Now some metrics would be aggregated over 1s window instead of 10s.
Bug: webrtc:8402
Change-Id: If0b5a70257c767a1741b451585f3da501b903374
Reviewed-on: https://webrtc-review.googlesource.com/11400
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20316}
This fix brings common_video/BUILD.gn on a par with other BUILD.gn files
that use $rtc_libyuv_dir to include libyuv header files when building
WebRTC. Doing so eliminates the failure in the building process when
a non-default $rtc_libyuv_dir is used.
Bug: webrtc:8399
Change-Id: I1bac0285f1869a334d116d0e1371aa10204137e5
Reviewed-on: https://webrtc-review.googlesource.com/11140
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20315}
List codecs from factory in settings, select by changing order in factory.
Bug: webrtc:7925
Change-Id: If3c45e56713104c88705c67560325d002e6d6700
Reviewed-on: https://webrtc-review.googlesource.com/3720
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20312}
Echo Control is enabled in capture_nonlocked_ when injected.
Renamed echo_canceller3_enabled to echo_controller_enabled.
Bug: webrtc:8346
Change-Id: Icf441f07ce64719358841544da7579feeb7cfdbb
Reviewed-on: https://webrtc-review.googlesource.com/10808
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20311}
...and at least one of our compilers (Visual Studio 64-bit) complains
about it.
BUG=none
Change-Id: I271334f4da564690ff2a16a8322e7ed4a00ae173
Reviewed-on: https://webrtc-review.googlesource.com/10809
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20309}
TBR=sprang@webrtc.org
This is a reland of af721b72cc
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
Bug: webrtc:8375
Change-Id: I06ea90ae8646ba11ddd8ddceb82ea82d75ae2109
Reviewed-on: https://webrtc-review.googlesource.com/11320
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20308}
Unit test now checks that ADM:Init() works before any test runs.
It means that all tests will be skipped on bots that lack Pulse
support which is as how it worked before this CL as well. But then,
we detected the lack of support by checking that the audio layer had
changed from Pulse to Alsa.
As a consequence, I also decided to inject fake/mock ADMs in more
unit tests. One was actually already injected for other reasons
(see https://codereview.webrtc.org/2997383002/) but it had accidentally
been "reverted" later in combination with other changes.
To summarize: before this change we had a set of unit tests where real
audio was tested but it was not the intention of the test or required.
In addition, some Linux bots (VM:s) did not support PulseAudio and on
them the tests relied on a fallback mechanism to ALSA to work, i.e.,
we had a rather complex dependency on hardware. This dependency has now
been removed and it should result in more stable tests.
Bug: webrtc:7306, webrtc:7806
Change-Id: Ia0485658c04a4ef3b3f2dc0d557d73738067304b
Reviewed-on: https://webrtc-review.googlesource.com/8640
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20307}
Since WEBRTC_ANDROID is defined by WebRTC while ANDROID is defined by
Chromium we should stop using ANDROID in WebRTC source code.
Bug: webrtc:8400
Change-Id: I1d59caaabd8af2423e86476b72e0e9185e6c7a3a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/10805
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20306}
`forward_variables_from(invoker, "*")` forwards all locally declared
variables, but not global variables. This means that while setting
`visibility` locally in an rtc_* build target works fine, it does not
currently work to set `visibility` globally for all build targets in a
file.
Fix this by manually forwarding `visibility`.
BUG=webrtc:8254, webrtc:8255
Change-Id: I9e1a5f8ac9cb5991fff2af7c094fe677e1483964
Reviewed-on: https://webrtc-review.googlesource.com/10806
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20305}
TBR=magjed@webrtc.org
This is a reland of bc675ff3fa
Original change's description:
> Reland "Use injectable hardware video decoder/encoder in AppRTCMobile."
>
> This is a reland of 0cbaf1a6f6
> Original change's description:
> > Use injectable hardware video decoder/encoder in AppRTCMobile.
> >
> > Also include a small fix for getting the encoder queue.
> >
> > Bug: webrtc:7760
> > Change-Id: I96dc8ffb363b90382276d88148f81d5f89dca5f2
> > Reviewed-on: https://webrtc-review.googlesource.com/2683
> > Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20022}
>
> Bug: webrtc:7760
> Change-Id: Ia82129fde7abb59120ba5bb23938db9eb576ae91
> Reviewed-on: https://webrtc-review.googlesource.com/4701
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20080}
Bug: webrtc:7760
Change-Id: I5f9bcbf0f18ac3d6b2d2d500300218e885e37d71
Reviewed-on: https://webrtc-review.googlesource.com/9383
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20302}
This reverts commit a8264dbdd9.
Reason for revert: Reverting to unblock rolls into Chromium.
See failure here:
https://build.chromium.org/p/tryserver.chromium.linux/builders/linux_chromium_rel_ng/builds/565449
Fails: external/wpt/webrtc/RTCPeerConnection-setRemoteDescription-offer.html
I'm guessing these lines from the output are relevant:
12:15:32.525 11839 [1:19:1015/121532.495175:16438293900:ERROR:webrtcsession.cc(350)] Failed to set remote offer sdp: The order of m-lines in subsequent offer doesn't match order from previous offer/answer.
