| 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_packet_history.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/rtp_packet_history.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_packet_history.h
|
| affa39cb39c77408109fef691533021533d969e1 |
|
21-Oct-2015 |
sprang <sprang@webrtc.org> |
Remove time constraint on first retransmit of a packet.
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.
Review URL: https://codereview.webrtc.org/1414563003
Cr-Commit-Position: refs/heads/master@{#10362}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
|
| e23e737177cf5d131a6d4a4d229aa513c5270a59 |
|
08-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Disable pacer disabling.
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
|
| c32d2db69bc94480ecb312268c6e6769d4a1cac6 |
|
11-Sep-2015 |
pbos <pbos@webrtc.org> |
Refactor RTPPacketHistory to use a packet struct.
Collects packet information within a struct instead of spreading it out
over different vectors. Adds a fixed-size buffer to the stored packet
instead of using vectors.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1340573002
Cr-Commit-Position: refs/heads/master@{#9926}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.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_packet_history.h
|
| c957ffc6dc36879e5ad72d7f0af2a014707d70fa |
|
02-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Fixed potential crash if rtp packet history is completely full.
Also performance enhanecement in rtp_sender (don't lookup if kDontStore)
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39759004
Cr-Commit-Position: refs/heads/master@{#8226}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8226 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
|
| 43c883954f5edc84bd8e0e901ef770fead218ed5 |
|
29-Jan-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Allow rtp packet history to dynamically expand in size.
When using the paced sender, packets will be put into the rtp packet
history and then retreived from there again when it is time to send.
In some cases (low send bitrate and very large frames created) this
may overflow, causing packets to be overwritten in the packet history
before they have been sent.
Check this condition and expand history size if needed.
This is primarily triggered during screenshare, when
switching to a large picture with lots of high frequency
details in it.
BUG=4171
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34879004
Cr-Commit-Position: refs/heads/master@{#8195}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.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_packet_history.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_packet_history.h
|
| 38344ed2806c8fed60d67d280ca44c32e36707c0 |
|
24-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Move thread_annotations.h to webrtc/base/.
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
|
| 420b2567f38241099907d30d8bece1c4db50262d |
|
30-May-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers.
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.
BUG=1811
R=asapersson@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12609005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
|
| 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/rtp_packet_history.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_packet_history.h
|
| a048d7cb0a5bad5ca49bbcc5273cb4cca28c1710 |
|
29-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in rtp_rtcp/
BUG=1662
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1557004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.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_packet_history.h
|
| 20ed36dada62ad56ec01263fc0eef0ed198f6476 |
|
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_packet_history.h
|
| e5b49a0472b97fa262b641b78cf4230bd824296f |
|
06-Nov-2012 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update timestamp offset for re-transmitted packets.
BUG=1059
Review URL: https://webrtc-codereview.appspot.com/930011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3046 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
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_packet_history.h
|