Commit Graph

8 Commits

Author SHA1 Message Date
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
942bc2e4b9 Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal.
|packet_overhead| field is added to rtc::NetworkRoute structure.

In PackTransportInternal:
1. network_route() is added which returns the current network route.
2. debug_name() is removed.
3. transport_name() is moved from DtlsTransportInternal and
IceTransportInternal to PacketTransportInternal.

When the selected candidate pair is changed, the P2PTransportChannel
will fire the SignalNetworkRouteChanged instead of
SignalSelectedCandidatePairChanged to upper layers.

The Rtp/SrtpTransport takes the responsibility of calculating the
transport overhead from the BaseChannel so that the BaseChannel
doesn't need to depend on P2P layer transports.

TBR=pthatcher@webrtc.org

Bug: webrtc:7013
Change-Id: If9928b25a7259544c2d9c42048b53ab24292fc67
Reviewed-on: https://webrtc-review.googlesource.com/22767
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20664}
2017-11-13 22:50:11 +00:00
8c316c1a89 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal."
This reverts commit 71677452f9cf210aa98162c6f4bd8d339e625337.

Reason for revert: Broke Chromium.

Original change's description:
> Replaced the SignalSelectedCandidatePairChanged with a new signal.
> 
> |transport overhead| field is added to rtc::NetworkRoute structure.
> 
> In PackTransportInternal:
> 1. network_route() is added which returns the current network route.
> 2. debug_name() is removed.
> 3. transport_name() is moved from DtlsTransportInternal and
>    IceTransportInternal to PacketTransportInternal.
> 
> When the selected candidate pair is changed, the P2PTransportChannel
> will fire the SignalNetworkRouteChanged instead of
> SignalSelectedCandidatePairChanged to upper layers.
> 
> The Rtp/SrtpTransport takes the responsibility of calculating the
> transport overhead from the BaseChannel so that the BaseChannel
> doesn't need to depend on P2P layer transports.
> 
> Bug: webrtc:7013
> Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c
> Reviewed-on: https://webrtc-review.googlesource.com/13520
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20661}

TBR=steveanton@webrtc.org,zhihuang@webrtc.org,pthatcher@webrtc.org

Change-Id: Ie0c76786855b65bb8caba7065593c961e4bf9de7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7013
Reviewed-on: https://webrtc-review.googlesource.com/22764
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20662}
2017-11-13 21:13:55 +00:00
71677452f9 Replaced the SignalSelectedCandidatePairChanged with a new signal.
|transport overhead| field is added to rtc::NetworkRoute structure.

In PackTransportInternal:
1. network_route() is added which returns the current network route.
2. debug_name() is removed.
3. transport_name() is moved from DtlsTransportInternal and
   IceTransportInternal to PacketTransportInternal.

When the selected candidate pair is changed, the P2PTransportChannel
will fire the SignalNetworkRouteChanged instead of
SignalSelectedCandidatePairChanged to upper layers.

The Rtp/SrtpTransport takes the responsibility of calculating the
transport overhead from the BaseChannel so that the BaseChannel
doesn't need to depend on P2P layer transports.

Bug: webrtc:7013
Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c
Reviewed-on: https://webrtc-review.googlesource.com/13520
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20661}
2017-11-13 20:57:31 +00:00
33f69db1c8 Fix clang style warnings in p2p/base/packettransport files
Bug: webrtc:163
Change-Id: I10fb72c1ae01b6715f4d13e43860c80a6dfc9d87
Reviewed-on: https://webrtc-review.googlesource.com/16362
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20485}
2017-10-30 19:51:49 +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