Based on limited iPhone7 device swarming pool capacity, we need to
increase tasks expiration time to 1 hour.
Bug: chromium:774381
No-Try: true
Change-Id: I66da6c2d5b03ba8c52eead5260b3c244f1df7894
Reviewed-on: https://webrtc-review.googlesource.com/23003
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20692}
Both of these features are in std::optional and the lack
of them is making Optional use in WebRTC more cumbersome.
We are currently looking at using a more fully-fledged library
for some of our standard utility classes. This is merely a
stop-gap measure.
Bug: None
Change-Id: I958a984fa97a42f6e407be1f38662553efeceac4
Reviewed-on: https://webrtc-review.googlesource.com/22920
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20688}
SetupScreenshareOrSVC() is called only if screen sharing is enabled.
This doesn't allow to setup regular SVC in video_loopback. I removed
that condition.
Bug: none
Change-Id: I6320f9210889288f20f7e8e1deea8ad23d006e0a
Reviewed-on: https://webrtc-review.googlesource.com/21660
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20687}
This will help keep ortc dependencies clean in the future, since
gn --check forces us to depend on components from which we include
headers.
cryptoparams.h moves into api/, but ortc appears to think it
should be there anyway. We could consider moving it into the ortc/ api,
but it doesn't appear to be specific to ortc.
Bug: webrtc:6828
Change-Id: Iddae438d10b5e84b2fbc52565364319e20f90613
Reviewed-on: https://webrtc-review.googlesource.com/22660
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20686}
If Webrtc-ClockEstimation experiment is enabled, median filtering is
applied to results of RtpToNtpEstimator and RemoteNtpEstimator to smooth
out random errors introduced by incorrect RTCP SR reports and networking
delays.
Bug: webrtc:8468
Change-Id: Iec6d094d2809d1efeb0b9483059167d9a03880da
Reviewed-on: https://webrtc-review.googlesource.com/22682
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20682}
This CL also merged native_handle_impl.cc and videoframe_jni.cc to keep
all JNI code for the same Java class in the same file, and also renames
this file to jni/videoframe.cc.
The classes AndroidVideoBufferFactory and JavaVideoFrameFactory are
now unnecessary since we cache everything and can be simplified to
global static functions instead.
Bug: webrtc:8278
Change-Id: I03d7b0bbde64cfb407cd6210478ddf9d5599cd8c
Reviewed-on: https://webrtc-review.googlesource.com/22923
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20677}
https://webrtc-review.googlesource.com/c/src/+/17480 changed the default behavior of HardwareVideoEncoderFactory and HardwareVideoDecoderFactory without providing any way to maintain the previous behavior. This breaks current users.
Making these constructors public at least gives existing users a way to maintain the previous behavior.
Bug: webrtc:7925
Change-Id: Id8f0ec25026592f5e9096ac5f39fdda22993ff09
Reviewed-on: https://webrtc-review.googlesource.com/22763
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20675}
In the SoftwareVideoCodecFactories, don't try to create VP9 encoder or
decoder if WebRTC was built without support for it.
Bug: None
Change-Id: I09b87fdcf798c763310af4998dbea8011843010d
Reviewed-on: https://webrtc-review.googlesource.com/22924
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20672}
VideoEncoderSoftwareFallbackWrapper is updated to take a VideoEncoder as
argument instead relying on built-in SW codecs. The purpose is to make
VideoEncoderSoftwareFallbackWrapper more modular and not depend on
built-in SW encoders.
Bug: webrtc:7925
Change-Id: I99896f0751cfb77e01efd29c97d3bd07bdb2c7c0
Reviewed-on: https://webrtc-review.googlesource.com/22320
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20671}
The listed directories are the same as in PRESUBMIT.py, which
was previously the only way to look this stuff up...
BUG=webrtc:8445
NOTRY=true
Change-Id: I218c91e554f3163ae13fc75b9cbf76b42f70624e
Reviewed-on: https://webrtc-review.googlesource.com/16432
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20670}
In preparation for a new remote clock estimation:
RtpToNtpEstimator will now store unwrapped rtp timestamps in
|measurements| and |calculated| flag is out of |Parameters| struct.
