And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
The new xcode-clang in xcode 12 turns on -Wrange-loop-analysis by
default, this CL tries to fix problems caused by this.
Bug: webrtc:12134
Change-Id: I77fd9a28486690c11dceceafc4d72f131b081788
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191762
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32559}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
This is a reland of 37d41eea04
Original change's description:
> Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."
>
> This is a reland of 6bfad33fd8
>
> Original change's description:
> > Remove placeholder Obj-C headers and use angle-bracketed headers.
> >
> > sdk/objc/Framework/Headers are just a placeholder headers
> > for backward compatibility and I don't think it is really need this for now.
> > Instead, we can generate the framework header in
> > ios/mac_framework_bundle_with_umbrella_header.
> > Also clang supports the -Wquoted-include-in-framework-header warning,
> > and in Xcode 12, it's in Xcode's recommended settings. This warnings
> > can be avoided by replacing double-quoted includes with angle-bracketed
> > includes when generate framework headers.
> >
> > No-Presubmit: True
> > Bug: webrtc:9627, webrtc:11984
> > Change-Id: I3f6258dfa77a5acee669614005b2747feee35e39
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185920
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32343}
>
> TBR=tommi@webrtc.org, andersc@webrtc.org
>
> No-Presubmit: True
> Bug: webrtc:9627
> Bug: webrtc:11984
> Change-Id: I8f44232f1a70b8ff2ce6a4b4792f0a18472fcec3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187280
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32429}
TBR=tommi@webrtc.org, andersc@webrtc.org
No-Presubmit: True
Bug: webrtc:9627
Bug: webrtc:11984
Change-Id: Ida92b8864dffaea37d3053d3c00381644988b54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32470}
This is a reland of 239f92ecf7
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32457}
This reverts commit 37d41eea04.
Reason for revert: Sorry for the back and forth here. I need to revert again because I need to fix a downstream project. The re-landing should
be without changes.
Original change's description:
> Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."
>
> This is a reland of 6bfad33fd8
>
> Original change's description:
> > Remove placeholder Obj-C headers and use angle-bracketed headers.
> >
> > sdk/objc/Framework/Headers are just a placeholder headers
> > for backward compatibility and I don't think it is really need this for now.
> > Instead, we can generate the framework header in
> > ios/mac_framework_bundle_with_umbrella_header.
> > Also clang supports the -Wquoted-include-in-framework-header warning,
> > and in Xcode 12, it's in Xcode's recommended settings. This warnings
> > can be avoided by replacing double-quoted includes with angle-bracketed
> > includes when generate framework headers.
> >
> > No-Presubmit: True
> > Bug: webrtc:9627, webrtc:11984
> > Change-Id: I3f6258dfa77a5acee669614005b2747feee35e39
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185920
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32343}
>
> TBR=tommi@webrtc.org, andersc@webrtc.org
>
> No-Presubmit: True
> Bug: webrtc:9627
> Bug: webrtc:11984
> Change-Id: I8f44232f1a70b8ff2ce6a4b4792f0a18472fcec3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187280
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32429}
TBR=mbonadei@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,daniel.l@hpcnt.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9627
Bug: webrtc:11984
Change-Id: I006d50c94ad2e336c8a56d49d5ce1c768685f696
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189542
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32432}
This is a reland of 6bfad33fd8
Original change's description:
> Remove placeholder Obj-C headers and use angle-bracketed headers.
>
> sdk/objc/Framework/Headers are just a placeholder headers
> for backward compatibility and I don't think it is really need this for now.
> Instead, we can generate the framework header in
> ios/mac_framework_bundle_with_umbrella_header.
> Also clang supports the -Wquoted-include-in-framework-header warning,
> and in Xcode 12, it's in Xcode's recommended settings. This warnings
> can be avoided by replacing double-quoted includes with angle-bracketed
> includes when generate framework headers.
