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12 Commits

Author SHA1 Message Date
Ilya Nikolaevskiy
7a82467d0d Fix vp9 svc singlecast mode and enable quality scaler for vp9 
1) Fix several typos and small mistakes which could lead to crashes
2) Adjust bitrates if leading layers are disabled
3) Wire up webrtc quality scaler

Bug: webrtc:11319
Change-Id: I16e52bdb1c315d64906288e4f2be55fe698d5ceb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177525
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31546}
 2020-06-18 21:24:46 +00:00
Elad Alon
a5c0ba1861 Reland "Fix LibvpxVp8Encoder::FrameDropThreshold" 
This is a reland of 159e53a66eceb79003741ba0d990c502595c27bb
Reason for reland: Got Aliby for Android FEC test flakes.

Original change's description:
> Fix LibvpxVp8Encoder::FrameDropThreshold
>
> Bug: webrtc:10387
> Change-Id: I17b575546d6718f905429ac45705cb923ee52c10
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126232
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27032}

Bug: webrtc:10387
Change-Id: Iaed760464d71ccfc6ad0e442b99f20c40c03e1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126762
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27058}
 2019-03-11 13:37:38 +00:00
Yves Gerey
55500d26b7 Revert "Fix LibvpxVp8Encoder::FrameDropThreshold" 
This reverts commit 159e53a66eceb79003741ba0d990c502595c27bb.

Reason for revert: Might break downstream Android projects.

Original change's description:
> Fix LibvpxVp8Encoder::FrameDropThreshold
> 
> Bug: webrtc:10387
> Change-Id: I17b575546d6718f905429ac45705cb923ee52c10
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126232
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27032}

TBR=ilnik@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org

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

Bug: webrtc:10387
Change-Id: I08055d1b0234dccd4166a4eaf8eff56f909f99a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126721
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27055}
 2019-03-11 12:16:09 +00:00
Elad Alon
159e53a66e Fix LibvpxVp8Encoder::FrameDropThreshold 
Bug: webrtc:10387
Change-Id: I17b575546d6718f905429ac45705cb923ee52c10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126232
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27032}
 2019-03-08 11:14:43 +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
Niels Möller
de8e6e6db3 Refactor bitrate configuration in CallTest 
All implementations of ModifyReceiverCallConfig and
ModifySenderCallConfig configure the bitrate_config member only. So
replace these methods by ModifyReceiverBitrateConfig and
ModifySenderBitrateConfig.

This is a preparation for injecting RtpTransportControllerSend via
CallConfig. Then bitrates should be passed when constructing
RtpTransportControllerSend, and they can be deleted from CallConfig.

Bug: webrtc:7135
Change-Id: I6714158bd463dd485018713d0e26815919e5afcc
Reviewed-on: https://webrtc-review.googlesource.com/c/110780
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25624}
 2018-11-13 16:03:00 +00:00
Danil Chapovalov
99b71dfd4a Use function_video_(en|de)coder_factory from api 
Remove them from test.
It is completion of the move started with
https://webrtc-review.googlesource.com/c/src/+/107705

Bug: None
Change-Id: Ib0b26db04a1ee814322851280ba1e59b4b3f7ce6
Reviewed-on: https://webrtc-review.googlesource.com/c/107891
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25392}
 2018-10-26 14:54:19 +00:00
Åsa Persson
8c1bf9595a Reland "Add initial support for RtpEncodingParameters max_framerate." 
This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad.

Reason for revert: downstream project fixed.

Original change's description:
> Revert "Add initial support for RtpEncodingParameters max_framerate."
>
> This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.
>
> Reason for revert: Breaks downstream project.
>
> Original change's description:
> > Add initial support for RtpEncodingParameters max_framerate.
> >
> > Add support to set the framerate to the maximum of |max_framerate|.
> > Different framerates are currently not supported per stream for video.
> >
> > Bug: webrtc:9597
> > Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> > Reviewed-on: https://webrtc-review.googlesource.com/92392
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24270}
>
> TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org
>
> Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9597
> Reviewed-on: https://webrtc-review.googlesource.com/94060
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24277}

TBR=steveanton@webrtc.org,mbonadei@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Bug: webrtc:9597
Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e
Reviewed-on: https://webrtc-review.googlesource.com/100080
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24720}
 2018-09-13 10:06:33 +00:00
Mirko Bonadei
948b7e3755 Revert "Add initial support for RtpEncodingParameters max_framerate." 
This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.

Reason for revert: Breaks downstream project.

Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
> 
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
> 
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}

TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
 2018-08-14 07:25:23 +00:00
Åsa Persson
ced5cfdb35 Add initial support for RtpEncodingParameters max_framerate. 
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.

Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
 2018-08-13 09:59:04 +00:00
Sebastian Jansson
7258224c3b Replaces call config create in tests with modify. 
This ensures the event logs in CallTest will be used by default.

Bug: webrtc:9510
Change-Id: I9df82b5ef39e1b2cba2789f8c5c7a9fed3c4c2f6
Reviewed-on: https://webrtc-review.googlesource.com/88562
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23970}
 2018-07-13 12:52:45 +00:00
Åsa Persson
2027b669b2 Add tests for quality scaling. 
Tests:
- Adapts down due to high QP.
- Adapts down initially due to low bitrate.

Bug: webrtc:9169
Change-Id: Ifcfc07ef6860d4dc3ede54333a56ba313e2f09d5
Reviewed-on: https://webrtc-review.googlesource.com/73160
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23108}
 2018-05-04 07:51:04 +00:00