Additional missing |Reset| method added to MovingMedianFilter and
PercentileFilter
Bug: webrtc:8468
Change-Id: I1bb454ba0c93720cbb30bfded19677daaf9f721c
Reviewed-on: https://webrtc-review.googlesource.com/22020
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20669}
This allows Objective C(++) users to supply their own implementations of the webrtc::AudioProcessing module.
Bug: webrtc:8163
Change-Id: I2302610c9b45b2f483146d6fe7d82be419b33ee2
Reviewed-on: https://webrtc-review.googlesource.com/18180
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20668}
Also, copy ios_arm64 to ios_arm64_dbg, so we can rename the bot.
Bug: chromium:774381
No-Try: true
Change-Id: I38a447e229ded6874a661bb6f39ae8ebfdcb194d
Reviewed-on: https://webrtc-review.googlesource.com/22580
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20665}
|packet_overhead| field is added to rtc::NetworkRoute structure.
In PackTransportInternal:
1. network_route() is added which returns the current network route.
2. debug_name() is removed.
3. transport_name() is moved from DtlsTransportInternal and
IceTransportInternal to PacketTransportInternal.
When the selected candidate pair is changed, the P2PTransportChannel
will fire the SignalNetworkRouteChanged instead of
SignalSelectedCandidatePairChanged to upper layers.
The Rtp/SrtpTransport takes the responsibility of calculating the
transport overhead from the BaseChannel so that the BaseChannel
doesn't need to depend on P2P layer transports.
TBR=pthatcher@webrtc.org
Bug: webrtc:7013
Change-Id: If9928b25a7259544c2d9c42048b53ab24292fc67
Reviewed-on: https://webrtc-review.googlesource.com/22767
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20664}
This fixes clients with custom JNI_OnLoad code. This is basically a
reland of https://webrtc-review.googlesource.com/21281, but called from
LoadGlobalClassReferenceHolder instead of InitGlobalJniVariables. The
problem it solves is that some clients call InitGlobalJniVariables
from a strange thread and without using the rest of our Java API.
TBR=lliuu@@webrtc.org,sakal@webrtc.org
Bug: webrtc:8278
Change-Id: I664e50920bcd7e087ec2d8bb5be13569e8ff3088
Reviewed-on: https://webrtc-review.googlesource.com/22801
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20663}
This reverts commit 71677452f9cf210aa98162c6f4bd8d339e625337.
Reason for revert: Broke Chromium.
Original change's description:
> Replaced the SignalSelectedCandidatePairChanged with a new signal.
>
> |transport overhead| field is added to rtc::NetworkRoute structure.
>
> In PackTransportInternal:
> 1. network_route() is added which returns the current network route.
> 2. debug_name() is removed.
> 3. transport_name() is moved from DtlsTransportInternal and
> IceTransportInternal to PacketTransportInternal.
>
> When the selected candidate pair is changed, the P2PTransportChannel
> will fire the SignalNetworkRouteChanged instead of
> SignalSelectedCandidatePairChanged to upper layers.
>
> The Rtp/SrtpTransport takes the responsibility of calculating the
> transport overhead from the BaseChannel so that the BaseChannel
> doesn't need to depend on P2P layer transports.
>
> Bug: webrtc:7013
> Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c
> Reviewed-on: https://webrtc-review.googlesource.com/13520
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20661}
TBR=steveanton@webrtc.org,zhihuang@webrtc.org,pthatcher@webrtc.org
Change-Id: Ie0c76786855b65bb8caba7065593c961e4bf9de7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7013
Reviewed-on: https://webrtc-review.googlesource.com/22764
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20662}
|transport overhead| field is added to rtc::NetworkRoute structure.
In PackTransportInternal:
1. network_route() is added which returns the current network route.
2. debug_name() is removed.
3. transport_name() is moved from DtlsTransportInternal and
IceTransportInternal to PacketTransportInternal.
When the selected candidate pair is changed, the P2PTransportChannel
will fire the SignalNetworkRouteChanged instead of
SignalSelectedCandidatePairChanged to upper layers.
The Rtp/SrtpTransport takes the responsibility of calculating the
transport overhead from the BaseChannel so that the BaseChannel
doesn't need to depend on P2P layer transports.
