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9cad4dccc96f09d76dce3766a076052df7d75ab8
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108 Commits

Author SHA1 Message Date
Johannes Kron
9cad4dccc9 Reland "Reland "Reland "Distinguish between send and receive video codecs""" 
This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c

This CL lands all code except the code that activates the change,
see media/engine/webrtc_video_engine.cc
Once downstream projects are fixed, there will be a one-line change to
activate the change to distinguish between send and receive video codecs.

Original change's description:
> Reland "Reland "Distinguish between send and receive video codecs""
>
> This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
>
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> >
> > Reason for revert: Downstream test updated.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive video codecs""
> > >
> > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > >
> > > Reason for revert: Breaks another downstream test.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > >
> > > > Reason for revert: Downstream tests have been updated.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > >
> > > > > Reason for revert: Breaks downstream test.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive video codecs
> > > > > >
> > > > > > Even though send and receive codecs are the same,
> > > > > > they might have different support in HW.
> > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30079}
> >
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30097}
>
> Bug: chromium:1029737
> Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30120}

Bug: chromium:1029737
Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30219}
 2020-01-10 23:37:11 +00:00
Mirta Dvornicic
873610ca68 Fix updating degradation preference in SetRtpParameters. 
Degradation preference could be changed before video send stream
is configured which would cause a crash.

Bug: None
Change-Id: If970e66fba0b9fdb9da789066861d919874de119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164463
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30141}
 2020-01-03 11:16:54 +00:00
Olga Sharonova
b5159fe4a7 Revert "Reland "Reland "Distinguish between send and receive video codecs""" 
This reverts commit 4e64e605894df287178c5a1b537fbe859b7d420c.

Reason for revert: breaks a bunch of WebRtcBrowserTests on Win: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/4843


Original change's description:
> Reland "Reland "Distinguish between send and receive video codecs""
> 
> This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
> 
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> >
> > Reason for revert: Downstream test updated.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive video codecs""
> > >
> > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > >
> > > Reason for revert: Breaks another downstream test.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > >
> > > > Reason for revert: Downstream tests have been updated.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > >
> > > > > Reason for revert: Breaks downstream test.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive video codecs
> > > > > >
> > > > > > Even though send and receive codecs are the same,
> > > > > > they might have different support in HW.
> > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30079}
> >
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30097}
> 
> Bug: chromium:1029737
> Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30120}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

Change-Id: I709ee0eb6246aa79dde3aacfc4c47e070c4e90ea
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162904
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30122}
 2019-12-20 13:57:12 +00:00
Johannes Kron
4e64e60589 Reland "Reland "Distinguish between send and receive video codecs"" 
This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228

Original change's description:
> Reland "Distinguish between send and receive video codecs"
>
> This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
>
> Reason for revert: Downstream test updated.
>
> Original change's description:
> > Revert "Reland "Distinguish between send and receive video codecs""
> >
> > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> >
> > Reason for revert: Breaks another downstream test.
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > >
> > > Reason for revert: Downstream tests have been updated.
> > >
> > > Original change's description:
> > > > Revert "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > >
> > > > Reason for revert: Breaks downstream test.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive video codecs
> > > > >
> > > > > Even though send and receive codecs are the same,
> > > > > they might have different support in HW.
> > > > > Distinguish between send and receive codecs to be able to keep
> > > > > track of which codecs have HW support.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30078}
> >
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30079}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: chromium:1029737
> Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30097}

Bug: chromium:1029737
Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30120}
 2019-12-20 11:44:42 +00:00
Ilya Nikolaevskiy
f9d92ed2c8 Revert "Reland "Distinguish between send and receive video codecs"" 
This reverts commit 77eb338ae48acb0cb1437da05d86941bb4063228.