>
> No-Presubmit: True
> Bug: webrtc:9627, webrtc:11984
> Change-Id: I3f6258dfa77a5acee669614005b2747feee35e39
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185920
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32343}
TBR=tommi@webrtc.org, andersc@webrtc.org
No-Presubmit: True
Bug: webrtc:9627
Bug: webrtc:11984
Change-Id: I8f44232f1a70b8ff2ce6a4b4792f0a18472fcec3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187280
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32429}
This reverts commit 239f92ecf7.
Reason for revert: Breaks downstream projects.
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
TBR=kthelgason@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com
Change-Id: I055fe001fe2757d79be7c304eccc43a8e3104f69
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3513
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32411}
This carries around unsupported content descriptions
(i.e. things where webrtc does not understand the media type
or protocol) in a special data type so that a rejected content or
mediasection is added to the answer SDP.
BUG=webrtc:3513
Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32410}
(Reland with no changes after the fix to the downstream project)
This can be overriden for kNative frame types to perform scaling efficiently.
Default implementations for existing buffer types require actual
buffer implementation, thus this CL also merges "video_frame"
with "video_frame_I420" build targets.
Originally Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186303
(Landing with TBR as it's unchaged reland of already approved CL)
TBR=nisse@webrtc.org,sakal@webrtc.org
Bug: webrtc:11976, chromium:1132299
Change-Id: Ia23f7d3e474bd9cdc177104cc5c6d772f04b210f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187345
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32362}
This reverts commit 6bfad33fd8.
Reason for revert: Breaks downstream project.
Original change's description:
> Remove placeholder Obj-C headers and use angle-bracketed headers.
>
> sdk/objc/Framework/Headers are just a placeholder headers
> for backward compatibility and I don't think it is really need this for now.
> Instead, we can generate the framework header in
> ios/mac_framework_bundle_with_umbrella_header.
> Also clang supports the -Wquoted-include-in-framework-header warning,
> and in Xcode 12, it's in Xcode's recommended settings. This warnings
> can be avoided by replacing double-quoted includes with angle-bracketed
> includes when generate framework headers.
>
> No-Presubmit: True
> Bug: webrtc:9627, webrtc:11984
> Change-Id: I3f6258dfa77a5acee669614005b2747feee35e39
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185920
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32343}
TBR=mbonadei@webrtc.org,andersc@webrtc.org,tommi@webrtc.org,daniel.l@hpcnt.com
Change-Id: I7a6f72ecb8feebf06ad0fe0ecef071da43b98fca
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9627
Bug: webrtc:11984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32344}
sdk/objc/Framework/Headers are just a placeholder headers
for backward compatibility and I don't think it is really need this for now.
Instead, we can generate the framework header in
ios/mac_framework_bundle_with_umbrella_header.
Also clang supports the -Wquoted-include-in-framework-header warning,
and in Xcode 12, it's in Xcode's recommended settings. This warnings
can be avoided by replacing double-quoted includes with angle-bracketed
includes when generate framework headers.
No-Presubmit: True
Bug: webrtc:9627, webrtc:11984
Change-Id: I3f6258dfa77a5acee669614005b2747feee35e39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185920
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32343}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
This reverts commit 60c25a303f.
Reason for revert: Breaks downstream iOS testers.
From an offline discussion, the GN argument is not needed and only
the 3 xctest needs the behavior of that argument. So I am reverting
this one and preparing 2 CLs to properly fix.
Original change's description:
> Reland "Switch from "rtc_ios_xctest_test" to "rtc_test"."
>
> This is a reland of 7a73c772e2
>
> The change to fix the downstream issue is just the switch from
> "test" to "rtc_test" which is a GN template that expands to
> "test".
>
> Original change's description:
> > Switch from "rtc_ios_xctest_test" to "test".
> >
> > Using the "test" GN template instead of the "ios_xctest_test" one we
> > will get iOS support for isolates via MB and GN for free, making it
> > easier to migrate the iOS recipe and fix bugs.webrtc.org/11604.
