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16 Commits

Author SHA1 Message Date
Tommi
2497a27b22 Store RtpPacketReceived::arrival_time as Timestamp. 
Previously this value was rounded up to a millisecond value.
This change is complementary to another change:
https://webrtc-review.googlesource.com/c/src/+/216398

Bug: webrtc:12722
Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33928}
 2021-05-05 16:22:33 +00:00
philipel
9465978a3b Remove framemarking RTP extension. 
BUG=webrtc:11637

Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31522}
 2020-06-15 11:18:00 +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
Chen Xing
cd8a6e2f38 Add writing and parsing of the abs-capture-time RTP header extension. 
This change adds the writing and parsing of the `abs-capture-time` RTP header extension defined at:

  http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time

We are still missing the code to:

- Negotiate the header extension.
- Collect capture time for audio and video and have the info sent with the header extension.
- Receive the header extension and use its info.

Bug: webrtc:10739
Change-Id: I75af492e994367f45a5bdc110af199900327b126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144221
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28468}
 2019-07-03 14:07:36 +00:00
Johannes Kron
54047bea1b Reland "Extend TransportSequenceNumber RTP header extension" 
This reverts commit 109b5fb5f5b2f46e1798c91c4a024ce26f57f0b0.

Reason for revert: The failing libfuzzer was fixed in commit d6c6f16063b81fc60206618ba06198e34ee0eacb

Original change's description:
> Revert "Extend TransportSequenceNumber RTP header extension"
> 
> This reverts commit 28c7362bc485d22bdc8c744bc725022780187a96.
> 
> Reason for revert: It breaks Linux64 Release (libfuzzer):
> https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket.appspot.com/8921003137877469920/+/steps/compile/0/stdout
> 
> Original change's description:
> > Extend TransportSequenceNumber RTP header extension
> > 
> > Extend TransportSequenceNumber RTP header extension to support
> > feedback on sender request.
> > 
> > Bug: webrtc:10262
> > Change-Id: Ibc1cf18162d15cd102e951c9dc697d8ea536ebb6
> > Reviewed-on: https://webrtc-review.googlesource.com/c/123233
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#26766}
> 
> TBR=danilchap@webrtc.org,aleloi@webrtc.org,kron@webrtc.org
> 
> Change-Id: Ie8a73f5fdffd99919ceaa1ae8911a1645f2077e9
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10262
> Reviewed-on: https://webrtc-review.googlesource.com/c/123522
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26767}

TBR=danilchap@webrtc.org,mbonadei@webrtc.org,aleloi@webrtc.org,kron@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:10262
Change-Id: I0f854299a46c042cfbdf8b8cc8cd965a228142c8
Reviewed-on: https://webrtc-review.googlesource.com/c/123764
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26798}
 2019-02-21 16:01:30 +00:00
Mirko Bonadei
109b5fb5f5 Revert "Extend TransportSequenceNumber RTP header extension" 
This reverts commit 28c7362bc485d22bdc8c744bc725022780187a96.

Reason for revert: It breaks Linux64 Release (libfuzzer):
https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket.appspot.com/8921003137877469920/+/steps/compile/0/stdout

Original change's description:
> Extend TransportSequenceNumber RTP header extension
> 
> Extend TransportSequenceNumber RTP header extension to support
> feedback on sender request.
> 
> Bug: webrtc:10262
> Change-Id: Ibc1cf18162d15cd102e951c9dc697d8ea536ebb6
> Reviewed-on: https://webrtc-review.googlesource.com/c/123233
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26766}

TBR=danilchap@webrtc.org,aleloi@webrtc.org,kron@webrtc.org

Change-Id: Ie8a73f5fdffd99919ceaa1ae8911a1645f2077e9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10262
Reviewed-on: https://webrtc-review.googlesource.com/c/123522
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26767}
 2019-02-20 13:11:54 +00:00
Johannes Kron
28c7362bc4 Extend TransportSequenceNumber RTP header extension 
Extend TransportSequenceNumber RTP header extension to support
feedback on sender request.

Bug: webrtc:10262
Change-Id: Ibc1cf18162d15cd102e951c9dc697d8ea536ebb6
Reviewed-on: https://webrtc-review.googlesource.com/c/123233
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26766}
 2019-02-20 12:23:45 +00:00
Johannes Kron
09d6588d93 Change HdrMetadataExtension to ColorSpaceExtension 
Bug: webrtc:8651
Change-Id: Ica6f8c6bd13bb07f89700b9c0a359b9a58feefbb
Reviewed-on: https://webrtc-review.googlesource.com/c/111758
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25800}
 2018-11-27 14:05:31 +00:00
Johannes Kron
ad1d9f0d78 Add RTP header extension for HDR metadata 
Bug: webrtc:8651
Change-Id: I1c956eaac1532ac0d3820084edb4054a4cc9c68d
Reviewed-on: https://webrtc-review.googlesource.com/c/109924
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25578}
 2018-11-09 11:10:12 +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
Johnny Lee
e0c8b230e7 Frame marking RTP header extension (PART 1: implement extension) 
Bug: webrtc:7765
Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb
Reviewed-on: https://webrtc-review.googlesource.com/85200
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24695}
 2018-09-11 22:35:30 +00:00
Dino Radaković
9e24cb344a Add move constructors and assignment operators to RtpPacketReceived and RtpPacketToSend. Since both are non-POD now, move would fall back to copy without these. 
Bug: webrtc:8935
Change-Id: I270e7daf68aa00411ad5ae00da739292600043f2
Reviewed-on: https://webrtc-review.googlesource.com/57621
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22186}
 2018-02-26 13:25:50 +00:00
Karl Wiberg
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
 2017-11-22 11:21:47 +00:00
Dino Radaković
21360eb01e Add application extension field to RtpPacketReceived. 
Bug: webrtc:8439
Change-Id: I372e90c81a68351d343554fb77ce6ef77d538e62
Reviewed-on: https://webrtc-review.googlesource.com/14820
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20410}
 2017-10-24 14:22:18 +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