1222 Commits

Author SHA1 Message Date
Marina Ciocea
412a31bbf8 Insert frame transformer between Depacketizer and Decoder.
Add a new API in RTReceiverInterface, to be called from the browser side
to insert a frame transformer between the Depacketizer and the Decoder.

The frame transformer is passed from RTReceiverInterface through the
library to be eventually set in RtpVideoStreamReceiver, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169130.

This change is part of the implementation of the Insertable Streams Web
API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30654}
2020-03-02 08:33:44 +00:00
Henrik Boström
6038383565 [Overuse] Separate getting adaptation target from applying it.
This CL takes us one step closer to being able to evaluate alternative
possible adaptation targets (e.g. multi-stream adaptation) by exposing
the target separately from applying it.

This is a refactoring of OnResourceUnderuse() and OnResourceOveruse().

Prior to this CL, the target resolution or frame rate was calculated
inside these methods and applied if possible. This CLs makes these two
steps (calculating a usable target + applying it) separate methods.

After this CL, the target is expressed as AdaptationTarget and is
calculated and returned by GetAdaptUpTarget() and GetAdaptDownTarget().
The target is only returned if it can be applied - and CanAdaptUp() +
CanAdaptDown() are merged with these methods.

Applying the target happens at ApplyAdaptationTarget().

Bug: webrtc:11222
Change-Id: I8e488be1d1590c23848db816d49a7738562e176d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169100
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30643}
2020-02-28 09:00:31 +00:00
Marina Ciocea
e77912ba8c Insert frame transformer between Encoded and Packetizer.
Add a new API in RTPSenderInterface, to be called from the browser side
to insert a frame transformer between the Encoded and the Packetizer.

The frame transformer is passed from RTPSenderInterface through the
library to be eventually set in RTPSenderVideo, where the frame
transformation will occur in the follow-up CL
https://webrtc-review.googlesource.com/c/src/+/169128.

Insertable Streams Web API explainer:
https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md

Design doc for WebRTC library changes:
http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk

Bug: webrtc:11380
Change-Id: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-28 07:43:13 +00:00
Åsa Persson
40b764a6ba VideoSendStreamTest: remove unused array and member.
Bug: none
Change-Id: I9049be00ba461e5212406c9a5b51c67ba98240ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168947
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30624}
2020-02-27 08:38:51 +00:00
Rasmus Brandt
9731a14ff8 Improve logging for UpdateActiveSimulcastLayers.
Bug: None
Change-Id: I56d14421044749e9bb89754a72a989820c025600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169220
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30620}
2020-02-26 16:24:46 +00:00
Mirta Dvornicic
4f34d78c85 Report available instead of encoding bitrate to VideoEncoderSelector.
The encoding bitrate might be limited depending on the current encoder.

Bug: webrtc:11341
Change-Id: I734fce12734b1e703e7948847cdb1365c08a137b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169123
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30619}
2020-02-26 15:56:36 +00:00
Henrik Boström
d6fb409d46 [Overuse] Make some should-be-const methods const.
The fact that they weren't const is probably a remenant of the good ol'
days this class being multi-threaded and having to acquire mutexes. Now
they can properly be made const.

In order to make GetConstAdaptCounter() const, this CL makes sure a
cleared adapt_counters_ map contains mappings for all degradation
preferences to default-constructed AdaptCounters. Previously, if the
mapping was missing it was implicitly inserted by
GetConstAdaptCounter(). Now it can DCHECK that mappings always exists
instead, and it always has something to return.

Bug: webrtc:11222
Change-Id: If33227fe6572eb1d6cc6b9f851d6d174d035c110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168953
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30611}
2020-02-25 17:58:21 +00:00
Evan Shrubsole
aa6fbc156e Support injecting new Resources for overuse
* This replaces the video stream methods for forcing adaptation
with a mock resource that triggers overuse.
* Resources can now be injected to the Module using the AddResource
function.
* Resources now have tests for adding and removing callbacks.
* Quality/EncoderUse% resources are tracked in the Resource list of
the adaptation module.
* The adaptation module ties all resources to a reason to keep stats
working as expected.

BUG=webrtc:11377

Change-Id: I1f5902f7416dc41b4915c0072e6f0da2bb3bb2b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168948
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30610}
2020-02-25 16:17:42 +00:00
Ilya Nikolaevskiy
ef0033bca1 Add BW limited vp9 k-svc test
This test would've cought the regression leading to chrome crashes.

