Also refactor packet router to use a map rather than iterate over all
rtp modules for each packet sent.
BUG=webrtc:4311
Review URL: https://codereview.webrtc.org/1247293002
Cr-Commit-Position: refs/heads/master@{#9670}
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}
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}
Clang version changed 223108:230914
Details: 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
This should fix the following error I'm seeing in Win8 GN trybot:
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(78)
: error C2220: warning treated as error - no 'object' file generated
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\bitrate_controller\bitrate_controller_impl.cc(30)
: warning C4373:
'webrtc::BitrateControllerImpl::RtcpBandwidthObserverImpl::OnReceivedEstimatedBitrate':
virtual function overrides 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate',
previous versions of the compiler did not override when parameters only differed by const/volatile qualifiers
e:\b\build\slave\win_gn\build\src\third_party\webrtc\modules\rtp_rtcp\interface\rtp_rtcp_defines.h(286)
: see declaration of 'webrtc::RtcpBandwidthObserver::OnReceivedEstimatedBitrate'
http://build.chromium.org/p/tryserver.chromium.win/builders/win8_chromium_gn_dbg/builds/23/steps/compile/logs/stdio
The above was triggered in CL https://codereview.chromium.org/802113002/
BUG=None
R=kjellander@google.com, kjellander@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37409004
Patch from Thiago Farina <tfarina@chromium.org>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7911 4adac7df-926f-26a2-2b94-8c16560cd09d
Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26399004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
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
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
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)
BUG=
TEST=Trybots, vie_ & voe_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/998007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.
In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.
BUG=
TEST=vie/voe_auto_test, trybots
Review URL: https://webrtc-codereview.appspot.com/1001006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
