Class renamed to indicated use of the rtcp::Empty class: it can only be used as container for other rtcp packets.
All tests that use Append function moved from rtcp_packet_unittest, even if they did not use Empty class.
This is because there is plan to make RtcpPacket class lighter by moving Append functionality to CompoundPacket class.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1582503002
Cr-Commit-Position: refs/heads/master@{#11234}
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1446513002
Cr-Commit-Position: refs/heads/master@{#10823}
to match name given in the RFC5450
private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
to make class usable for parsing packet
Review URL: https://codereview.webrtc.org/1434213004
Cr-Commit-Position: refs/heads/master@{#10636}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
If the buffer becomes full an OnPacketReady callback will be used to
send the packets created so far. On success the buffer can be reused.
The same callback will be called when the last packet has beed created.
Also made some changes to RawPacket. Buffer will now be heap-allocated
rather than (potentially) stack-allocated, but on the plus side it can
now be allocted with variable size and also avoids one memcpy.
BUG=
patch from issue 56429004 at patchset 160001 (http://crrev.com/56429004#ps160001)
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1165113002
Cr-Commit-Position: refs/heads/master@{#9390}
