| 2f7dea164dc49ae8a0322e3c9edb1dd23266c664 |
|
13-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Empty moved into own file and renamed to CompoundPacket on the way
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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 92e677a1f8d24dfa0031d307c4a7d8e530cd4eb4 |
|
12-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1551893002
Cr-Commit-Position: refs/heads/master@{#11228}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 7e8145f05d5f6921ffca3d62e9c4d1301c1d8bcb |
|
11-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Tmmbr moved into own file
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1575023002
Cr-Commit-Position: refs/heads/master@{#11206}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| ef3d805f6e50bc488f8e4e9e353068b78c73d17f |
|
11-Jan-2016 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] rtcp::Tmmbn moved into own file
explicetly unchanged.
BUG=webrtc:5260
R=åsapersson
Review URL: https://codereview.webrtc.org/1578713002
Cr-Commit-Position: refs/heads/master@{#11201}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| a8890a57a5d03f942924ff61d3c62244f2bdab10 |
|
22-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Nack packet moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1461623003
Cr-Commit-Position: refs/heads/master@{#11111}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 54999d411b97e3df54121e5f7bfb28846f3c8086 |
|
16-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Dlrr block moved into own file and got Parse function
BUG=webrtc:5260
Review URL: https://codereview.webrtc.org/1453973005
Cr-Commit-Position: refs/heads/master@{#11044}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 91941ae493cb37a4e1250c7d3aad1c7394b5850e |
|
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::VoipMetric block moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1452733002
Cr-Commit-Position: refs/heads/master@{#11030}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 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/rtcp_packet.cc
|
| fc47ed6c0524d7ee0bc7947f0ad65fcefda34a29 |
|
07-Dec-2015 |
Danil Chapovalov <danilchap@webrtc.org> |
rtcp::Rrtr block moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org, åsapersson
Review URL: https://codereview.webrtc.org/1496883002 .
Cr-Commit-Position: refs/heads/master@{#10912}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 97f7e13c23ddb26543f33bce944d501e58d1dd9b |
|
04-Dec-2015 |
Danil Chapovalov <danilchap@webrtc.org> |
rtcp::ReceiverReport moved into own file and got Parse function
BUG=webrtc:5260
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1453083002 .
Cr-Commit-Position: refs/heads/master@{#10897}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| f8385aded0943c7889d6e9b92f3c0978f3657bb2 |
|
27-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Pli moved into own file and got a Parse function
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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 50c5136cb2ad11eb9ba3df1a1d54d527c8a0dc77 |
|
22-Nov-2015 |
danilchap <danilchap@webrtc.org> |
RTCP Bye packet moved to own file
Bye class got support for Parsing
Reason field implemented
Review URL: https://codereview.webrtc.org/1430013003
Cr-Commit-Position: refs/heads/master@{#10741}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 0219c9b4bfcbb778137756210eb95f40d936cc66 |
|
18-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::App moved into own file and got Parse function
Review URL: https://codereview.webrtc.org/1437353003
Cr-Commit-Position: refs/heads/master@{#10688}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| f8506cbdd88ce538d9e6c28ee39111345189778f |
|
13-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::Ij renamed to rtcp::ExtendedJitterReport
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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| df948f03b34dc652c2b3a944535fc01ec22395ce |
|
13-Nov-2015 |
danilchap <danilchap@webrtc.org> |
rtcp::ReportBlock refactored to contain parsing
Review URL: https://codereview.webrtc.org/1420283022
Cr-Commit-Position: refs/heads/master@{#10633}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| ebc0b4e99365443111857a0c7cfcc8944d8f1b6e |
|
28-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h for rtp_rtcp.
BUG=webrtc:5118
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1422023002 .
Cr-Commit-Position: refs/heads/master@{#10437}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 73a93e82579d6eeb3a1c4a63ef4b64c3c4d9bb18 |
|
14-Sep-2015 |
sprang <sprang@webrtc.org> |
Add a ParseHeader method to RtcpPacket, for parsing common RTCP header.
Also refactor TransportFeedback to use this.
BUG=
Review URL: https://codereview.webrtc.org/1307663004
Cr-Commit-Position: refs/heads/master@{#9935}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| a3b8769860bdb0a45dbff6d1e0092486fa59aaa4 |
|
29-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Add packetization and coding/decoding of feedback message format.
BUG=webrtc:4312
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1175263002 .
Cr-Commit-Position: refs/heads/master@{#9651}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| bdc0b0d869e9a14bbfafcbb84e294a13383e6fa6 |
|
22-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
Use RtcpPacket classes for SenderReport/ReceiveReport in RTCPSender
BUG=2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1170723002.
Cr-Commit-Position: refs/heads/master@{#9483}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| c1b9d4e686c184e4b1779d442d447128477d3b8b |
|
08-Jun-2015 |
Erik Språng <sprang@webrtc.org> |
Add support for fragmentation in RtcpPacket.
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}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 779c3d16b9623f38a72439bc013102aeb0077a62 |
|
17-Mar-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Use ByteReader/ByteWriter instead of rtputility and manual shift/add.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41289004
Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| c891fee7aba4e6bcc33f6e03ec9e7f3a2940e03c |
|
13-Aug-2014 |
fbarchard@google.com <fbarchard@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make a int64 constant use ULL suffix so it wont get truncated.
BUG=3690
TESTED=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 62bafae6618fe3aefbd18657062abc98a40c3375 |
|
08-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some refactoring inside rtp_rtcp/.
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.
BUG=
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 3b84b3a58cf4093204749fa7ba782f49b8934246 |
|
25-Jun-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add RTCP packet types to packet builder:
REMB, TMMBR, TMMBN and
extended reports: RRTR, DLRR, VoIP metric.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9299005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6537 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 4b12d400089f324293b8c313ba8257d9247e9cc6 |
|
16-Jun-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19559004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6449 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| a826006132b3606b7325befcbffd608df6714f6c |
|
20-May-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add NACK and RPSI packet types to RTCP packet builder.
Fixes bug found when parsing received RPSI packet.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| dc80bae2a62a1bdbe0d342b3260a7e5b2cb958df |
|
08-Apr-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
Clean some logs and add asserts in the way.
BUG=3153
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|
| 0f2809a5ac5477a6134ebafb4680597252f8a5c5 |
|
21-Feb-2014 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add RTCP packet class.
Adds packet types: sr, rr, bye, fir.
BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
|