Reason for revert: Speculative revert, as it seems to have broken webrtc-importer

Original change's description:
> Reland "Distinguish between send and receive video codecs"
> 
> This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> 
> Reason for revert: Downstream test updated.
> 
> Original change's description:
> > Revert "Reland "Distinguish between send and receive video codecs""
> > 
> > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > 
> > Reason for revert: Breaks another downstream test.
> > 
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > > 
> > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > 
> > > Reason for revert: Downstream tests have been updated.
> > > 
> > > Original change's description:
> > > > Revert "Distinguish between send and receive video codecs"
> > > > 
> > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > 
> > > > Reason for revert: Breaks downstream test.
> > > > 
> > > > Original change's description:
> > > > > Distinguish between send and receive video codecs
> > > > > 
> > > > > Even though send and receive codecs are the same,
> > > > > they might have different support in HW.
> > > > > Distinguish between send and receive codecs to be able to keep
> > > > > track of which codecs have HW support.
> > > > > 
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > 
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > 
> > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > 
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > 
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > 
> > > Bug: chromium:1029737
> > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30078}
> > 
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > 
> > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30079}
> 
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: chromium:1029737
> Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30097}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

Change-Id: I73d4fe3bb18e40a01f1b1b0c71f9dc7b85c513b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162208
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30100}
 2019-12-16 15:28:41 +00:00
Johannes Kron
77eb338ae4 Reland "Distinguish between send and receive video codecs" 
This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.

Reason for revert: Downstream test updated.

Original change's description:
> Revert "Reland "Distinguish between send and receive video codecs""
> 
> This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> 
> Reason for revert: Breaks another downstream test.
> 
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> > 
> > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > 
> > Reason for revert: Downstream tests have been updated.
> > 
> > Original change's description:
> > > Revert "Distinguish between send and receive video codecs"
> > > 
> > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > 
> > > Reason for revert: Breaks downstream test.
> > > 
> > > Original change's description:
> > > > Distinguish between send and receive video codecs
> > > > 
> > > > Even though send and receive codecs are the same,
> > > > they might have different support in HW.
> > > > Distinguish between send and receive codecs to be able to keep
> > > > track of which codecs have HW support.
> > > > 
> > > > Bug: chromium:1029737
> > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > 
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > 
> > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30042}
> > 
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > 
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> > 
> > Bug: chromium:1029737
> > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30078}
> 
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> 
> Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30079}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

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

Bug: chromium:1029737
Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30097}
 2019-12-16 14:03:46 +00:00
Johannes Kron
f2d6fe62f2 Revert "Reland "Distinguish between send and receive video codecs"" 
This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.

Reason for revert: Breaks another downstream test.

Original change's description:
> Reland "Distinguish between send and receive video codecs"
> 
> This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> 
> Reason for revert: Downstream tests have been updated.
> 
> Original change's description:
> > Revert "Distinguish between send and receive video codecs"
> > 
> > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > 
> > Reason for revert: Breaks downstream test.
> > 
> > Original change's description:
> > > Distinguish between send and receive video codecs
> > > 
> > > Even though send and receive codecs are the same,
> > > they might have different support in HW.
> > > Distinguish between send and receive codecs to be able to keep
> > > track of which codecs have HW support.
> > > 
> > > Bug: chromium:1029737
> > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30041}
> > 
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > 
> > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30042}
> 
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: chromium:1029737
> Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30078}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30079}
 2019-12-12 22:30:25 +00:00
Johannes Kron
26e6afe93f Reland "Distinguish between send and receive video codecs" 
This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.

Reason for revert: Downstream tests have been updated.

Original change's description:
> Revert "Distinguish between send and receive video codecs"
> 
> This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> 
> Reason for revert: Breaks downstream test.
> 
> Original change's description:
> > Distinguish between send and receive video codecs
> > 
> > Even though send and receive codecs are the same,
> > they might have different support in HW.
> > Distinguish between send and receive codecs to be able to keep
> > track of which codecs have HW support.
> > 
> > Bug: chromium:1029737
> > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30041}
> 
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> 
> Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30042}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

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

Bug: chromium:1029737
Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30078}
 2019-12-12 22:13:02 +00:00
philipel
dcb4fcc361 Execute cached video encoder switching request if encoder switching is allowed after the switch request was made. 
Bug: webrtc:10795
Change-Id: Ib045794bf7ecec67812e1fad2ec8db987f6011df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161943
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30067}
 2019-12-11 17:16:04 +00:00
Johannes Kron
f22af3cca7 Revert "Distinguish between send and receive video codecs" 
This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.

Reason for revert: Breaks downstream test.

