| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.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/rtcp_packet.h
|
| 162abd3562d7b08ab36569800d757b52739b9249 |
|
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint whitespace warning removed from most rtp_rtcp/source/ files
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| 5eb4988c0ac0665701e9bccba0fad3dcadfcfcd0 |
|
09-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint build/header_guard errors fixed
BUG=webrtc:5277
R=mflodman
Review URL: https://codereview.webrtc.org/1506043003
Cr-Commit-Position: refs/heads/master@{#10949}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| 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.h
|
| 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.h
|
| f7c5776d4254e31e51107388a05c66d14108a8af |
|
04-Dec-2015 |
Erik Språng <sprang@webrtc.org> |
Refactorings to send RTCP packets directly via the RtcpPacket callback, with some simplifications enabled by this. NACK now also sent via RtcpPacket.
BUG=webrtc:2450
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1309833002 .
Cr-Commit-Position: refs/heads/master@{#10888}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.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/rtcp_packet.h
|
| 6b8d3551681f40b880507cecc88f478a12383cc7 |
|
24-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/
The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc
BUG=webrtc:4173
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1362303002 .
Cr-Commit-Position: refs/heads/master@{#10052}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| c9bbeb03542cffc14b7d306e5f88b6c0e593864d |
|
23-Sep-2015 |
Erik Språng <sprang@webrtc.org> |
Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29
Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}
TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1362923002 .
Cr-Commit-Position: refs/heads/master@{#10029}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| ef165eefc79cf28bb67779afe303cc2365885547 |
|
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Wire up send-side bandwidth estimation.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1338203003
Cr-Commit-Position: refs/heads/master@{#10012}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| 3c089d751ede283e21e186885eaf705c3257ccd2 |
|
16-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to contructormagic 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.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.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/rtcp_packet.h
|
| 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.h
|
| 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.h
|
| 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.h
|
| 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.h
|