Bug: chromium:1051476
Change-Id: I011cb21e333e623412f57f93f0096dbd2dc10505
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168958
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30606}
2020-02-25 14:11:52 +00:00
Henrik Boström
d2a1f09b18 [Overuse] Make Most Adaptation Preconditions Explicit
Today OnResourceOveruse() and OnResourceUnderuse() implicitly checks
preconditions and if they pass calculate the next target, and if those
are usable it applies them to the VideoSourceRestrictions. These are two
big "MaybeAdapt" methods.

This CL takes us one step closer to "GetNextTarget", "CanApplyTarget?"
and "DoApplyTarget!"-logic, which will allow us to more easily evaluate
a multitude of alternative configurations and decide which one to pick
(e.g. multi-stream adaptation).

But it does not take us all the way there. In this CL we have:
- CanAdaptUp, CanAdaptDown: This covers *most* of the preconditions.
- OnResourceUnderuse, OnResourceOveruse: This aborts if CanAdapt returns
  false. If they pass, we calculate the next target and maybe-adapt it.

This is roughly outlined in document still in draft:
https://docs.google.com/document/d/1YMg-AycFZR1DS6hEav9OzJ3hqxFil09qPhlTAgQrU1g/edit?usp=sharing.

A future CL should make the target more explicit and we should know if
the target can be applied before we even try.

Bug: webrtc:11222
Change-Id: If18d9572884aa6ba2350e4670a1516da5835cc98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168721
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30605}
2020-02-25 13:17:11 +00:00
Henrik Boström
02956feb2d [Overuse] Can[Increase/Decrease][Resolution/FrameRate]?
Adapting up or down is currently a "Maybe Adapt" method. In the future
we will want to be able to decide which stream to adapt, and as such we
need to be able to tell if a stream is able to do so.

This takes us one step in that direction, by refactoring
OveruseFrameDetectorResourceAdaptationModule::VideoSourceRestrictor's
methods to follow a simple pattern:

- What is the next step?
  GetHigherFrameRateThan, GetLowerFrameRateThan,
  GetHigherResolutionThan, GetLowerResolutionThan
- Can we adapt?
  CanIncreaseFrameRate, CanDecreaseFrameRate,
  CanIncreaseResolution, CanDecreaseResolution
- Do adapt!
  IncreaseFrameRateTo, DecreaseFrameRateTo,
  IncreaseResolutionTo, DecreaseResolutionTo

Hopefully this makes the code easier to follow.
This CL changes the "Request Higher/Lower" methods to take the target
as input instead of implicitly calculating the target from the current
input resolution or frame rate.

Bug: webrtc:11222
Change-Id: If625834e921a24a872145105f5d553fb8f9f1795
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168966
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30600}
2020-02-25 09:52:13 +00:00
Danil Chapovalov
ce515f7625 Add an integration test frame encryption works with DependencyDescriptor
Bug: webrtc:10342
Change-Id: I3a18c1fbe222eada7a484f8f62a0b5bad76eb073
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168888
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30595}
2020-02-24 16:01:04 +00:00
Patrik Höglund
0e089db913 Roll chromium_revision ce459ab383..6d60176510 (742528:743892)
Manual changes:
  - Changed git repos for libcxx, libcxxabi and libunwind since they
  changed in Chromium.
  - Suppressed failing test on MSAN.

