This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'media ortc p2p':
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I19167714af7cc1436d34cfcba6c8b3718d8e677b
Reviewed-on: https://webrtc-review.googlesource.com/83731
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23638}
candidate keepalive intervals.
StunStats for a STUN candidate cannot be updated after the initial report
in the stats collector. This is caused by the early return of cached
candidate reports for future queries after the initial report creation.
The STUN keepalive interval cannot be configured for UDPPort because of
incorrect type screening, where only StunPort was supported.
TBR=pthatcher@webrtc.org
Bug: webrtc:8951
Change-Id: I0c9c414f43e6327985be6e541e17b5d6f248a79d
Reviewed-on: https://webrtc-review.googlesource.com/58560
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22278}
STUN candidates use STUN binding requests to keep NAT bindings open.
Related stats including packet loss and RTT can be now collected via the
legacy GetStats in PeerConnection.
Bug: None
Change-Id: I7b0eee1ccb07eb670a32ee303c9590047b25f31c
Reviewed-on: https://webrtc-review.googlesource.com/54100
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22113}
STUN candidates use STUN binding requests to keep NAT bindings open. The
interval at which the STUN keepalive pings are sent is configurable now
via RTCConfiguration.
TBR=sakal@webrtc.org
Bug: None
Change-Id: I5f99ea3fe1e9042fa2bf7dcab0aace78f57739e6
Reviewed-on: https://webrtc-review.googlesource.com/54180
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22109}
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}
