF data race). Also includes some bonus cleanup of RTPSenderVideo which removes the send critsect completely as all methods using it was always called from RTPSender under its send_critsect. R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org BUG=3001, chromium:454654 Review URL: https://webrtc-codereview.appspot.com/41869004 Cr-Commit-Position: refs/heads/master@{#8348} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
9ffd8fe96b6f7126420200ac78317756e855f1f1 |
|
21-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Indentation changes. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32149004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
0b0c24177bac6eaa27cd520595ba799e48e84a0c |
|
13-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Only return Rtx mode in RTXSendStatus(). There is no need to return 'ssrc' and 'payloadtype' inside this function since they are never used. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38569004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_impl.h
|
8f27fcce79584378da97f0d84574564799e138d6 |
|
09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 8028 "Support associated payload type when registering Rt..." Reasons for revert: 1. glaznev discovered potentially related problems using the Android AppRTCDemo. 2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky. > Support associated payload type when registering Rtx payload type. > > Major changes include, > - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. > - Receiver: Restore RTP packets by the new RTX-APT map. > - Sender: Send RTP packets by checking RTX-APT map. > - Add RTX payload type for RED in the default codec list. > > BUG=4024 > R=pbos@webrtc.org, stefan@webrtc.org > TBR=mflodman@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/26259004 > > Patch from Changbin Shao <changbin.shao@intel.com>. TBR=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
2a169640a3225a559f926fe74f1fe1af239e191f |
|
09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Support associated payload type when registering Rtx payload type. Major changes include, - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. - Receiver: Restore RTP packets by the new RTX-APT map. - Sender: Send RTP packets by checking RTX-APT map. - Add RTX payload type for RED in the default codec list. BUG=4024 R=pbos@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26259004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
d16e839c6d29831e79312180085b6a19149df43f |
|
19-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rtp-Rtcp sender cleanup. Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions. Also removed const on non-pointer/reference types for related files. BUG= R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34469004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
cb79141eabdf1d2de736fd4285dc59bb44de4682 |
|
18-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap. Removed unused function ResetRTT. BUG=4114 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33659005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_impl.h
|
ba8138ba384f72674dad9b91b9a095a0fd1b27dd |
|
08-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Change type of nack_last_time_sent_full_ from uint32_t to int64_t. Could cause nack requests to be sent too frequently. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27339004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7825 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
9334ac2d78f760b37f512ef6c12bff220d1654c1 |
|
24-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Use vector of CSRCs for DeliverFrame & SetCSRCs. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28029004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_impl.h
|
49ff40e32e408bc77e8c9bec6090f6aa2e445173 |
|
13-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make SetREMBData accept vector of SSRCs. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
1972ff8a6e45f7ad3fb7e4ed51dc0135c72f6c9d |
|
11-Sep-2014 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. This will make a subsequent change I intend to do safer, where I'll change the return type of one of the base Module functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions (in many cases apparently virtual "overrides" of no-longer-existent base functions). I've removed some of these. This also highlighted several cases where "virtual" was used unnecessarily to mark a function that was only defined in one class. Removed "virtual" in those cases. BUG=none TEST=none R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
2f4b14e3f31b34a50310357c6c7be86c3bca1537 |
|
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RTCP sender report send media bytes. r6654 changed RtpSender::Bytes() to return the number of bytes sent instead of number of media bytes. This is used by VideoEngine for stats. This change broke RTCP which sends this same count as the number of payload bytes sent (excluding headers and padding). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
4ef438e2defd6c46404f6b367287364cde66b7fb |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove the send-side cname getter APIs from voice and video engine. These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname. R=henrika@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
8f1512140ed57ce57635a1cd561b631dfdc5e05f |
|
10-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
d11bec40b25e5990bf05b410676587f6f38b9b8c |
|
08-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
2bb1bdab8d11f5445693c028335fb3ace631f636 |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Preserve RTP states for restarted VideoSendStreams. A restarted VideoSendStream would previously be completely reset, causing gaps in sequence numbers and potentially RTP timestamps as well. This broke SRTP which requires fairly sequential sequence numbers. Presumably, were this sent without SRTP, we'd still have problems on the receiving end as the corresponding receiver is unaware of this reset. Also adding annotation to RTPSender and addressing some unlocked access to ssrc_, ssrc_rtx_ and rtx_. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
a15fbfdcdee391bd87bb1c2721f0fbb824f5fbfb |
|
17-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
ef92755780253c6a7940c89598a206e58e05b810 |
|
05-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out. BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15629005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
ebdb0e3ad0a787bee066d12cdcd115a38b0a10d1 |
|
07-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. - Add ability to VoE to send Absolute Sender Time header extension. - Refactor handling of RTP header extensions in VoE to work the same as in ViE. - Add API to enable receiving Absolute Sender Time in VoE. This is part of the work to include audio packets in bandwidth estimation, for better accuracy in estimates. BUG= TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
8098e0747879b191335e8de1e16b87cf6adbdf54 |
|
19-Feb-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). Add counter to RTCP sender and RTCP receiver. Add video api GetRtcpPacketTypes(). BUG=2638 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