Original change's description:
> Distinguish between send and receive video codecs
> 
> Even though send and receive codecs are the same,
> they might have different support in HW.
> Distinguish between send and receive codecs to be able to keep
> track of which codecs have HW support.
> 
> Bug: chromium:1029737
> Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30041}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30042}
 2019-12-09 14:48:55 +00:00
Johannes Kron
18314bd8d2 Distinguish between send and receive video codecs 
Even though send and receive codecs are the same,
they might have different support in HW.
Distinguish between send and receive codecs to be able to keep
track of which codecs have HW support.

Bug: chromium:1029737
Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30041}
 2019-12-09 13:56:55 +00:00
Florent Castelli
907dc806c7 Reland "Add support for RtpEncodingParameters::max_framerate" 
Perf test failure was fixed separately.

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

Original change's description:
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
>
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

Bug: webrtc:11117
Change-Id: I9c1daf7887c2024c6669dc79bff89d737417458c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161445
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30030}
 2019-12-06 15:11:54 +00:00
Saurav Das
749f6604a1 Enable SSRC 0 in MediaChannel methods 
Refactor voice engine and video engine to use default methods instead of
treating 0 as a special value.

Bug: webrtc:8694
Change-Id: I47c211c6e870cdec737d6b0d05df29a9b534a011
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158600
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30010}
 2019-12-04 23:49:04 +00:00
Markus Handell
32565f684b WebRtcVideoEngine: Enable encoded frame sink. 
This change ultimately enables wiring up VideoRtpReceiver::OnGenerateKeyFrame
and OnEncodedSinkEnabled into internal::VideoReceiveStream so that encoded
frames can flow to sinks installed in VideoTrackSourceInterface.

Bug: chromium:1013590
Change-Id: I136132c210e5811547f2522ddc371d0acac90664
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161093
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30001}
 2019-12-04 11:15:51 +00:00
Saurav Das
934afc6ba1 Deprecate RtpReceiver's SetParameters method 
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.


Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
 2019-12-03 19:50:42 +00:00
Florent Castelli
5cef9c3581 Revert "Add support for RtpEncodingParameters::max_framerate" 
This reverts commit 15be5282e91ba38894e6ad51fe9a35a38a6b7f29.

Reason for revert: crbug.com/1028937

Original change's description:
> Add support for RtpEncodingParameters::max_framerate
> 
> This adds the framework support for the max_framerate parameter.
> It doesn't implement it in any encoder yet.
> 
> Bug: webrtc:11117
> Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29907}

TBR=steveanton@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,orphis@webrtc.org

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

Bug: webrtc:11117
Change-Id: Ic44dd36bea66561f0c46e73db89d451cb3e22773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160941
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29935}
 2019-11-27 14:01:53 +00:00
Bjorn A Mellem
7a9a092708 Delete media transport integration. 
MediaTransport is deprecated and the code is unused.

No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
 2019-11-26 19:19:36 +00:00
Florent Castelli
15be5282e9 Add support for RtpEncodingParameters::max_framerate 
This adds the framework support for the max_framerate parameter.
It doesn't implement it in any encoder yet.

Bug: webrtc:11117
Change-Id: I329624cc0205c828498d3623a2e13dd3f97e1629
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160184
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29907}
 2019-11-25 16:43:59 +00:00
Johannes Kron
00376e190a Add totalInterFrameDelay to RTCInboundRTPStreamStats 
Bug: webrtc:11108
Change-Id: I0e0168ba303b127a8db3946d5fa5f97a1c90fb27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160042
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29894}
 2019-11-25 10:50:37 +00:00
philipel
16cec3be2c Added allow_codec_switching parameter to RTCConfig. 
Bug: webrtc:10795
Change-Id: I5507f1d801e262223bd18198c685b5fffa644b0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157891
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29612}
 2019-10-25 11:06:31 +00:00
Åsa Persson
fcf79cca7b Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. 
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp

Partial implementation: currently only populated when a/v sync is enabled.