12:15:32.525 11839 [1:20:1015/121532.497199:16438296127:WARNING:delay_based_bwe.cc(326)] BWE Setting start bitrate to: 300000
12:15:32.525 11839 [1:1:1015/121532.498272:16438296963:ERROR:webrtcsdp.cc(359)] Failed to parse: "Invalid SDP". Reason: Expect line: v=
12:15:32.525 11839 [1:1:1015/121532.498364:16438297040:ERROR:rtc_peer_connection_handler.cc(2183)] Failed to create native session description. Type: offer SDP: Invalid SDP
12:15:32.525 11839 [1:1:1015/121532.498432:16438297104:ERROR:rtc_peer_connection_handler.cc(1458)] Failed to parse SessionDescription. Invalid SDP Expect line: v=
Original change's description:
> Reject the subsequent offer with fewer m= sections.
>
> If the subsequent offer contains fewer m= sections than the existing
> description, it would be rejected.
>
> The helper method MediaSectionsInSameOrder is modified and it will
> compare the number of m= sections before matching the media type.
>
> Bug: chromium:773620
> Change-Id: Ic8999445f4bc023da1d85a65659583db1687ec37
> Reviewed-on: https://webrtc-review.googlesource.com/9621
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20298}
TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:773620
Change-Id: I4a3ff7a42abb95144615b1dd37fb21585ee07b5d
Reviewed-on: https://webrtc-review.googlesource.com/10920
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20300}
Added EchoCanceller3Factory that implements EchoControlFactory and can
be used for injecting EchoCanceller3 into Audio Processing Module.
Renamed InitializeEchoCanceller3 to InitializeEchoController.
Bug: webrtc:8346
Change-Id: I47078da6a49aca1ee41f6a9d5b7b8e91bb5c11a3
Reviewed-on: https://webrtc-review.googlesource.com/9220
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20299}
If the subsequent offer contains fewer m= sections than the existing
description, it would be rejected.
The helper method MediaSectionsInSameOrder is modified and it will
compare the number of m= sections before matching the media type.
Bug: chromium:773620
Change-Id: Ic8999445f4bc023da1d85a65659583db1687ec37
Reviewed-on: https://webrtc-review.googlesource.com/9621
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20298}
Specifically:
ScreenCapturerTest.StartCapture
ScreenCapturerTest.Capture
WindowCapturerTest.Capture
Also adding a DCHECK for the capturer actually being created, since it
seems like that's the problem.
TBR=zijiehe@chromium.org
Bug: webrtc:7830
Change-Id: I200dc0c15f5039b95f591597bc00d3f1084ae876
Reviewed-on: https://webrtc-review.googlesource.com/9562
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20297}
This will allow Chrome to provide a RtcEventLogOutput object that reports the log back to Chrome, allowing Chrome to manage the log by itself - write it to a file, upload it to a server, etc.
Bug: webrtc:8111
Change-Id: I6a2a6945fc8586ef10e0fb9c56eaa8fda00dfc98
Reviewed-on: https://webrtc-review.googlesource.com/8081
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20295}
This will cause the application to be aborted before it encounters
something worse like a heap overflow, in case any bug in this code
exists or is introduced in the future.
TBR=zhihuang@webrtc.org
Bug: chromium:773620
Change-Id: Idd4e31aa63a3f673eefd3e8cb2ae3f4a5092ca4e
Reviewed-on: https://webrtc-review.googlesource.com/9040
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20293}
math.h was being implicitly included, which can break the build with
alternative libc implementations.
Bug: None
Change-Id: I969b320b65d0f44abb33d3e1036cfbcb859a4952
Reviewed-on: https://webrtc-review.googlesource.com/9384
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#20292}
This fixes a crash caused by access to already freed memory returned
by VideoDecoderWrapper::ImplementationName method.
Bug: webrtc:7760
Change-Id: Ia4b020d1dd861e6a45637abde35f12951b7c43ea
Reviewed-on: https://webrtc-review.googlesource.com/9420
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20290}
This CL adds all the scaffolding needed to support xctest target
in sdk/objc/Framework and adds the target to the bot configuration.
The benefits of this are two-fold.
1. We'll separate framework unittests from their current target,
`rtc_unittests`, that has many many other tests.
This way framework unit tests will have nice, compact, selfcontained target.
2. We'll harvest the power of XCTest (native testing framework)
that should hopefully make adding and writing objc tests easier.
This CL migrates only one test to prove the setup works.
More tests will be migrated in follow up cls.
Bug: webrtc:8382
Change-Id: I0b5b9596c2a6d91683d13632323441de1aa461e0
Reviewed-on: https://webrtc-review.googlesource.com/8501
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20289}
Track IDs are assigned by application during track creation.
Track IDs are used by custom bitrate allocation strategies to identify tracks.
Track ID can be empty, in that case bitrate allocation strategies will not be able to handle
these tracks specifically and will handle them as a default.