> >
> > Bug: webrtc:11881
> > Change-Id: I72b90f8494c473fa567e6296caf7a771e4caba92
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182680
> > Reviewed-by: Dirk Pranke <dpranke@google.com>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32064}
>
> Bug: webrtc:11881
> Change-Id: Ia5338859f4e893b9f19bcca6b26b8cf66d5984e8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183766
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@google.com>
> Cr-Commit-Position: refs/heads/master@{#32075}
TBR=mbonadei@webrtc.org,dpranke@google.com,jeffyoon@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11881, webrtc:11937
Change-Id: Ie6eea6b2a8ba5c46af40b115c6db4fd0a38a25b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32112}
and add a unit test
BUG=webrtc:11796
Change-Id: I8e73b22f007c15c862faad7ca881d93c14a3a46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184160
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32104}
Since https://webrtc-review.googlesource.com/c/src/+/161447,
frame_extra_info_ is modified from both the encoder thread and the
callback thread.
Bug: None
Change-Id: Idece4d19cae6d8363428234721616f6ca6f85832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184121
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32095}
All it provides is a method to call a signal on the network thread,
so it's not worth the added complexity. Implementations of
NetworkMonitorInterface must hop to the network thread anyway to
guard their members.
Also added some thread annotations to AndroidNetworkMonitor.
Bug: webrtc:9883
Change-Id: I64bb82ea593433f3a52871dbb75eb2ac4f47d69c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181420
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32087}
This is a reland of 7a73c772e2
The change to fix the downstream issue is just the switch from
"test" to "rtc_test" which is a GN template that expands to
"test".
Original change's description:
> Switch from "rtc_ios_xctest_test" to "test".
>
> Using the "test" GN template instead of the "ios_xctest_test" one we
> will get iOS support for isolates via MB and GN for free, making it
> easier to migrate the iOS recipe and fix bugs.webrtc.org/11604.
>
> Bug: webrtc:11881
> Change-Id: I72b90f8494c473fa567e6296caf7a771e4caba92
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182680
> Reviewed-by: Dirk Pranke <dpranke@google.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32064}
Bug: webrtc:11881
Change-Id: Ia5338859f4e893b9f19bcca6b26b8cf66d5984e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183766
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Dirk Pranke <dpranke@google.com>
Cr-Commit-Position: refs/heads/master@{#32075}
This reverts commit 7a73c772e2.
Reason for revert: Breaks downstream test.
Original change's description:
> Switch from "rtc_ios_xctest_test" to "test".
>
> Using the "test" GN template instead of the "ios_xctest_test" one we
> will get iOS support for isolates via MB and GN for free, making it
> easier to migrate the iOS recipe and fix bugs.webrtc.org/11604.
>
> Bug: webrtc:11881
> Change-Id: I72b90f8494c473fa567e6296caf7a771e4caba92
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182680
> Reviewed-by: Dirk Pranke <dpranke@google.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32064}
TBR=mbonadei@webrtc.org,dpranke@google.com,jeffyoon@google.com
Change-Id: Ia4d6257fee42661c10303217980bd0a9126d2709
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11881
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183765
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32068}
Using the "test" GN template instead of the "ios_xctest_test" one we
will get iOS support for isolates via MB and GN for free, making it
easier to migrate the iOS recipe and fix bugs.webrtc.org/11604.
Bug: webrtc:11881
Change-Id: I72b90f8494c473fa567e6296caf7a771e4caba92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182680
Reviewed-by: Dirk Pranke <dpranke@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32064}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
Some versionss of WebKit.framework export these symbols. Even if they
are private symbols, WebRTC needs to provide a way to prefix them like
the OBJC API symbols (see [1]).
[1] - https://webrtc-review.googlesource.com/c/src/+/173781
Bug: None
Change-Id: Ibb9ca2c89796a0d5e2ca65c549ba8799f24bbe7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182421
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31987}
Notably, this should detect whether an interface is "available" or not,
which should prevent the failure is with dual SIM card setups.
This is gated behind a field trial for now, to ensure this doesn't cause
any regressions due to false negatives (interfaces that are usable
but not listed as available by NWPathMonitor).
