Commit graph

25 commits

Author SHA1 Message Date
Peter Thatcher
4a2e0e5d45
Update to 4896 (M100) (#72) 2022-04-15 17:13:23 -06:00
Artem Titov
913cfa76ec Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp
Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
2021-08-09 15:51:03 +00:00
Danil Chapovalov
26e1b7ac01 Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
Bug: webrtc:11152
Change-Id: I1a6225701ecd6f7a34c946d7296f0ab0cbb5eaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165342
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30190}
2020-01-09 12:10:19 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Yves Gerey
988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00
Danil Chapovalov
fa5ec8d20d Use signed integers for limiting packet size in video packetizers
Using signed integers allows to centralize checking of edge cases
in RtpPacketizer::SplitAboutEqually and
reduce chance of hitting issues with size_t underflow

Bug: webrtc:9680
Change-Id: Ic05bf0a9565a277c4608f43061ca46cf44e82d08
Reviewed-on: https://webrtc-review.googlesource.com/98602
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24618}
2018-09-07 09:24:18 +00:00
Danil Chapovalov
376e1147e6 in RtpPacketizerVp8 factor out payload splitter function
so that it can be shared between different packetizers
and thus easier to extend

Bug: webrtc:9680
Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f
Reviewed-on: https://webrtc-review.googlesource.com/97661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24555}
2018-09-04 15:28:05 +00:00
Danil Chapovalov
8d1b582f33 Cleanup RtpPacketizerVp8
Merge SetPayloadData into constructor,
Remove payload size member because now used only during construction.
Remove member that should be constant

Bug: None
Change-Id: Ib2083439f466ad9151ce8e54fceede6cef51d955
Reviewed-on: https://webrtc-review.googlesource.com/96740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24491}
2018-08-30 10:16:33 +00:00
Danil Chapovalov
f7f8a1f176 Cleanup RtpPacketizer interface
merge construction and call to SetPayloadData
Add NumPackets instead of SetPayloadData
Remove virtual ToString() as unused
move CHECK(rtp_video_header) from RtpPacketizer::Create to RtpSenderVideo::SendVideo

Bug: webrtc:9680
Change-Id: I074644e048c797eb836f79979df363fe1ea0075e
Reviewed-on: https://webrtc-review.googlesource.com/96543
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24474}
2018-08-29 08:44:08 +00:00
Niels Möller
a12c42a6b2 Delete root header file typedef.h.
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +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
Danil Chapovalov
2a5ce2bcf8 Fix clang style errors in rtp_rtcp and dependant targets
Mark functions with override instead of virtual.
Add explicit non-trivial constructors/assign operators/destructors.
Define them in .cc files instead of inlining
use auto* instead of auto when deduced type is raw pointer

Bug: webrtc:163
Change-Id: I4d8a05d6a64fcc2ca16d02c5fcf9488fda832a6d
Reviewed-on: https://webrtc-review.googlesource.com/48781
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21927}
2018-02-07 09:48:28 +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
Mirko Bonadei
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +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