Change log: ce459ab383..6d60176510
Full diff: ce459ab383..6d60176510

Changed dependencies
* src/base: 1d6cd336dc..0794106942
* src/build: 188f078b2d..3e271e1ba5
* src/buildtools: afc5b798c7..feb2d0c562
* src/buildtools/linux64: git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65..git_revision:4166e9fbc1fa5ceab69b69710a0f8b430c50127b
* src/buildtools/mac: git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65..git_revision:4166e9fbc1fa5ceab69b69710a0f8b430c50127b
* src/buildtools/third_party/libc++/trunk: 78d6a7767e..d9040c75cf
* src/buildtools/third_party/libc++abi/trunk: 0d529660e3..196ba1aaa8
* src/buildtools/third_party/libunwind/trunk: 69d9b84cca..d999d54f4b
* src/buildtools/win: git_revision:97cc440d84f050f99ff0161f9414bfa2ffa38f65..git_revision:4166e9fbc1fa5ceab69b69710a0f8b430c50127b
* src/ios: 084a00adec..c5aa761a80
* src/testing: 688f493e49..f07276793c
* src/third_party: c6a4254b5e..f4d9303129
* src/third_party/android_deps/libs/com_google_dagger_dagger: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_dagger_dagger_compiler: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_dagger_dagger_producers: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_dagger_dagger_spi: version:2.17-cr0..version:2.26-cr0
* src/third_party/android_deps/libs/com_google_guava_guava: version:27.0.1-jre-cr0..version:27.1-jre-cr0
* src/third_party/android_deps/libs/com_squareup_javapoet: version:1.11.0-cr0..version:1.11.1-cr0
* src/third_party/android_deps/libs/org_checkerframework_checker_compat_qual: version:2.3.0-cr0..version:2.5.3-cr0
* src/third_party/android_deps/libs/org_jetbrains_kotlin_kotlin_stdlib: version:1.3.41-cr0..version:1.3.50-cr0
* src/third_party/android_deps/libs/org_jetbrains_kotlin_kotlin_stdlib_common: version:1.3.41-cr0..version:1.3.50-cr0
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/7e43e2e8ee..6432bb46ab
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9508452e18..d5a5c48017
* src/third_party/depot_tools: 10e0e6d6c1..1773f37de6
* src/third_party/ffmpeg: bcc5d9fec0..545152f302
* src/third_party/freetype/src: d09e831559..fa147af4a5
* src/third_party/libvpx/source/libvpx: 36133b04c0..55f2e4a0a8
* src/tools: af708e0676..e64334fd9c
Added dependencies
* src/third_party/android_deps/libs/org_jetbrains_kotlinx_kotlinx_metadata_jvm
* src/third_party/android_deps/libs/net_ltgt_gradle_incap_incap
DEPS diff: ce459ab383..6d60176510/DEPS

No update to Clang.

TBR=phoglund@webrtc.org,marpan@webrtc.org, jianj@chromium.org,
BUG=webrtc:11376

Change-Id: I5c45376e397c4ce6f9c151626b2280c750ca420c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168946
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30591}
2020-02-24 13:08:34 +00:00
Mirta Dvornicic
5ed40cfa2e Do not request encoder switch when the video is suspended.
Bug: None
Change-Id: I0ecd4db4ee53e1eb6682a2a98b684fcdf5c2e93b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168924
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30585}
2020-02-21 17:54:52 +00:00
Tim Na
9526c557be Refactoring mock_transport to be used separately
Bug: webrtc:11251
Change-Id: I0a494c34c8d5c458b4d9b1b3616ae360d04df0d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168980
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30584}
2020-02-21 17:02:52 +00:00
Mirko Bonadei
2e161c4dd6 Revert "Remove ResourceAdaptationModule::OnMaybeEncodeFrame"
This reverts commit 93d9ae8a17f2e7b90641cbac28e740afc67d383a.

Reason for revert: Perf regression.

Original change's description:
> Remove ResourceAdaptationModule::OnMaybeEncodeFrame
>
> We can react just as well at OnEncodeVideoFrame, which is the same
> behaviour except after checking if the Encoder is paused and the frame
> dropper.
>
> For the initial frame drop, the frame dropper is irrelevant as the frame
> can not be dropped until we are accepting frames. If we didn't drop the
> frame, the encoder can't be paused as the data rate
> is over 0.
>
> For the quality rampup experiment, similar for encoder paused - we can't
> rampup if we are paused anyways since the data rate needs to be non-zero.
> If we are dropping frames we likely don't want to do quality rampup
> anyways.
>
> Bug: webrtc:11222
> Change-Id: Ie3e09d9d8d509dc17ba7a1443cf4747f61c04f6a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168601
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/master@{#30539}

TBR=ilnik@webrtc.org,hbos@webrtc.org,eshr@google.com

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

No-Try: True
Bug: webrtc:11222
Change-Id: Ifb2fc74eb7572568fb0ee1b53a09e4180f87b30c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168880
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30568}
2020-02-20 11:03:25 +00:00
Danil Chapovalov
e8f4e09be9 Parse DependencyDescriptor rtp header extension
Bug: webrtc:10342
Change-Id: I1b5914232f73803774523fae215cf719c92da305
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168481
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30563}
2020-02-20 09:09:27 +00:00
Mirko Bonadei
4a14f4997c Remove wildcard ownership for build files.
No-Try: True
Bug: webrtc:10381
Change-Id: I852d9a2da7e0c5c12f508a1c788b0b5753503aba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168769
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30558}
2020-02-19 14:05:46 +00:00
Danil Chapovalov
b42c54f949 Refactor parsing generic descriptor extension into own function
Before making it even more complicated that it is right now.

