Commit graph

5 commits

Author SHA1 Message Date
Danil Chapovalov
1a778a24ba Avoid using legacy rtp header parser in the rtp_to_text tool
Bug: None
Change-Id: I4c0ab1ba7730bdcdd826aa41b67b80a96d92c8f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221204
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34231}
2021-06-04 16:41:23 +00:00
Tommi
25eb47ccf1 Make the RtpHeaderParserImpl available to tests and tools only.
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
  (a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
  See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
  sufficient for most production cases.

Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
2019-08-29 15:56:40 +00:00
Yves Gerey
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
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/remote_bitrate_estimator/tools/bwe_rtp.h (Browse further)