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14 Commits

Author SHA1 Message Date
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
Sebastian Jansson
7a994334f1 Avoids crash in VCMJitterEstimator due to incorrectly calculated mean FPS. 
Bug: webrtc:10728
Change-Id: I9e50ebdcfd9ef7d408cbf00111f29f0c80fd6c75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141402
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28226}
 2019-06-11 12:04:52 +00:00
“Michael
e0f370471a Add cap to video jitter buffer size/latency in experiment branches only. 
Bug: webrtc:10664
Change-Id: I03762c8b318f26f2689e89545aa8cc8e5b4a4329
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138081
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28155}
 2019-06-04 15:50:27 +00:00
Stefan Holmer
4ed7e511f6 Revert "Add ability to cap the video jitter estimate to a max value." 
This reverts commit a8ae407a480a2a9982eecf9e3a9b10da5373cd9a.

Reason for revert: This CL incorrectly affects non-experiment branch.  A new CL affecting only the experiment will be uploaded.

Original change's description:
> Add ability to cap the video jitter estimate to a max value.
>
> Bug: webrtc:10572
> Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27744}

TBR=stefan@webrtc.org,mhoro@webrtc.org

Bug: webrtc:10572
Change-Id: I4af334168ca70ecfae7fd18fc7c852819a98d866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138063
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28025}
 2019-05-22 15:07:33 +00:00
Niels Möller
c5ba5e9572 Delete unused methods of VCMJitterEstimator 
ResetNackCount and UpdateMaxFrameSize were unused.

Bug: None
Change-Id: I314b3edce368ee3230bc3510e1bba520806d1493
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134201
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27789}
 2019-04-26 15:21:09 +00:00
“Michael
a8ae407a48 Add ability to cap the video jitter estimate to a max value. 
Bug: webrtc:10572
Change-Id: I21112824dc02afa71db61bb8c2f02723e8b325b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133963
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27744}
 2019-04-24 16:03:15 +00:00
Åsa Persson
3fcc5be59d Remove unused members in VCMJitterEstimator. 
Bug: none
Change-Id: I0b6649906d4e73ef0819e00884b5a17d317c7619
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131260
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27484}
 2019-04-08 12:27:47 +00:00
Sebastian Jansson
aa01f27667 Removes all const Clock*. 
This prepares for making the Clock interface fully mutable.

Calls to the time functions in Clock can have side effects in some
circumstances. It's also questionable if it's a good idea to allow
repeated calls to a const method return different values without
any changed to the class instance.

Bug: webrtc:9883
Change-Id: I96fb9230705f7c80a4c0702132fd9dc73899fc5e
Reviewed-on: https://webrtc-review.googlesource.com/c/120347
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26467}
 2019-01-30 13:03:37 +00:00
Yves Gerey
3e70781361 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. 
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.

Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
 2018-11-28 18:25:07 +00:00
Erik Språng
b1e031a156 JitterEstimator: Remove old LowRate exp and add trial for upper bound. 
The frame time deltas are now capped based on the current noise.
This has been tested in various conditions using both screen content
and typical mobile video settings, to produce delays that are not overly
high screen content, and simultaneously not negatively affect mobile
calls on really bad network that may have high natural jitter.

Bug: webrtc:9898
Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48
Reviewed-on: https://webrtc-review.googlesource.com/c/107349
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25469}
 2018-11-01 14:51:38 +00:00
Gustavo Garcia
f7a7c8a8bc Stop adding RTT delay if there was not packet loss for enough time 
Bug: none
Change-Id: I9105c0ee6a4f75381e133a496c3728dece9aca4b
Reviewed-on: https://webrtc-review.googlesource.com/c/23261
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25309}
 2018-10-23 10:38:39 +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