Bug: webrtc:10342
Change-Id: I54f67309b8832cd85b6c5213f9b090908814ebd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168766
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30557}
2020-02-19 13:50:36 +00:00
Danil Chapovalov
cad3e0e2fa Replace DataSize and DataRate factories with newer versions
This is search and replace change:
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g"
git cl format

Bug: webrtc:9709
Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30545}
2020-02-18 16:09:50 +00:00
Markus Handell
c1cbf6be7e Ship GenericDescriptor00 by default.
The change ships GenericDescriptor00 and authentication by default,
but doesn't expose it by default, and makes WebRTC respond to
offers carrying it.

The change adds a unit test for the new semantics.

Tests well in munge-sdp. Frame marking replaced by
http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00
in the offer results in an answer containing the
extension as first entry.

Bug: webrtc:11367
Change-Id: I0ef91b7d4096d949c3d547ece7d6c4d39aa241da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168661
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30542}
2020-02-18 11:11:48 +00:00
Evan Shrubsole
93d9ae8a17 Remove ResourceAdaptationModule::OnMaybeEncodeFrame
We can react just as well at OnEncodeVideoFrame, which is the same
behaviour except after checking if the Encoder is paused and the frame
dropper.

For the initial frame drop, the frame dropper is irrelevant as the frame
can not be dropped until we are accepting frames. If we didn't drop the
frame, the encoder can't be paused as the data rate
is over 0.

For the quality rampup experiment, similar for encoder paused - we can't
rampup if we are paused anyways since the data rate needs to be non-zero.
If we are dropping frames we likely don't want to do quality rampup
anyways.

Bug: webrtc:11222
Change-Id: Ie3e09d9d8d509dc17ba7a1443cf4747f61c04f6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168601
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30539}
2020-02-18 10:50:08 +00:00
Åsa Persson
0e57858fa9 StreamSynchronizationTest: rename and make some variables const.
Bug: none
Change-Id: I5c452b0d2f58b2821db31b19506de2ba73480748
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168125
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30537}
2020-02-18 10:25:47 +00:00
Danil Chapovalov
e209fe6c68 Do not propagate generic descriptor on receiving frame
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.

This relands commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.
with adjustments.

Change-Id: I935977179bef31d8e1023964b967658e9a7db92d
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168489
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30532}
2020-02-17 14:52:03 +00:00
Mirta Dvornicic
6799d732d5 Delete DefaultVideoBitrateAllocator.
It was removed from tests in https://webrtc-review.googlesource.com/c/src/+/123540.

If simulcast is not used, SimulcastRateAllocator returns the
same allocation as DefaultVideoBitrateAllocator.

Bug: webrtc:10164
Change-Id: I3d3e1aefe2fcc2bf853cd63c75e008b86eff9241
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168496
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30509}
2020-02-12 21:29:09 +00:00
Danil Chapovalov
bc1750d52b Revert "Do not propagate generic descriptor on receiving frame"
This reverts commit abf73de8eae90e9ac7e88ce1d52728e8102e824f.

Reason for revert: breaks downstream tests

Original change's description:
> Do not propagate generic descriptor on receiving frame
> 
> It was used only for the frame decryptor.
> Decryptor needs only raw representation that it can recreate
> in a way compatible with the new version of the descriptor.
> 
> Bug: webrtc:10342
> Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30501}

TBR=danilchap@webrtc.org,sprang@webrtc.org,philipel@webrtc.org

Change-Id: I6634df06ee75aa8cdfda614994ab11f7a5845c70
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168488
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30502}
2020-02-11 16:54:07 +00:00
Danil Chapovalov
abf73de8ea Do not propagate generic descriptor on receiving frame
It was used only for the frame decryptor.
Decryptor needs only raw representation that it can recreate
in a way compatible with the new version of the descriptor.

Bug: webrtc:10342
Change-Id: Ie098235ebb87c6f5e2af42d0022d2365cd6bfa29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166163
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30501}
2020-02-11 16:12:16 +00:00
Henrik Boström
8cfecac6e8 [Overuse] Move initial framedrop logic into private inner class.
This is a subset of the module's behavior and accounts for 6 of the
member variables of the OveruseFrameDetectorResourceAdaptationModule.

Isolating this behavior to an inner class makes the module slightly less
convoluted.

