Commit Graph

16 Commits

Author SHA1 Message Date
4d92b8467c Ad-hoc rate limiting for UDPPort::SendTo failures.
Bug: chromium:856088
Change-Id: I8b9edd8c7392834a7a88987963de2f8e9d37be11
Reviewed-on: https://webrtc-review.googlesource.com/93881
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24411}
2018-08-23 18:06:57 +00:00
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
00c7183614 Replace rtc::Optional with absl::optional in media, ortc, p2p
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}
2018-06-16 07:09:59 +00:00
6e641e64b2 Signal detailed packet info for each packet sent.
Per-packet info is now signaled in SentPacket to provide useful stats
for bandwidth consumption and overhead analysis in the network stack.

Bug: webrtc:9103
Change-Id: I2b8f6491567d0fa54cc559fc5a96d7aac7d9565e
Reviewed-on: https://webrtc-review.googlesource.com/66281
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22834}
2018-04-12 04:46:06 +00:00
d7d762d08d Remove LOG_J and LOG_JV, tweak p2p logs.
Bug: webrtc:9077
Change-Id: I54ecf10592add33692fc6e694c2f10a646e81345
Reviewed-on: https://webrtc-review.googlesource.com/56142
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22667}
2018-03-29 08:21:27 +00:00
866e08d282 Make rtc::Optional IceConfig parameters interpreted consistently.
The convention is reinforced so that setting a rtc::Optional IceConfig
parameter to null restores the default value. Helper getters are added
to IceConfig to provide either user-defined value or the default.
Shared constants and config defaults used in p2p are moved to
p2pconstants.h/cc for future management with sanity checks.

Bug: webrtc:8993
Change-Id: I976cf1eef5a654b8911f449248bb2f3086279db8
Reviewed-on: https://webrtc-review.googlesource.com/61149
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22575}
2018-03-23 02:10:54 +00:00
4ff5443e4e Fix bugs in collecting STUN candidate stats and configuring STUN
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}
2018-03-04 21:37:21 +00:00
d5e0fcdd97 Add immediate sorting of candidate pairs after the network preference
is configured.

An immediate (re)sorting of candidate paris reduces the latency of
network switching when it is necessary in ICE after (re)configuring the
network preference. A fix of comment and boilerplate code is also
included.

Bug: None
Change-Id: I8685235172d97193ffa6b53d4d2c7796fd01f861
Reviewed-on: https://webrtc-review.googlesource.com/57340
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22197}
2018-02-27 04:07:42 +00:00
72a43a1d2c Collect packet loss and RTT stats of STUN binding requests.
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}
2018-02-21 00:49:26 +00:00
db53f8e604 Add configurable STUN binding request interval.
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}
2018-02-20 23:32:46 +00:00
45cc890560 Assorted logging pedantry
This cl fixes various minor issues found during a quick scan of the current log
usage.

Bug: webrtc:8529
Change-Id: I1e1eb02ef220177dbb327203509736ad7f70cc1c
Reviewed-on: https://webrtc-review.googlesource.com/52262
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21996}
2018-02-13 10:47:24 +00:00
6c38cc7c9e Fix cpplint errors in p2p/
Bug: webrtc:5273
Change-Id: Ida3c2ca1bbb611026f34d7144db9ec24840bf7fb
Reviewed-on: https://webrtc-review.googlesource.com/26444
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21133}
2017-12-06 21:33:36 +00:00
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
1cf1b7d66f Fix clang style warnings in p2p/base/port.h and its subclasses
Bug: webrtc:163
Change-Id: I8308bf1f1b4cf57edd2eb8fda010cb8b667771a2
Reviewed-on: https://webrtc-review.googlesource.com/16361
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20483}
2017-10-30 18:44:09 +00:00
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
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