e6b871bb29dcdd9da08509ab3a39d90424f73781 |
|
17-Dec-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added method for getting default module state and protect agains a read/write race for child_modules_. BUG=2731 TEST=tsan R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5919005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
e7b1e112833517c334a12aac16be17a27d798944 |
|
16-Dec-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..." > Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." > > > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. > > > > R=holmer@google.com > > > > Review URL: https://webrtc-codereview.appspot.com/5049004 > > TBR=asapersson@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5799004 TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
86bb56a7f5b08ac285656bd95ddac34a7922c43a |
|
13-Dec-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..." > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. > > R=holmer@google.com > > Review URL: https://webrtc-codereview.appspot.com/5049004 TBR=asapersson@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5799004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
6811b6e308d16f160ba4c32650f195d5d3d9a2b1 |
|
13-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Callback for send bitrate estimates - new roll Issue https://webrtc-codereview.appspot.com/4459004/ was commited as r5259, after which flakiness was detected and a rollback was performed at r5261. Patch Set 1 of this issue is the code submitted in r5259. Subsequent patch sets fixes a race condition which caused the seen problems. The root cause was a dead lock between a thread sending rtp packets and and a timed module processing thread: webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock webrtc::Bitrate::Process() // Get Bitrate lock webrtc::RTPSender::ProcessBitrate() webrtc::ModuleRtpRtcpImpl::Process() ... webrtc::Bitrate::Update() // Get Bitrate lock webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock webrtc::RTPSender::SendToNetwork() ... This is fixed in Bitrate::Process() by releasing the lock before calling the callback. BUG=2235 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
096e8d9f944abeee5fb75d165d91f7a68258f073 |
|
11-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5259 "Callback for send bitrate estimates" CL is causing flakiness in RampUpTest.WithoutPacing. > Callback for send bitrate estimates > > BUG=2235 > R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/4459004 R=mflodman@webrtc.org, pbos@webrtc.org TBR=mflodman Review URL: https://webrtc-codereview.appspot.com/5579005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
2656cf9f4c37fe1360e2392a5b0101df38660403 |
|
11-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Callback for send bitrate estimates BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
96a9b2dcdcaee150f7c19f229f4b7297df76e13b |
|
05-Dec-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Use the RTT from RtcpRttStats class if provided when sending/receiving NACK. R=holmer@google.com Review URL: https://webrtc-codereview.appspot.com/5049004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5228 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
ebad765ee00b90c48507bff1997ea8c1070a9316 |
|
05-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for send channel rtp statistics BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
0a3c1471b873ea7f81bff2faa7cf0d9e563c7d53 |
|
05-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add API to query video engine for the send-side delay. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4559005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
a6ad6e5b589465f6a51ce46ee87d50e00bfd85b2 |
|
05-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for send channel rtcp statistics BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
71f055fb41336316324942f828e022e2f7d93ec7 |
|
04-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add send frame rate statistics callback BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
7e9315b42ebe8f7df860030af93618de81326503 |
|
04-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for sending redundant payloads over RTX. TEST=trybots BUG=1812 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
8d02f5dc7146ebc35c30fc3f7e1cbfa6802486a2 |
|
21-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added API for enabling/disabling RTCP Receiver Reference Time extension. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3419005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
1ae1d0c47145f1036c3844a5cd1b536c22565325 |
|
20-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2383004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
9b82f5a6ed2ceb04f72b66c1d3cca67a2bbcec3a |
|
13-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for RTX in combination with pacing. Retransmissions didn't get sent over RTX when pacing was enabled since the pacer didn't keep track of whether a packet was a retransmit or not. BUG=1811 TEST=trybots R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
766154aa1d9cdb7a8f9ac16611a1e4a13060b85b |
|
04-Nov-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed unused code. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
7d6bd2201938e4b6b7e5219c0fc971b0e1ba05b1 |
|
31-Oct-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Propagate estimated RTT from receivers to rtt observer. BUG=1613 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
8469f7b328ec980f80fa79931b4e07872d0feb23 |
|
02-Oct-2013 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Added support for sending and receiving RTCP XR packets: - Receiver reference time report block - DLRR report block (RFC3611). BUG=1613 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2196010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4898 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
28a331eededf17dc3a0860bb1bdf5b2dc3f9e763 |
|
17-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support for multiple report blocks. Use a weighted average of fraction loss for bandwidth estimation. TEST=trybots and vie_auto_test --automated BUG=1811 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2198004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
b2c8a952a7a996b89c6ff2ecdc1364641f2571f6 |
|
06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improving padding rules and breaking out bw allocation to ViEEncoder. BUG=1837 TESTS=vie_auto_test --automated, trybots R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2170004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_impl.h
|
f3e4ceee47d747c8868d919c179ecc640b9541f0 |
|
31-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ BUG=163 R=pwestin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1904005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4444 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_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/rtp_rtcp_impl.h
|
2e402ce873f48e0848468345d848bd3fff75dd3e |
|