Bug: webrtc:11222
Change-Id: Ibb5442afb03a1ee850da590b83cd5afbbb14783d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168309
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30500}
2020-02-11 16:11:11 +00:00
Evan Shrubsole
e67c6bcd06 Remove unused fields and includes from VideoStreamEncoder
Bug: webrtc:11222
Change-Id: Iec496d0955c1a30c61da147f0407fd76534129b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168184
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30496}
2020-02-11 13:58:33 +00:00
Åsa Persson
74d2b1ded5 Add periodic logging of sync delays.
Bug: none
Change-Id: Ib2371651c7a912231c93742410a8aa1b01cc9896
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168344
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30495}
2020-02-11 09:43:49 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 12:21:17 +00:00
philipel
9b05803e19 Implement injectable EncoderSelectorInterface and wire it up in the VideoStreamEncoder.
The EncoderSelectorInterface is meant to replace the "WebRTC-NetworkCondition-EncoderSwitch" field trial, so the field trial will be ignored if an EncoderSelectorInterface object has been injected.

Bug: webrtc:11341
Change-Id: I5371fac9c9ad8e38223a81dd1e7bfefb2bb458cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168193
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30490}
2020-02-10 12:12:47 +00:00
Mirko Bonadei
d4c3c3a454 Move video_replay under rtc_tools/.
As pointed out in [1], RTC public tools should live in rtc_tools.

[1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50

Bug: chromium:942546
Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 17:57:30 +00:00
Ying Wang
9b881abea9 Enable congestion window pushback to reduce bitrate by only drop video frames.
With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.

Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
2020-02-07 14:14:47 +00:00
Max Moroz
f12231d742 Add wildcard visibility to video_replay to make it buildable in Chromium.
Bug: chromium:942546
Change-Id: Ib798b58e854a2471ab1bb94725cb0ee2b04b84da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168320
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Max Moroz <mmoroz@chromium.org>
Cr-Commit-Position: refs/heads/master@{#30477}
2020-02-06 21:41:31 +00:00
Henrik Boström
065348503c [Overuse] Move EncodeUsageResource/QualityScalerResource to own files.
This CL changes EncodeUsageResource and QualityScalerResource from
private inner classes of OveruseFrameDetectorResourceAdaptationModule to
standalone classes, moving them into separate files.

This CL does not intend to change any lines of code, only move them.
Except for removing an unused method quality_scaler().

Bug: webrtc:11222
Change-Id: I86bf7eb78c80031888c403ac43c2bdf9b24eaea6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168198
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30472}
2020-02-06 14:08:39 +00:00
Henrik Boström
48258acabf [Overuse] Implement Resource and ResourceUsageListener.
The Resource interface (previously a skeleton not used outside of
testing) is updated to inform listeners of changes to resource
usage. Debugging methods are removed (Name, UsageUnitsOfMeasurements,
CurrentUsage). The interface is implemented by
OveruseFrameDetectorResourceAdaptationModule's inner classes
EncodeUsageResource and QualityScalerResource.

The new ResourceUsageListener interface is implemented by
OveruseFrameDetectorResourceAdaptationModule. In order to avoid adding
AdaptationObserverInterface::AdaptReason to the ResourceUsageListener
interface, the module figures out if the reason is "kCpu" or "kQuality"
by looking which Resource object triggered
OnResourceUsageStateMeasured(). These resources no longer need an
explicit reference to OveruseFrameDetectorResourceAdaptationModule and
could potentially be used by a different module.

In this CL, AdaptationObserverInterface::AdaptDown()'s return value is
still needed by QualityScaler. This is mirrored in the return value of
ResourceUsageListener::OnResourceUsageStateMeasured(). A TODO is added
to remove it and a comment explains how the current implementation
seems to break the contract of the method (as was the case prior to
this CL).

Follow-up work include:
- Move EncodeUsageResource and QualityScalerResource to separate files.
- Make resources injectable, allowing fake resources in testing and
  removing OnResourceOveruseForTesting() methods.
  (Investigate adding the necessary input signals to the Resource
  interface or relevant sub-interfaces so that the module does not need
  to know which Resource implementation is used.)
- And more! See whiteboard :)

Bug: webrtc:11222
Change-Id: I0a46ace4a2e617874e3ee97e67e3a199fef420a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168180
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30469}
2020-02-06 12:45:14 +00:00
Henrik Boström
7875c99e82 [Overuse] Add EncodeUsageResource and QualityScalerResource.
This refactors the usage of OveruseFrameDetector in
OveruseFrameDetectorResourceAdaptationModule into an inner class of the
module, making the interaction between the detector and the module the
responsibility of this helper class instead.

Similarly, QualityScaler usage is moved into QualityScalerResource.

