Commit graph

10 commits

Author SHA1 Message Date
Danil Chapovalov
cbbfd08423 Replace virtual RtcpPacket::SetSenderSsrc with base member
to slightly improve binary size.

Bug: None
Change-Id: I894c7d67a72f4a8077963d2ba0a7bb471a2e7e4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156300
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29428}
2019-10-10 09:14:11 +00:00
Per Kjellander
955f8fd047 Add virtual method rtcp::RtcpPacket::SetSenderSsrc
This will allow RtcpPackets to be sent in a more generic way where the
PacketRouter does not have to know about the type.

App::SetSsrc is replaced with SetSenderSsrc

Bug: webrtc:10742
Change-Id: I9fa18d408250f15818dc6898093d9b116603facb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156166
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29420}
2019-10-09 14:01:53 +00:00
Artem Titov
741daaf039 Move rtc::FunctionView to the public API
Bug: webrtc:10138
Change-Id: Icc25a2a277a9608701aaddd546882366739991ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 15:23:05 +00:00
Johannes Kron
99b9149cee Enable padding bit in TransportFeedback packets
Set padding bit if the TransportFeedback packet contains zero padding.
Also write number of padding elements at the last position of the packet.

Bug: webrtc:10263
Change-Id: I8d17bc0e889f658ac383dec64ddcb95d438c9702
Reviewed-on: https://webrtc-review.googlesource.com/c/122240
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26646}
2019-02-12 11:55:34 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Niels Möller
65ec0fc81e Delete unneeded includes of basictypes.h.
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.

Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.

Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
2018-05-21 19:35:08 +00:00
Danil Chapovalov
5c3cc41cef Change RtcpPacket::PacketReadyCallback to rtc::FunctionView
from interface


Bug: webrtc:5565
Change-Id: I2df5d7a0554b938888581f1c73dbdb8b85c387cc
Reviewed-on: https://webrtc-review.googlesource.com/8680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21136}
2017-12-07 11:20:08 +00:00
Danil Chapovalov
6c170578e6 Move rtcp packet classes from rtp_rtcp to rtp_rtcp_format target
Bug: None
Change-Id: I353228fd5b75bd4fceeaee1bb6fd07b01dac56a1
Reviewed-on: https://webrtc-review.googlesource.com/1480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19867}
2017-09-15 17:36:30 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/modules/rtp_rtcp/source/rtcp_packet.h (Browse further)