Bug: webrtc:10966
Change-Id: Ia3942c4c57b525d08d8b340e2325f3705cfd0304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180923
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31977}
This patch adds a computed estimate on how long the ice stack
was disconnected before switching to a new connection.
The metric is currently computed as now - max(connection->last_data_recevied())
and has resonably good precision.
Bug: webrtc:11862
Change-Id: I8950d55f0eadcf164de089cdb715b4f7eed0a4c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31969}
This clears the decks for deprecating and eventually removing
the nonstandard SetDirection method.
Bug: chromium:980879
Change-Id: Ibc291de3db690e9ef4e6cb3550390d7728f02a83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181860
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31948}
Also moves implementation of legacy setDirection() without error to the
api/ directory.
This is one step in the plan for changing the API
to return RTCError.
Bug: chromium:980879
Change-Id: Ibce8edf8e3c6d41de7ce49d2ffc33f5b282a0e9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31943}
It's now passed through CreatePeerConnectionFactory like all similar
injectable dependencies.
Also removing network_monitor.h from native_api/base, since clients
should all be moved to the new location now.
Bug: webrtc:9883
Change-Id: I3776262ce2a2a380d01c163f4d18a0dfad4b5b41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181401
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31931}
This is a reland of 11dc6571cb
One fix that makes Web Platform Tests pass in debug mode is applied.
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
This reverts commit 11dc6571cb.
Reason for revert: Breaks Chromium WPT tests
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
TBR=sakal@webrtc.org,kthelgason@webrtc.org,hta@webrtc.org,guidou@webrtc.org,marinaciocea@webrtc.org
Change-Id: Ibdc24f7d41e481293ca74ba6d1572de64f7e4654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:980879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181262
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31897}
This adds RtpTransceiver.StopStandard(), which behaves according to
the specification at
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
It modifies RTCPeerConnection.getTransceivers() to return only
transceivers that have not been stopped.
Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
Bug: chromium:980879
Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31893}
This patch adds an interface for os/firmware to set a network
preference NOT_PREFERRED / NEUTRAL that can be picked up by
an IceController and used when selection ice candidate pair.
The patch exposes this using an Android Intent based interface.
BUG: webrtc:11825
Change-Id: Ic12b6bf704fde7f9c912020dd7bc79ccae4613ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180883
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31877}
Since the descriptions can be modified on the signaling thread,
ToString can only be safely called on that thread.
Bug: webrtc:11791
Change-Id: Icf6aada8aa66d00be94c6bda7b22e41b5d3bbc17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180541
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31862}
Not sure why it was lumped in with base in the first place.
But now that it's not set via a static method, it makes less sense.
Bug: webrtc:9883
Change-Id: Ia46834865fa485c9949a01fec10cecba465246ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180741
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31852}
This member of the CodecInfo struct was set in several places, but not
used for anything. To aid deletion, this cl defines a default implementation
of VideoEncoderFactory::QueryVideoEncoder.
The next step is to delete almost all downstream implementations of that method,
since the only classes that have to implement it are the few factories that
produce "internal source" encoders, e.g., for Chromium remoting.
Bug: None
Change-Id: I1f0dbf0d302933004ebdc779460cb2cb3a894e02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179520
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31844}
This is a reland of 003c9be817
Found some downstream code that relies on
NetworkMonitorFactory::SetFactory, so I'm adding those methods back
temporarily. BasicNetworkManager will fall back to the static factory
if the one passed into PeerConnectionFactory is null.
Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}
TBR=hta@webrtc.org, sakal@webrtc.org
Bug: webrtc:9883
Change-Id: I2e817c423f21936f87532a9694eb9a0a1b70c212
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180722
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31824}
This reverts commit 7ded733518.
Reason for revert: Found more code calling NetworkMonitorFactory::SetFactory...
Original change's description:
> Reland "Pass NetworkMonitorFactory through PeerConnectionFactory."
>
> This is a reland of 003c9be817
>
> Original change's description:
> > Pass NetworkMonitorFactory through PeerConnectionFactory.
> >
> > Previously the instance was set through a static method, which was
> > really only done because it was difficult to add new
> > PeerConnectionFactory construction arguments at the time.
