Depends on Chromium to roll: https://crrev.com/c/924114
This will clobber all Android builds once, since after this, we can't
make Android-specific landmines anymore.
Bug: chromium:756691
Change-Id: Ic7588329e567e3f6e596b04de8f990dc720eb153
Reviewed-on: https://webrtc-review.googlesource.com/54721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Michael Achenbach <machenbach@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22091}
These can't be auto-rolled (yet) but we need to follow
Chromium with this way of specifying dependencies, because
"Android CIPD Ensure" is gone.
Bug: chromium:755920
Change-Id: Iac952db98f0b382b69bc87a109b5c2b284f122ed
Reviewed-on: https://webrtc-review.googlesource.com/54960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22089}
This should be seen as a temporary workaround because we will likely
want to roll these together with Chromium and drop 'Android CIPD Ensure'
like in crrev.com/b59866870a96d6dd39cf573e304ca551848520b9
but it's difficult to update a Python-syntax file like that.
Bug: chromium:755920
No-Try: True
TBR: phoglund@webrtc.org
Change-Id: Ifc508c48ea29ce570cf624d783fa22381ea03fd4
Reviewed-on: https://webrtc-review.googlesource.com/54902
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22088}
After https://webrtc-review.googlesource.com/c/src/+/49060 changed the
gn check config for sdk/.
Add nogncheck for some conditionally imported headers.
Bug: webrtc:7925
Change-Id: I57499e990332636991563c6f550a7c9154e7c2ee
Reviewed-on: https://webrtc-review.googlesource.com/54820
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22083}
Make min_sev_ and dbg_sev_ file-local, and don't inline Loggable().
This should shrink the size of each RTC_LOG statement by a few
instructions. In my local tests the android binary becomes ~12k smaller.
Bug: webrtc:8529
Change-Id: Ic90cf8a7b042d49370cc8d0b1b08058940b615f8
Reviewed-on: https://webrtc-review.googlesource.com/53680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22081}
Deleting the apparently unused include of api/rtp_headers from
common/video/include/video_frame.h broke the PayloadRouter and
VideoSendStream code under video/. Missing declaration of the
RtpPayloadState struct declared in api/rtp_headers.h. Moving the
declaration of that struct to payload_router.h (outside of the api),
since it's used only internally in video/, and that seemed to be a
more logical place for it.
Bug: webrtc:7504
Change-Id: Ibed8233dfeea8bdf144db5422cdf897da824d6ee
Reviewed-on: https://webrtc-review.googlesource.com/53701
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22080}
I hit a problem in a separate CL where targets depended on
rtc_task_queue_for_test were being built while rtc_include_tests
was set to false. So this addresses a future problem.
Bug: webrtc:8848
Change-Id: Id049049d60edd6abdb6d9c56162b7554dc48b057
Reviewed-on: https://webrtc-review.googlesource.com/54880
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22078}
Targets containing files in api/audio are moved from api/BUILD.gn to
api/audio/BUILD.gn.
Bug: webrtc:8844
Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981
Reviewed-on: https://webrtc-review.googlesource.com/54360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22074}
This is a reland of 5af97ee3ad36cb6d386cfefa8c89d7c178015a07.
What's changed from the original?
- Moved the #include for <process.h> for Fuchsia to the types header.
Original change's description:
> Remove criticalsection.cc dependency on platform_thread.cc.
>
> As part of this, I'm moving global thread related functions over to
> platform_thread_types.* and introducing platform_thread_types.cc
> for the implementation.
>
> Change-Id: I4624877fb379e77d215f4ecd60f20b06d8df3ce5
> Bug: webrtc:8893
> Reviewed-on: https://webrtc-review.googlesource.com/53940
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22037}
Bug: webrtc:8893
Change-Id: Idd0baa6756efd10ad11a5c6e4791eaa7a9dbc97f
Tbr: danilchap@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/54800
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22068}
Original change's description:
> Set session error if SetLocal/RemoteDescription ever fails
>
> This changes SetLocalDescription/SetRemoteDescription to set a
> session error which will cause any future calls to fail early if
> there is an error when applying a session description.
>
> This is needed since until better error recovery is implemented
> failing a call to SetLocalDescription or SetRemoteDescription
> could leave the PeerConnection in an inconsistent state.
>
> Bug: chromium:800775
> Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
> Reviewed-on: https://webrtc-review.googlesource.com/54061
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22061}
Bug: chromium:800775
Change-Id: I0016108264e013452e9d34239c012baf23240e99
Reviewed-on: https://webrtc-review.googlesource.com/54720
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22067}
This adds a callback corresponding to the ontrack event as defined
in the WebRTC specification.
Bug: webrtc:7600
Change-Id: Ied8c55e11dcea864428fb194623c1595c21657c7
Reviewed-on: https://webrtc-review.googlesource.com/52660
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22066}
Original change's description:
> Update RTCStatsCollector to work with RtpTransceivers
>
> Bug: webrtc:8764
> Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
> Reviewed-on: https://webrtc-review.googlesource.com/49580
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22026}
Bug: webrtc:8764
Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3
Reviewed-on: https://webrtc-review.googlesource.com/54160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22064}
This reverts commit 71439a60e7915179be96dd42dc732dc51c279884.
