This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
They're about to be removed.
BUG=webrtc:8396
Change-Id: Ie9a45f4c0dccb4414d2a2f939aa5f142edc6e4b6
Reviewed-on: https://webrtc-review.googlesource.com/12280
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20328}
Extract and save some simple annotations for the clean speech input.
The annotations are estimated level, VAD (assuming clean speech) and speech level.
TBR=
Bug: webrtc:7494
Change-Id: Id73358e228fac721a77fc8a61a3474a5d52bdc84
Reviewed-on: https://webrtc-review.googlesource.com/12321
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20327}
We are using <math.h>, not <cmath>. While the latter defines additional
overloads for abs(), including abs(float), they are not guaranteed to be
available in <math.h>.
libc++ ships its own math.h with the additional overloads, and libstdc++ (v6
or later) has a math.h that includes <cmath>, but this is not always
expected to work: for example, GCC 5.x's libstdc++ does not have these
additional overloads and causes the build to fail.
Just use fabsf() from the C standard library directly, as it achieves the
same thing in a more portable fashion.
Bug: None
Change-Id: I805728269b35051edb54126e204eccd2706e3a92
Reviewed-on: https://webrtc-review.googlesource.com/11460
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#20325}
PacedSender::probing_send_failure_ and PacedSender::packet_counter_ should probably also be protected by the critical section.
(This isn't the cause of webrtc:8331.)
TBR=stefan@webrtc.org
Bug: webrtc:8331
Change-Id: I94ebe77341137aa511c736d18a63e3e8ec0d1bac
Reviewed-on: https://webrtc-review.googlesource.com/12220
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20324}
YuvConverter was left package protected by mistake in the previous
change.
Bug: None
Change-Id: I2235f745c2f36f9b49199a3dad09da78f63c33e3
Reviewed-on: https://webrtc-review.googlesource.com/11980
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20322}
This CL changes the filter delay detection to rely on the largest peak
while the correctness of the filter is changed to be based on the
performance achieved by the filter.
Bug: webrtc:8397,chromium:774867
Change-Id: I70c953815192478f9a8e0da9f2b8fd9edac3f481
Reviewed-on: https://webrtc-review.googlesource.com/10803
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20321}
This test was disabled for being flaky on the bots.
Try reenabling it to see if something has changed.
Bug: webrtc:7247
Change-Id: I65ce2cf6ce7a3761247369255d9ba106aa3e53f9
Reviewed-on: https://webrtc-review.googlesource.com/3262
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20320}
This CL changes the AEC3 behavior to be more transparent when there
is uncertainty about the amount of echo in the microphone signal.
Bug: webrtc:8398, chromium:774868
Change-Id: I88e681f8decd892f44397b753df371a1c4b90af0
Reviewed-on: https://webrtc-review.googlesource.com/10801
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20319}
By having a unique_ptr own the callback data instead of a raw pointer,
the compiler helps us ensure that it's destroyed exactly once,
and never used after being destroyed.
(This made the callback object move-only, so I had to add support
for move-only callbacks to rtc::Thread::Invoke().)
BUG=webrtc:8111
Change-Id: Ia0804e4662e63e91e5cee18ecc3f38d2cfe8a26b
Reviewed-on: https://webrtc-review.googlesource.com/10812
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20317}
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 af721b72cc1bdc5d945629ad78fbea701b6f82b9
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 bc675ff3fa71549b0a0fdeca56803b41f4de0f53
Original change's description:
> Reland "Use injectable hardware video decoder/encoder in AppRTCMobile."
>
> This is a reland of 0cbaf1a6f6ad13a25993f6ea3be931894a196834
> 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 a8264dbdd97f5e125d45fd0e84356f2e1f747df1.
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}