20-Jun-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enqueue packet in pacer if sending fails If a packet cannot be sent while pacer is in use it should be queued. This avoid packet loss due to congestion. BUG=1930 R=pwestin@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1693004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
508a84b25597a8d12177eabed002b71f5730d4c8 |
|
17-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up pacer-based padding. This connects the pacer-based padding with the RTP modules, which will generate padding packets roughly according to what the pacer suggests. It will only generate padding packets of maximum size to keep the number off padding packets as small as possible. This also sets a limit of how much padding + media bitrate which the pacer is allowed to "request" from the RTP modules. Padding will for now only be generated by the first sending RTP module. BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1612005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
a5cb98cbbd11e93cb6d0a6232387814aac168c7d |
|
29-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out RTP header parsing from the RTP module. This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
9f5ebb525130f207229dfa350ce8c2bdd22163c7 |
|
12-Apr-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding a payload type for RTX. BUG=736 TEST=Modified RTP unittests. Review URL: https://webrtc-codereview.appspot.com/1278004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
7da3459b2ac83923c1ccbf11ad24d3f700305feb |
|
09-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." This reverts commit 4954b3650192d78037714138a5c519ef08f2670e. Reverts r3799 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1308004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
afcc6101d01be8c6cd9cf246dcf5b37b31ce0cd0 |
|
09-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. We should consider making the same change to the render timestamps generated at the receiver. BUG=1563 Review URL: https://webrtc-codereview.appspot.com/1283005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
2f44673d665899ca788ae44247a9a7f4764f5e2b |
|
08-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t for rtp_rtcp/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1279007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
bda7f305c5d7d675f1c35813bd2b2a5732775bb9 |
|
16-Mar-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding RTX on source Review URL: https://webrtc-codereview.appspot.com/1190004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
b7edd065306329309dac6767fe4914c185f941f8 |
|
12-Mar-2013 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove DTMF detection. Talk team has been in the loop and there is no need for DTMF detection at the receiver side. test=voe_auto_test, VoE extended test DTMF Review URL: https://webrtc-codereview.appspot.com/1168004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3663 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
a27107004d8544c6dbf8eaa231e6079b73c90efe |
|
05-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Split the NACK list into multiple RTCPs if it's too big. TEST=rtp_rtcp_unittests BUG=1434 Review URL: https://webrtc-codereview.appspot.com/1148006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3609 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
becf9c897c41eea3f021f99d87889c32c78b0de9 |
|
01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix mismatch between different NACK list lengths and packet buffers. This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors. BUG=1289 Review URL: https://webrtc-codereview.appspot.com/1065007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
b5865079868c4dec49571e7aef0aa52124b50c64 |
|
01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. Also make sure RTT is computed independently of whether it's time to send RTCP messages or not. BUG=1298 Review URL: https://webrtc-codereview.appspot.com/1060005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
43da54a458a7a992c702d85f0327e1d394ec5cf3 |
|
25-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted rtp_sender: made lint clean. TESTED=rtp_rtcp_unittests BUG= Review URL: https://webrtc-codereview.appspot.com/1062004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
5accd370e70b94517a39e622c75b794cc7a28829 |
|
22-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. BUG= TESTED=vie/voe_auto_test, rtp_rtcp_unittests Review URL: https://webrtc-codereview.appspot.com/1058004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
|
a678a3baee2e680bd521f3a6caf97707fffd6093 |
|
21-Jan-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1044004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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20ed36dada62ad56ec01263fc0eef0ed198f6476 |
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17-Jan-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RtpClock to system_wrappers and make it more generic. The goal with this new clock interface is to have something which is used all over WebRTC to make it easier to switch clock implementation depending on where the components are used. This is a first step in that direction. Next steps will be to, step by step, move all modules, video engine and voice engine over to the new interface, effectively deprecating the old clock interfaces. Long-term my vision is that we should be able to deprecate the clock of WebRTC and rely on the user providing the implementation. TEST=vie_auto_test, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1041004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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acfdd96ee3d23ad9c77df18523bf6d154deb390e |
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16-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted rtp_rtcp_impl*. BUG= TEST=Trybots. Review URL: https://webrtc-codereview.appspot.com/1035004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3374 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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2f225cadde8627a64d2cede283965bac25a2807c |
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09-Jan-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add logs when no RTCP RR has been received for three regular RTCP intervals. BUG=1267 TEST=Unittest added. Review URL: https://webrtc-codereview.appspot.com/1019006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3346 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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7c894b7cc718773f32d21985ff33a64f9e13946e |
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26-Nov-2012 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up CallStats to provide modules with correct RTT. BUG=769 TEST=Manual test since there is no ViE APi to get RTT for receive channels. Review URL: https://webrtc-codereview.appspot.com/937027 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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571a1c035be6b0afd7f357001bef775c51ec9364 |
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13-Nov-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
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