Bug: webrtc:7013
Change-Id: I60d30d785666a50a95052d00bf08f829d8f57e9c
Reviewed-on: https://webrtc-review.googlesource.com/13520
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20661}
This reverts commit b3a8c980f7b33e7457f9a1ee3582ccbc28b7f70c.
Reason for revert: Breaking internal projects
Original change's description:
> Move InitClassLoader from JNI_OnLoad to InitGlobalJniVariables.
>
> This fixes clients with custom JNI_OnLoad code.
>
> Bug: webrtc:8278
> Change-Id: I67b45c2ca07b13b0bf968c5ac17ec6f1bcc58a86
> Reviewed-on: https://webrtc-review.googlesource.com/21281
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20629}
TBR=magjed@webrtc.org,sakal@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8278
Change-Id: I6add8182172501cb93bed998f68ce848c7280a98
Reviewed-on: https://webrtc-review.googlesource.com/22780
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20660}
Introduces the public API interface corresponding to the
standardized RtpTransceiver object in the WebRTC spec.
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
The RtpTransceiver will be the internal representation for both
Plan B and Unified Plan SDP, but the public API interface will
only support Unified Plan (existing users should continue to use
GetSenders/GetReceivers, which will still be supported).
Bug: webrtc:7600
Change-Id: I417ffda683209ba9a9b4cbd274f91ca8295779a7
Reviewed-on: https://webrtc-review.googlesource.com/21460
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20659}
Conditional visibility is complex to maintain and it is not well
supported by other build systems.
This CL removes it and falls back on the more relaxed visibility value
("*" in this case).
It is not a problem because the targets that are using conditional
visibility are all marked as "testonly" and this is probably enough to
keep the build graph clean.
Bug: None
Change-Id: I2d2b5ac9a02d08c2863950116db455976ee1459c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14902
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20658}
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.
Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
video_frame_api, since it no longer contains i420 code
Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
In Set[Local/Remote]Description we have a raw pointer |desc| whose
ownership is passed to a helper function. Before this CL we continue
to use |desc| after ownership is passed under the assumption that the
object is not deleted.
In this CL, we instead rely on [local/remote]_description() after the
helper function has been called. In practice, this is a pointer to
the same object, but it removes the assumption about |desc| being
valid after its ownership is passed.
Bug: None
Change-Id: I144a190ea00f303f4713b64c45aa3e811c0f4b2e
Reviewed-on: https://webrtc-review.googlesource.com/21320
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20654}
hbos knows and makes changes to the webrtc-pc spec[1] and works on
making Chromium's RTCPeerConnection spec-compliant. This includes
knowing and interacting with WebRTC-layer PeerConnection/Interface and
sometimes making changes to it.
hbos would like to share the peerconnection* ownership responsibilty as
it is relevant and owning it will speed up some of the process.
[1] https://w3c.github.io/webrtc-pc/
Bug: None
NOTRY: True
Change-Id: I8f419b7fc6c7fcf19951aa3f304769c915300d1b
Reviewed-on: https://webrtc-review.googlesource.com/21327
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20649}
This is similar to https://webrtc-review.googlesource.com/c/src/+/3620
for iOS.
Using the new WebRtcMediaEngineFactory::Create API, the built-in
software video codecs are no longer appended to the injected codecs.
To be able to use the software codecs, they are exposed as Java
classes through SoftwareVideoEncoderFactory etc.
There is also a new DefaultVideoEncoderFactory used by AppRTCMobile.
This factory tries to use hardware implementations where available,
but falls back to using the injected software codecs.
The HardwareVideoEncoderFactory is temporarily also falling back on
the software codecs in its default configuration in order to
maintain backwards compatibility.
Bug: webrtc:7925
Change-Id: I3e8c5ed492ccd160aca968986ad217d7978a951c
Reviewed-on: https://webrtc-review.googlesource.com/17480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20647}
To facilitate this change, I replaced the graph style with one style
config for lines/interpolation and one style config for points.
The output functions were updated to make use of the new styles.
Bug: None
Change-Id: I42404a8ce274d6e433bcdd6aee4b15b640e78b40
Reviewed-on: https://webrtc-review.googlesource.com/22000
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20645}