Bug: webrtc:7065
Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29581}
 2019-10-23 07:46:39 +00:00
Niels Möller
9429888602 Delete deprecated bytes_sent/bytes_rcvd stat values 
Bug: webrtc:10525
Change-Id: Id3c863fc064de97f77a2f25ed9589dae34c266bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156941
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29503}
 2019-10-17 06:41:38 +00:00
Florent Castelli
8038541a4f Update the header extensions capabilities with mid, rid and rrid 
Video and audio senders are missing mid, rid and rrid extensions in
their GetCapabilities call.

Bug: chromium:1007894
Change-Id: Ie9edba28ae32fda5e501913cac694f43bfb185ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156560
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29493}
 2019-10-15 14:45:58 +00:00
Niels Möller
ac0a4cbbd8 Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" 
This is a reland of fbde32e596f06893d6dda13eb7d29f4c251cf08b

The chromium problem should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/1862437

Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
>
> Changes the standard GetStats, legacy GetStats unchanged.
>
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29462}

Tbr: kwiberg@webrtc.org
Bug: webrtc:10525
Change-Id: I3b61f9535aa3f1fca2ed84f068233803d4ec9fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157045
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29485}
 2019-10-15 10:43:59 +00:00
Mirko Bonadei
ef0627fb50 Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" 
This reverts commit fbde32e596f06893d6dda13eb7d29f4c251cf08b.

Reason for revert: It seems to break WebRTC FYI tests in Chromium.

https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/4763

Original change's description:
> Fix GetStats bytesSent/Received, wireup headerBytesSent/Received
> 
> Changes the standard GetStats, legacy GetStats unchanged.
> 
> Bug: webrtc:10525
> Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29462}

TBR=kwiberg@webrtc.org,hbos@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: I6a983ea4d5ff38e49f096a8ff5cd9b426768f955
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10525
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157043
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29478}
 2019-10-15 08:55:06 +00:00
Niels Möller
fbde32e596 Fix GetStats bytesSent/Received, wireup headerBytesSent/Received 
Changes the standard GetStats, legacy GetStats unchanged.

Bug: webrtc:10525
Change-Id: Ie10fe8079f1d8b4cc6bbe513f6a2fc91477b5441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156084
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29462}
 2019-10-14 13:07:13 +00:00
Elad Alon
80f53b785b Extend WebRTC-Video-MinVideoBitrate to experiment per-codec 
The experiment was extended to support per-codec minimum bitrates
for the following codecs:
 * VP8
 * VP9
 * H.264

The old semantic meaning for the field trial is retained, in that
specifying "br:" applies a minimum bitrate to all codecs. If "br:"
is not specified, the per-codec minimum config is consulted.

Bug: webrtc:11024
Change-Id: I89630262c7710771d5e25d039fe35f0bd217b58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156171
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29450}
 2019-10-11 15:34:33 +00:00
Saurav Das
ff27da5ca1 Add/remove receive streams with SSRC 0 from media channels 
This enables creation and removal of receive streams with SSRC 0.
Several related methods, for example SetOutputVolume, still use 0 as a
special value.

Bug: webrtc:8694
Change-Id: I341e6bd6c981c9838997510d8d712ad2948f6460
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152780
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29398}
 2019-10-07 23:01:28 +00:00
Sebastian Jansson
1b83a9e400 Only handle each RTCP once. 
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.

The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.

Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
 2019-09-18 16:54:39 +00:00
Mirko Bonadei
53227ccba9 Remove webrtc::MinPositive from api/. 
Follow-up of https://webrtc-review.googlesource.com/c/src/+/153220,
where during code review it was suggested to move webrtc::MinPositive
out of the api/ directory.

Bug: None
Change-Id: I0c3b87a9ffd1cd205a85dddd9f44cfd95eb02206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153480
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29220}
 2019-09-18 12:52:09 +00:00
philipel
d9cc8c08dc Encoder switching based on network and/or resolution conditions. 
In this CL:
 - Renamed EncoderFailureCallback to EncoderSwitchRequestCallback. An encoder
   switch request can now also be made with a configuration that specifies which
   codec/implementation to switch to.
 - Added "WebRTC-NetworkCondition-EncoderSwitch" field trial that specifies
   switching conditions and desired codec to switch to.
 - Added checks to trigger the switch based on these conditions.