Reason for revert: https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Mac%20Tester/47796
Original change's description:
> Set session error if SetLocal/RemoteDescription ever fails
>
> This changes SetLocalDescription/SetRemoteDescription to set a
> session error which will cause any future calls to fail early if
> there is an error when applying a session description.
>
> This is needed since until better error recovery is implemented
> failing a call to SetLocalDescription or SetRemoteDescription
> could leave the PeerConnection in an inconsistent state.
>
> Bug: chromium:800775
> Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
> Reviewed-on: https://webrtc-review.googlesource.com/54061
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22061}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org
Change-Id: I8af271f2b6dd6a896e390a6fe736e809329b4f4a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:800775
Reviewed-on: https://webrtc-review.googlesource.com/54700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22063}
This changes SetLocalDescription/SetRemoteDescription to set a
session error which will cause any future calls to fail early if
there is an error when applying a session description.
This is needed since until better error recovery is implemented
failing a call to SetLocalDescription or SetRemoteDescription
could leave the PeerConnection in an inconsistent state.
Bug: chromium:800775
Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
Reviewed-on: https://webrtc-review.googlesource.com/54061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22061}
Due to flakiness. See bug.
TBR=stefan@webrtc.org
Bug: webrtc:8878
Change-Id: Iece487ad9883fbe6209d25291fb70bb52c8afbd3
Reviewed-on: https://webrtc-review.googlesource.com/54601
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22059}
Apparently left over since 2015 refactorings, see cl
https://codereview.webrtc.org/1417283007
Bug: webrtc:5095
Change-Id: I899f2c018d1906b4336b2e80b511f7398bac4947
Reviewed-on: https://webrtc-review.googlesource.com/53200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22055}
This CL adds the Level Estimator of the new gain controller. The Level
Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub
frames. We take the maximal sample values in every sub frame. We then
apply attack/decay smoothing. This is the final level estimate.
The results will be used with InterpolatedGainCurve (see this CL
https://webrtc-review.googlesource.com/c/src/+/51920). For every level
estimate value, we look up a gain with
InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to
the signal.
Bug: webrtc:7949
Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1
Reviewed-on: https://webrtc-review.googlesource.com/52381
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22054}
It turns out that some headers were not owned by any targets.
These were:
RTCVideoCodec.h
RTCVideoCodecFactory.h
RTCVideoCodecH264.h
RTCVideoEncoderVP8.h
RTCVideoDecoderVP8.h
RTCVideoEncoderVP9.h
RTCVideoDecoderVP9.h
And some were owned by multiple targets, namely:
RTCPeerConnectionFactory+Native.h
RTCPeerConnectionFactory+Private.h
RTCVideoFrameBuffer.h
These have all been moved to their appropriate homes.
This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files.
Bug: webrtc:8855
Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7
Reviewed-on: https://webrtc-review.googlesource.com/49060
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22052}
Moves AndroidNetworkMonitor out of pc folder. Even clients not using
PeerConnection seem to be using it and it doesn't have any dependencies
to the PeerConnection API.
Bug: webrtc:8769
Change-Id: I2bdeff9f5c9925e13388fbc77aa9b264a7583548
Reviewed-on: https://webrtc-review.googlesource.com/53260
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22051}
This CL creates empty placeholders for EchoCanceller3Factory. This
allows for moving the factory of AEC3 as soon as downstream has been
updated to include echo_canceller3_factory.h.
Bug: webrtc:8844
Change-Id: I77c53d8257291f189c637e1c9ed76c4e74be1858
Reviewed-on: https://webrtc-review.googlesource.com/53862
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22050}
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.
The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().
This CL contains
* build changes to make modules/audio_processing/agc2 an independent
target
* a new MutableFloatAudioFrame which is the audio interface between
AGC2 and APM
* move of the fixed gain application from GainController2 to
FixedGainController.
If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#
Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
The functions replace some existing code and will be used in the
the new AutomaticGainController.
Bug: webrtc:7949
Change-Id: I9a32132d4a4699a507b8548a2eac10972a2f3fd6
Reviewed-on: https://webrtc-review.googlesource.com/53141
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22045}
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}
The functions "memcpy" and "memset" are defined in "string.h" which
was not included. Found this when compiling with g++ 5.4 on Ubuntu
Xenial.
Bug: None
Change-Id: Ife9a9ce2a168ecc24d983afcfc0a39784cbedf9f
Reviewed-on: https://webrtc-review.googlesource.com/54121
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22043}
Bug: webrtc:7600
Change-Id: Ic4e5560fdeb9848c65c59e0f45ca3a2a4a22a2ad
Reviewed-on: https://webrtc-review.googlesource.com/53401
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22042}