Bug: webrtc:8243
Change-Id: I89987e33328320bfd0539ad532342df6da144c98
Reviewed-on: https://webrtc-review.googlesource.com/4820
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20285}
The field trial effects two things: after a frame length change the IAT
histogram is scaled to prevent an immediate change in target buffer
level. Also, the peak history in the delay peak detector is cleared,
because the size of the peaks is stored in number of packets (which
will be incorrect after a frame length change).
Bug: webrtc:8381
Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8
Reviewed-on: https://webrtc-review.googlesource.com/8101
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20284}
Nothing is different from the original change.
The original change was reverted because it was thought it might
have caused an outage for the windows bots. (It didn't.)
This is a reland of 28addd03c1
Original change's description:
> Make it possible to isolate bwe_simulations_tests
>
> Add missing data dependencies and add it to gn_isolate_map.pyl
>
> Bug: chromium:749648
> Change-Id: I6b6c1bb2e4d647471a2747042788a691ce2e1e5d
> Reviewed-on: https://webrtc-review.googlesource.com/8721
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20258}
Bug: chromium:749648
No-Try: true
Change-Id: I5531c5f8b58e0d5b8b6ceeea90acb4bd8e6e6e4f
Reviewed-on: https://webrtc-review.googlesource.com/9320
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20280}
In preparation of coming CLs that will add an AGC interface to make the
gain controller injectable.
This CL simplifies AGC2 (dummy sub-module of audioproc_f) since it only
implements the fixed digital mode with hard-clipping - i.e., no limiter
is used.
The AGC2 config now includes the fixed gain to apply and audioproc_f
has been adapted accordingly.
Finally, this CL slightly simplifies the AGC2 integration into APM.
This CL is a continuation of https://codereview.webrtc.org/2995043002/
Bug: webrtc:7494
Change-Id: I3d554ea4dc6208928352059feb14987edabf14c7
Reviewed-on: https://webrtc-review.googlesource.com/4661
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20278}
The config-history is (logically) intended to be unlimited, but in practice, it would be good to cap it, even though the cap is never expected to be reached, so as to prevent a possible attack that would cause memory overuse.
Bug: webrtc:8111
Change-Id: I1f60cf10215bf8191a8ab3c9b19345104c585483
Reviewed-on: https://webrtc-review.googlesource.com/8980
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20276}
I couldn't repro the problem locally, speculatively reland to see if the issue
was solved by recent changes.
TBR=magjed@webrtc.org
This is a reland of bc675ff3fa
Original change's description:
> Reland "Use injectable hardware video decoder/encoder in AppRTCMobile."
>
> This is a reland of 0cbaf1a6f6
> Original change's description:
> > Use injectable hardware video decoder/encoder in AppRTCMobile.
> >
> > Also include a small fix for getting the encoder queue.
> >
> > Bug: webrtc:7760
> > Change-Id: I96dc8ffb363b90382276d88148f81d5f89dca5f2
> > Reviewed-on: https://webrtc-review.googlesource.com/2683
> > Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20022}
>
> Bug: webrtc:7760
> Change-Id: Ia82129fde7abb59120ba5bb23938db9eb576ae91
> Reviewed-on: https://webrtc-review.googlesource.com/4701
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20080}
Bug: webrtc:7760
Change-Id: If2131b4cdac58cb4d26521fa31e657e29085e0b7
Reviewed-on: https://webrtc-review.googlesource.com/6804
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20275}
In order to enable errorprone [1] we have to fix this finding.
third_party/errorprone has been recently enabled in Chromium and in
order to unblock the Chromium Roll we have to fix these errors.
[1] - https://cs.chromium.org/chromium/src/third_party/errorprone/
Bug: webrtc:8390
Change-Id: Ic737def5ae2a8c6ad1216d9b485af59987fe511c
Tbr: magjed@webrtc.org
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/9161
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20273}
The factory for EchoControl is changed from an rtc::Callback1 to an
interface. This avoids using rtc::Callback1 outside of WebRTC.
This also makes the EchoControl factory more similar to other
factories in the code base.
Bug: webrtc:8345
Change-Id: Ie61b9416ed771f8c756326736d17e339eb768469
Reviewed-on: https://webrtc-review.googlesource.com/8900
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20272}
it allows to create less boil-plate code when marshaling call
with move-only parameters (e.g. unique_ptr) to TaskQueue
Bug: None
Change-Id: I97ddf4f8409af2f83d69fd33267e9a87fb60d901
Reviewed-on: https://webrtc-review.googlesource.com/7619
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20271}
This project can be used for testing the AAR before publishing. Removes
dependency to Chromium from the tests to support Android Studio.
Bug: webrtc:8365
Change-Id: I7568a3f636fd7d478d274b4766f33ab00f28a6f0
Reviewed-on: https://webrtc-review.googlesource.com/7608
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20269}
This should be a separate utility class, rather than internal thing in
ReceiveStatisticsProxy.
Bug: webrtc:8347
Change-Id: I9c8238e625999dba5776d6038c819732d07e9656
Reviewed-on: https://webrtc-review.googlesource.com/7609
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20267}