Bug: webrtc:10795
Change-Id: I9d3a9a39a7c4827915a40bdceed10b581d70b90a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29196}
 2019-09-16 13:43:29 +00:00
Niels Möller
7bf7a427bf Delete flag VideoReceiveStream::Config::Rtp::remb 
This flag became unused in https://codereview.webrtc.org/2789843002;
it was set, but the setting had no effect.

Bug: webrtc:7135
Change-Id: I012a7c3600bc7a371c7a589695823b30ed5647a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29192}
 2019-09-16 11:20:55 +00:00
Niels Möller
65f17ca6b4 Move MediaTransportInterface out of the libjingle_peerconnection_api target 
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.

Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
 2019-09-13 10:49:56 +00:00
Evan Shrubsole
cc62b16658 Add qualityLimitationResolutionChanges stat 
Implements the stat qualityLimitationResolutionChanges [1].

[1] https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges

Bug: webrtc:10935
Change-Id: I391f2be5958a96b442e32c40ab7043752f3f71dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150882
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#29113}
 2019-09-09 15:22:57 +00:00
Ying Wang
8c5520cfca Reland "Make the min video bitrate in VideoSendStream configurable." 
This reverts commit 1d2149c59c2c1b2834b8cb7983ad56b213129a42.

Reason for revert: The failed test is flaky recently.

Original change's description:
> Revert "Make the min video bitrate in VideoSendStream configurable."
> 
> This reverts commit b2fb0b937ce97b4ccf6363d4f91620a7ab02e87e.
> 
> Reason for revert: breaking downstream projects
> 
> Original change's description:
> > Make the min video bitrate in VideoSendStream configurable.
> > 
> > "WebRTC-VP8-Forced-Fallback-Encoder-v2" affect VP8 only, "WebRTC-Video-MinVideoBitrate" apply to all codec. When both field trial string are set, the bitrate set by "WebRTC-VP8-Forced-Fallback-Encoder-v2" will be used.
> > 
> > Bug: webrtc:10915
> > Change-Id: I63da5909c04ecfad99e93a535fbf71293890fd11
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151135
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29047}
> 
> TBR=ilnik@webrtc.org,asapersson@webrtc.org,crodbro@webrtc.org,yinwa@webrtc.org
> 
> Change-Id: If61c18a36ac2778226da4d2631da1c18e7d4ef81
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10915
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151240
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29049}

TBR=ilnik@webrtc.org,alessiob@webrtc.org,asapersson@webrtc.org,crodbro@webrtc.org,yinwa@webrtc.org

Change-Id: I8df97f7b8ecbea1215eef44d485c179dc4e6246c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151241
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29050}
 2019-09-03 15:25:31 +00:00
Alessio Bazzica
1d2149c59c Revert "Make the min video bitrate in VideoSendStream configurable." 
This reverts commit b2fb0b937ce97b4ccf6363d4f91620a7ab02e87e.

Reason for revert: breaking downstream projects

Original change's description:
> Make the min video bitrate in VideoSendStream configurable.
> 
> "WebRTC-VP8-Forced-Fallback-Encoder-v2" affect VP8 only, "WebRTC-Video-MinVideoBitrate" apply to all codec. When both field trial string are set, the bitrate set by "WebRTC-VP8-Forced-Fallback-Encoder-v2" will be used.
> 
> Bug: webrtc:10915
> Change-Id: I63da5909c04ecfad99e93a535fbf71293890fd11
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151135
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29047}

TBR=ilnik@webrtc.org,asapersson@webrtc.org,crodbro@webrtc.org,yinwa@webrtc.org

Change-Id: If61c18a36ac2778226da4d2631da1c18e7d4ef81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151240
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29049}
 2019-09-03 15:12:31 +00:00
Ying Wang
b2fb0b937c Make the min video bitrate in VideoSendStream configurable. 
"WebRTC-VP8-Forced-Fallback-Encoder-v2" affect VP8 only, "WebRTC-Video-MinVideoBitrate" apply to all codec. When both field trial string are set, the bitrate set by "WebRTC-VP8-Forced-Fallback-Encoder-v2" will be used.

Bug: webrtc:10915
Change-Id: I63da5909c04ecfad99e93a535fbf71293890fd11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151135
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29047}
 2019-09-03 14:35:13 +00:00
Ying Wang
4271afbc30 Fix the bug and reland "Make min video target bitrate configurable." 
This reverts commit 7e896d01623e136313757b6f97d99ea21586f4c4.

