Hard-coding default values forces IDs over 14 to be used even
when we offer less than 15 different extensions.
Note that the code relies on MergeRtpHdrExts for making sure
that extension IDs are kept consistent and non-colliding between
different streams (audio/video).
Bug: webrtc:10288
Change-Id: I3e59f7ddc8ca43cea91084a6b7f36df70fb6be4a
Reviewed-on: https://webrtc-review.googlesource.com/c/121646
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26622}
With this CL, we normalize the resolution coming from the
capturer, before applying the requested scaling factors.
That has the benefit that the actual scale factor between
two layers will be the fraction of the requested scale
factors of the two layers.
Prior to this CL, when the normalization was done per layer,
the actual scale factor between two layers might not
have been the fraction of the requested scale factors
of the two layers.
Bug: webrtc:10069
Change-Id: I9ca4d394f259d5d37faee96a41204ff8df898907
Reviewed-on: https://webrtc-review.googlesource.com/c/121425
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26550}
This adds the following non-standardized metrics to video receiver
stats:
- freezeCount
- pauseCount
- totalFreezesDuration
- totalPausesDuration
- totalFramesDuration
- sumOfSquaredFrameDurations
For description of these metrics see
https://henbos.github.io/webrtc-provisional-stats/#RTCVideoReceiverStats-dict*
Bug: webrtc:10145
Change-Id: I4c76d5651102e73b1592ffed561e6224f2badeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/114840
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26523}
Reland with fixes. Previous iteration affected media bitrate in bunch of tests.
Always use real VideoStreamsFactory in full stack tests
Because quality scaling is enabled now in full stack test, correct
factory should be used to compute actual resolution.
Also, since analyzed stream may be disabled completely now, change how
analyzer considers the test finished --- count captured frames and
stop if required amount of frames is captured and no new comparison were made.
Original Reviewed-on: https://webrtc-review.googlesource.com/c/118687
Bug: webrtc:10204
Change-Id: Id1d9066add185d56fe3cb6856b700d350576c6b2
Reviewed-on: https://webrtc-review.googlesource.com/c/119950
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26460}
Support varies by codec, especially in the simulcast case, but using
the EncoderSimulcastProxy codec should fix this.
Bug: webrtc:10069
Change-Id: Idb6a5f400ffda1cdb139004f540961a9cf85d224
Reviewed-on: https://webrtc-review.googlesource.com/c/119400
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26449}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
Further, strictly require VideoReceiveStream::Config::rendererer
to be non-null when the VideoReceiveStream is started. This is
already true by construction in the production code.
Bug: None
Change-Id: Ia0a41cfafa44215efc195a9eb6204194930c3dde
Reviewed-on: https://webrtc-review.googlesource.com/c/115040
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26384}
This class collects the field trial based configuration of audio
allocation and bandwidth in one place. This makes it easier
overview and prepares for future cleanup of the trials.
Bug: webrtc:9718
Change-Id: I34a441c0165b423f1e2ee63894337484684146ac
Reviewed-on: https://webrtc-review.googlesource.com/c/118282
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26370}
Previously, when abs-send-time was negotiated, it was not sent if TWCC
was enabled. With this FieldTrial, abs-send-time header extension is
sent even if TWCC was negotiated in addition to abs-send-time.
Bug: webrtc:10234
Change-Id: I3af85720760882e89760888d43996fe85def619a
Reviewed-on: https://webrtc-review.googlesource.com/c/118936
Commit-Queue: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26368}
The lowest and highest resolution layers are also identified instead
of assuming they are the first and last ones.
Bug: webrtc:10069
Change-Id: If9c76d647415c5065b79dc71850709db6bf16f61
Reviewed-on: https://webrtc-review.googlesource.com/c/114429
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26343}
This CL add new data to the VideoEncoder::EncoderInfo struct, indicating
how the encoder intends to allocate frames across spatial and temporal
layers.
This metadata will be used in upcoming CLs to control how the encoder's
rate controller performs.
Bug: webrtc:10155
Change-Id: Id56fae04bae5f230d1a985171097d7ca83a3be8a
Reviewed-on: https://webrtc-review.googlesource.com/c/117900
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26300}
kMaxSimulcastStreams, kMaxSpatialLayers and kMaxTemporalStreams don't
really beling on VideoBitrateAllocation.
common_types.h is going away and it feels dubious to requrie include
of the full VideoEncoder api to use them. Therefore moving them into a
seprate file/target.
Also includes some remaining cleanup of includes.
Bug: webrtc:9271
Change-Id: I7ded3d97a9a835ac756159700774445a2b93a697
Reviewed-on: https://webrtc-review.googlesource.com/c/117305
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26299}
It only tests the videoadaptation in the test class FakeVideoCapturer.