This takes us one step closer to separate the act of detecting
overuse/underuse of a resource and the logic of what to do when
overuse/underuse happens.

Follow-up CLs should build on this in order to materialize the concept
of having resources, streams and a central decision-maker deciding how
to reconfigure the streams based on resource usage state.

Bug: webrtc:11222
Change-Id: I99a08a42218a871db8f477f31447a6379433ad05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168057
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30468}
2020-02-06 11:29:02 +00:00
Mirko Bonadei
a9e1026304 Make video_replay buildable from Chromium.
Bug: chromium:942546
Change-Id: Ic127e74b75ccb1fa65b317711d20344d0caee5fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168280
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30467}
2020-02-06 10:55:22 +00:00
Evan Shrubsole
e331a122aa Move quality rampup experiment to overuse module
Bug: webrtc:11222
Change-Id: I8d0860bfe8bdfe0a051f5a6165cdcfa0cc25cfb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168181
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30465}
2020-02-06 08:38:39 +00:00
Evan Shrubsole
7c3a1fc082 Move initial quality experiment to adaptation module
Bug: webrtc:11222
Change-Id: Iaa33bd6369a11f91e677b015eb2db412d0fbff23
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168053
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30456}
2020-02-05 10:10:22 +00:00
Evan Shrubsole
c81798b0c4 Configure QP scaler in adaptation module
Bug: webrtc:11222
Change-Id: Ia50ba3d024d0cbbaeddf8bf67ee652be602c5df9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168052
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30453}
2020-02-04 14:46:06 +00:00
Evan Shrubsole
f2be3eff26 Move initial frame drop to overuse module
It would be nice for this to stay in video stream encoder,
but this feature is mostly related to quality scaling. Perhaps
something easier to understand is possible in the future.

Bug: webrtc:11222
Change-Id: I71705f33ff94bbcf2fb9b5c94226c8e76dcba94c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168051
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30446}
2020-02-03 11:56:31 +00:00
Evan Shrubsole
c809e8bd62 Move quality scaling frame drop logic to adaptation module
Bug: webrtc:11222
Change-Id: I43db57fa128924ccaa3e44cd58098e7938e5ff5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168050
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30445}
2020-02-03 10:55:11 +00:00
Evan Shrubsole
bfe3ef8feb Report frame qp to quality scaler via overuse module
Bug: webrtc:11222
Change-Id: I63938adf5f623429eab1bcd668cde8fa5a1a083a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167924
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30435}
2020-01-31 09:18:28 +00:00
Evan Shrubsole
cf0595234c Move quality scaler into adaptation module
This allows for further refactoring, eventually moving
all of quality scaler out of video stream encoder.

Bug: webrtc:11222
Change-Id: Id121608da56f57549a616ccc5f141bb598668b40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167728
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30417}
2020-01-30 09:05:19 +00:00
Evan Shrubsole
73a5e916a9 Remove task_queue dependency for QualityScaler
This allows for the possiblity to move the QualityScaler
out of the VideoStreamEncoder in the future.


Bug: webrtc:11222
Change-Id: I1d563cf08791e27ff5065ce90bcb150a7974d868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167534
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30406}
2020-01-29 12:14:10 +00:00
Danil Chapovalov
97ffbefdab Pass and store PacketBuffer::Packet by unique_ptr
to avoid expensive move of the Packet and prepare PacketBuffer
to return list of packets as a frame.

Bug: None
Change-Id: I19f0452c52238228bbe28284ebb197491eb2bf4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167063
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30404}
2020-01-29 11:48:55 +00:00
Henrik Boström
ad515a255b [Overuse] Move GetCpuOveruseOptions() to adaption module.
This removes the last remaining explicit reference from
OveruseFrameDetectorResourceAdaptationModule to
VideoStreamEncoder.

VideoStreamEncoder's call to SetEncoderSettings() inside
ReconfigureEncoder() is moved a few lines down - it was discovered that
during these lines the EncoderInfo config could get modified in
response to InitEncode() - so this fixes a potential bug.

Bug: webrtc:11222
Change-Id: I9746f28a4df8e631e297669c10636bf17b39acec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167363
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30381}
2020-01-27 13:51:40 +00:00
Danil Chapovalov
159c414ff8 Detach LossNotificationController from RtpGenericFrameDescriptor
To allow to use the LossNotificationController with
an updated version of the frame descriptor extension

Bug: webrtc:10342
Change-Id: I5ac44dc5549dfcfc73bf81ad1e8eab8bd5dd136e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166166
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30369}
2020-01-24 11:53:28 +00:00