Reason for revert: Fixed the bug and submit again.

Original change's description:
> Revert "Make min video target bitrate configurable."
>
> This reverts commit a471e797bc6bb5d26375e4c56ff4aacbab08b8a9.
>
> Reason for revert: This CL adds a new symbol to .data instead of .rodata and the symbol should be a constant.
>
> Original change's description:
> > Make min video target bitrate configurable.
> >
> > Change-Id: I5adf1e675be2114b648878078a8f2e6808c390c7
> > Bug: webrtc:10915
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150331
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#28959}
>
> TBR=nisse@webrtc.org,sprang@webrtc.org,crodbro@webrtc.org,yinwa@webrtc.org
>
> Change-Id: I90f23c2c849a6ec518710bbcbdd8e9eb249e9de8
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10915
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150534
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28967}

TBR=mbonadei@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,crodbro@webrtc.org,yinwa@webrtc.org

Change-Id: Ieef4972502e3c1e5a6e80d8909718dd312486a8e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150537
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28971}
 2019-08-27 11:12:12 +00:00
Johannes Kron
0c141c591a Fix frames dropped statistics 
The |frames_dropped| statistics contain not only frames that are dropped
but also frames that are in internal queues. This CL changes that so
that |frames_dropped| only contains frames that are dropped.

Bug: chromium:990317
Change-Id: If222568501b277a75bc514661c4f8f861b56aaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150111
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28968}
 2019-08-27 07:43:01 +00:00
Mirko Bonadei
7e896d0162 Revert "Make min video target bitrate configurable." 
This reverts commit a471e797bc6bb5d26375e4c56ff4aacbab08b8a9.

Reason for revert: This CL adds a new symbol to .data instead of .rodata and the symbol should be a constant.

Original change's description:
> Make min video target bitrate configurable.
> 
> Change-Id: I5adf1e675be2114b648878078a8f2e6808c390c7
> Bug: webrtc:10915
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150331
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28959}

TBR=nisse@webrtc.org,sprang@webrtc.org,crodbro@webrtc.org,yinwa@webrtc.org

Change-Id: I90f23c2c849a6ec518710bbcbdd8e9eb249e9de8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150534
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28967}
 2019-08-27 07:28:44 +00:00
Ying Wang
a471e797bc Make min video target bitrate configurable. 
Change-Id: I5adf1e675be2114b648878078a8f2e6808c390c7
Bug: webrtc:10915
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150331
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28959}
 2019-08-26 14:21:31 +00:00
Niels Möller
d77cc24f67 New const method StreamStatistician::GetStats 
And a corresponding struct RtpReceiveStats. This is intended
to hold the information exposed via GetStats, which is quite
different from the stats reported to the peer via RTCP.

This is a preparation for moving ReceiveStatistics out of the
individual receive stream objects, and instead have a shared instance
owned by RtpStreamReceiverController or maybe Call.

Bug: webrtc:10679,chromium:677543
Change-Id: Ibb52ee769516ddc51da109b7f2319405693be5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148982
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28943}
 2019-08-23 08:38:59 +00:00
Steve Anton
2d2bbb16e5 Filter out duplicate receive codecs in the media engine 
A malformed session description can assign the same codec to
different payload types which would hit a DCHECK in the
WebRtcVideoEngine. This changes the video engine to just ignore
the duplicate payload type instead of failing.

Bug: chromium:987598
Change-Id: I2034dd11d315ef05448630c860c7ca3f69ef700b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147943
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28796}
 2019-08-07 17:29:12 +00:00
Bjorn A Mellem
da4f09315f Reland "Only include payload in bytes sent/received." 
This is a reland of 74a1b4b1321b426392d4c32e4a02361226ad5358

Original change's description:
> Only include payload in bytes sent/received.
>
> According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
> https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
> statistic should not include headers or padding.
>
> Similarly, according to
> https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
> received are calculated the same way as bytes sent (eg. not including
> padding or headers).
>
> This change stops adding padding and headers to these statistics.
>
> Bug: webrtc:8516,webrtc:10525
> Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28647}

Bug: webrtc:8516, webrtc:10525
Change-Id: Iaa1613e5becdfaa0af0f6b9f00e5b871937a719c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147520
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28731}
 2019-08-01 01:08:24 +00:00
Bjorn Mellem
bcd068d045 Revert "Only include payload in bytes sent/received." 
This reverts commit 74a1b4b1321b426392d4c32e4a02361226ad5358.