Bug: webrtc:6353
Change-Id: I4766eebc5cfa7412fde9fd6e5173f1b2381a5d87
Reviewed-on: https://webrtc-review.googlesource.com/c/118042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26297}
In this reland, I disabled high bitrate webrtc perf test on Android32.
This is a reland of 15df2774f4e85cf8900768c1793edcf17d651dcd
Original change's description:
> This CL adds a fake codec factory in WebRTC that can be used in tests to
> produce target bitrate output.
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org
Bug: chromium:879723
Change-Id: I31a4b48d986bef9ca003ae71afeb567ae3e562c9
Reviewed-on: https://webrtc-review.googlesource.com/c/117980
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26285}
Test wiring to DegradationPreference passed to
VideoSendStream::SetSource, but not the adaptation implemented in the
test class FakeVideoCapturer.
Bug: webrtc:6353
Change-Id: Iec2ae89283fb856822ea2829db17eaa02337b467
Reviewed-on: https://webrtc-review.googlesource.com/c/117641
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26278}
NetEq currently only passes `jitterBufferDelay` to `getStats()`. We need its paired `jitterBufferEmittedCount` denominator stat for the calculations to be accurate.
Bug: webrtc:10192
Change-Id: I655aea629026ce9101409c2e0f18c2fa57a1c3ab
Reviewed-on: https://webrtc-review.googlesource.com/c/117320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#26276}
Currently, CPU overuse settings for HW encoders are sometimes being used
even though the actual encoder is a SW encoder, e.g. in case of SW fallback
when the encoder is initialized. Polling is_hardware_accelerated after the
encoder has been created and initialized will improve choosing the correct
CPU overuse settings.
Bug: webrtc:10065
Change-Id: Ic6bd67630a040b5a121c13fa63dd074006973929
Reviewed-on: https://webrtc-review.googlesource.com/c/116688
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26266}
A capturer with this flag was set in
WebRtcVideoChannelTest.PreviousAdaptationDoesNotApplyToScreenshare.
But the flag is used only by the VideoCapturerTrackSource class, which
isn't used in this test.
Bug: webrtc:6353
Change-Id: I58058c882c5a65b5cfa9921e302c422c8ccb20a9
Reviewed-on: https://webrtc-review.googlesource.com/c/117561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26256}
Removes the deprecated video codec factories and the related flag and
helper classes.
Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
Replaced with a combination of cricket::FakeFrameSource and
webrtc::test::FrameForwarder. This cl converts the first three
affected tests, the rest will follow.
Bug: webrtc:6353
Change-Id: I556f6b58f4ca81234ffae3dc6e1319f9c60a76ae
Reviewed-on: https://webrtc-review.googlesource.com/c/117260
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26239}
Configuration for AudioProcessing::voice_detection() is moving into
AudioProcessing::Config, to get rid of the pointer-to-submodule
interfaces (such as voice_detection()).
Bug: webrtc:9947
Change-Id: Ia64ae996a43d44423aa0d612a3f1185b52a3e534
Reviewed-on: https://webrtc-review.googlesource.com/c/116067
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26216}
This reverts commit 15df2774f4e85cf8900768c1793edcf17d651dcd.
Reason for revert: It's causing the Android perf bots to fail. E.g.: https://ci.chromium.org/buildbot/client.webrtc.perf/Android32%20Tests%20%28L%20Nexus4%29/6666
Original change's description:
> Add a high bitrate full stack test with fake codec
>
> This CL adds a fake codec factory in WebRTC that can be used in tests to
> produce target bitrate output.
>
> We also add a high bitrate test that makes use of fake codec. This test assumes
> ideal network conditions with target bandwidth being available and exercises
> WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
>
> Bug: chromium:879723
> Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
> Reviewed-on: https://webrtc-review.googlesource.com/c/97185
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26182}
TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,emircan@webrtc.org,kron@webrtc.org
Change-Id: I33cd01ce345d81d66543f9be99750fa100760b5c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:879723
Reviewed-on: https://webrtc-review.googlesource.com/c/116785
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26192}
This CL adds a fake codec factory in WebRTC that can be used in tests to
produce target bitrate output.
We also add a high bitrate test that makes use of fake codec. This test assumes
ideal network conditions with target bandwidth being available and exercises
WebRTC calls with a high target bitrate(100 Mbps) end-to-end.
Bug: chromium:879723
Change-Id: I981124e2087054ed72c5447e239f28aae0878e29
Reviewed-on: https://webrtc-review.googlesource.com/c/97185
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26182}
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.
Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
The class has been renamed to EncoderSimulcastProxy.
Bug: webrtc:10069
Change-Id: Ief03cfb27145798ac46692d9e51371d2e119eeb0
Reviewed-on: https://webrtc-review.googlesource.com/c/114551
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26031}
