Commit graph

15 commits

Author SHA1 Message Date
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
Niels Möller
520ca4e3b8 Delete enum RtpVideoCodecTypes, replaced with VideoCodecType.
Bug: webrtc:8995
Change-Id: I0b44aa26f2f6a81aec7ca1281b8513d8e03228b8
Reviewed-on: https://webrtc-review.googlesource.com/79561
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23507}
2018-06-04 11:53:17 +00:00
Mirko Bonadei
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
Niels Möller
34fa295d47 Delete unused VP8 packetization modes.
Always use the packetization formely known as kEqualSize.
The RTPFragementation header is ignored, which is no change 
in behaviour, since the caller previously always passed null.

Bug: webrtc:6471
Change-Id: Id9e2f985280c2ee8cc33fcf0e5c1fc3ee61c1aff
Reviewed-on: https://webrtc-review.googlesource.com/15222
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20459}
2017-10-27 09:18:17 +00:00
Danil Chapovalov
bef945f36c in Vp8 packetizer use DCHECK instead of assert
Remove redundant null pointer checks.
move header fields validation when they passed in rather when used.
Validate all used fields from the header.

Bug: webrtc:8335
Change-Id: I20b132c6fb8966e49a5414fe757f74d504f4a61a
Reviewed-on: https://webrtc-review.googlesource.com/7400
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20194}
2017-10-06 21:18:14 +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
niklase@google.com
5adc73aad3 git-svn-id: http://webrtc.googlecode.com/svn/trunk@166 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-07-07 08:46:41 +00:00
hlundin@google.com
f0a476bf76 Add PictureID and NonReference to codec information
The PictureID and NonReference information is now routed from the
encoder to the RTP packetizer through CodecSpecificInfo and 
RTPVideoHeaderVP8.
Review URL: http://webrtc-codereview.appspot.com/51003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@155 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-07-07 08:04:23 +00:00
hlundin@google.com
b7686af27c Remove warnings on Windows
Make member variable payload_size_ int instead of unsigned
to avoid warnings when comparing (> and >=).
Review URL: http://webrtc-codereview.appspot.com/40001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@86 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 11:35:59 +00:00
hlundin@google.com
7c53a0c67e Make r80 build on Windows
Re-submitting revision r80, but with bugfix to make it
build on Windows.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@85 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 09:38:28 +00:00
henrika@google.com
f561f488fc Temporary rollback to be able to build on Windows. Will be fixed soon.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@82 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 08:37:18 +00:00
hlundin@google.com
0c32a8d65e VP8 RTP packetizer rewrite
Rewriting the RTP packetizer for VP8 to accommodate more functionality.
This CL does not change the formatting other than that the kStrict
mode now produces equal-sized fragments.
Review URL: http://webrtc-codereview.appspot.com/33006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@80 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-15 07:43:28 +00:00
hlundin@google.com
d2c7bff3a1 Implement VP8 packetizer and unit tests
Implemented a new VP8 packetizer with three modes. The packetizer
class needs access to the fragmentation information, which is
now created in the codec wrapper and passed through the callback
chain to the RTPSenderVideo::SendVP8().

A unit test for the VP8 packetizer was also implemented. It tests the
three different modes. The tests could definitely be more elaborate.
Review URL: http://webrtc-codereview.appspot.com/34003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@48 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-06-07 12:23:14 +00:00
niklase@google.com
77ae29bc81 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d 2011-05-30 11:22:19 +00:00