Reason for revert: requested by chromium

Original change's description:
> Only include payload in bytes sent/received.
> 
> According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
> https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
> statistic should not include headers or padding.
> 
> Similarly, according to
> https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
> received are calculated the same way as bytes sent (eg. not including
> padding or headers).
> 
> This change stops adding padding and headers to these statistics.
> 
> Bug: webrtc:8516,webrtc:10525
> Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28647}

TBR=steveanton@webrtc.org,ilnik@webrtc.org,hbos@webrtc.org,ossu@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,mellem@webrtc.org

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

Bug: webrtc:8516, webrtc:10525
Change-Id: Ibd31a8264c19f0c6f57d8deb3974593d198046ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147400
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28701}
 2019-07-29 23:39:49 +00:00
Bjorn A Mellem
74a1b4b132 Only include payload in bytes sent/received. 
According to https://www.w3.org/TR/webrtc-stats/#sentrtpstats-dict* and
https://tools.ietf.org/html/rfc3550#section-6.4.1, the bytes sent
statistic should not include headers or padding.

Similarly, according to
https://www.w3.org/TR/webrtc-stats/#inboundrtpstats-dict*, bytes
received are calculated the same way as bytes sent (eg. not including
padding or headers).

This change stops adding padding and headers to these statistics.

Bug: webrtc:8516,webrtc:10525
Change-Id: I891ad5a11a493cc3212afe93e13f62795bf4031f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146180
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28647}
 2019-07-23 13:52:55 +00:00
Jonas Olsson
0182a0300f Reland "Remove the injectable bitrate allocation strategy API." 
This is a reland of 80cb3f6db622442b6360e67851e8903aa0d06d03

Original change's description:
> Remove the injectable bitrate allocation strategy API.
>
> This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
> plus a ton of now-dead code.
>
> Bug: webrtc:10556
> Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28523}

TBR=kwiberg@webrtc.org

Bug: webrtc:10556
Change-Id: Ic17a7a7cc447292306876ee9582ad62fd2499765
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28585}
 2019-07-17 10:20:45 +00:00
Mirko Bonadei
e95b57cdfc Revert "Remove the injectable bitrate allocation strategy API." 
This reverts commit 80cb3f6db622442b6360e67851e8903aa0d06d03.

Reason for revert: Performance regression on downstream project.

Original change's description:
> Remove the injectable bitrate allocation strategy API.
> 
> This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
> plus a ton of now-dead code.
> 
> Bug: webrtc:10556
> Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28523}

TBR=henrika@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,srte@webrtc.org,alexnarest@webrtc.org,jonasolsson@webrtc.org

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

Bug: webrtc:10556
Change-Id: Ife905d661e7b1a227662395c729a9336c62fd2d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145338
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28560}
 2019-07-12 15:27:19 +00:00
philipel
0bb0881892 Add VideoEncoderFactory::GetImplementations function. 
The GetImplementations function is similar to the GetSupportedFormats function, but instead of providing one SdpVideoFormat per codec it provides one per codec implementation. These SdpVideoFormats can then be tagged so that a certain implementation can be instantiated when CreateVideoEncoder is called.

Bug: webrtc:10795
Change-Id: I79f2380aa03d75d5f9f36138625abf3543c2339d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145215
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28553}
 2019-07-12 09:24:47 +00:00
Florent Castelli
66b3860fc9 Remove WebRTC-SimulcastScreenshare and enable it by default 
As per the spec, you should be able to use simulcast with screenshare.
We remove the field trial for it and keep the old behavior only for
screenshare sources with conference flag on.

Bug: webrtc:8785
Change-Id: I1d6d4e18256fb5cfe0195620706de068f25b8d9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144785
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28543}
 2019-07-11 16:47:10 +00:00