> >
> > Now that we have PeerConnectionFactoryDependencies it's easy to clean
> > this up.
> >
> > I'm doing this because I plan to add a NetworkMonitor implementation
> > for iOS, and don't want to inherit this ugliness.
> >
> > Bug: webrtc:9883
> > Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31815}
>
> TBR=hta@webrtc.org, sakal@webrtc.org
>
> Bug: webrtc:9883
> Change-Id: Ibf69a22e8f94226908636c7d50ff9eda65bd4129
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180720
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31822}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,hta@webrtc.org
Change-Id: Iae51b94072cec9abc021eed4e51d1fbeee998adc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180721
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31823}
This is a reland of 003c9be817
Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}
TBR=hta@webrtc.org, sakal@webrtc.org
Bug: webrtc:9883
Change-Id: Ibf69a22e8f94226908636c7d50ff9eda65bd4129
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180720
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31822}
This reverts commit 003c9be817.
Reason for revert: Breaks downstream build which is still using
SetFactory/ReleaseFactory. Probably will need to update this in lockstep.
Original change's description:
> Pass NetworkMonitorFactory through PeerConnectionFactory.
>
> Previously the instance was set through a static method, which was
> really only done because it was difficult to add new
> PeerConnectionFactory construction arguments at the time.
>
> Now that we have PeerConnectionFactoryDependencies it's easy to clean
> this up.
>
> I'm doing this because I plan to add a NetworkMonitor implementation
> for iOS, and don't want to inherit this ugliness.
>
> Bug: webrtc:9883
> Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31815}
TBR=deadbeef@webrtc.org,sakal@webrtc.org,hta@webrtc.org
Change-Id: I1f09df7be9c860017d515e5a87488340afa6eda6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180640
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31818}
Previously the instance was set through a static method, which was
really only done because it was difficult to add new
PeerConnectionFactory construction arguments at the time.
Now that we have PeerConnectionFactoryDependencies it's easy to clean
this up.
I'm doing this because I plan to add a NetworkMonitor implementation
for iOS, and don't want to inherit this ugliness.
Bug: webrtc:9883
Change-Id: Id94dc061ab1c7186b81af8547393a6e336ff04c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31815}
This name change communicates that the recursive critical section
should not be used for new code.
The relevant files are renamed rtc_base/critical_section* ->
rtc_base/deprecated/recursive_critical_section*
Bug: webrtc:11567
Change-Id: I73483a1c5e59c389407a981efbfc2cfe76ccdb43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179483
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31754}
Also, add the prefix of SW Codecs in Codec2.0.
Bug: None
Change-Id: Ifc7a079a68506975cd9e52ddaf6da69744ac0614
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177800
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31723}
Starting from http://crrev.com/c/2289614, transitive dependencies are
not allowed anymore for java targts. This CL prepares WebRTC for the
next Chromium Roll.
Bug: None
Change-Id: I2aafa7be66c215b70d79e0f95272233fe7b37d3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179061
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31695}
GN recently added support for Apple frameworks to link, rather than
overloading the libs lists. This pulls .frameworks out of the libs
lists, so that GN can stop supporting .frameworks in libs in the
future.
Bug: chromium:1052560
Change-Id: I263230ddd3c468061584423bba9e1f887503bcaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178601
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/master@{#31632}
This allows forcing a minimum frame rate if the producer doesn’t update
the SurfaceTexture often.
This needs to be done in SurfaceTextureHelper to keep the
synchronization of the texture access consistent.
Bug: b/149383039
Change-Id: I0e3c82dd51d486b931bd8dda0fd9d5cdb1a90901
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177001
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#31504}
Previously a histogram was added to track the requested buffer size,
this CL adds a histogram for the actually used buffer size.
Bug: b/157429867
Change-Id: I04016760982a4c43b8ba8f0e095fe1171b705258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176227
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31385}
The Android native audio code asks the OS to provide an appropriate
buffer size for real-time audio playout. We should add logging for this
value so we can see what values are used in practice.
Bug: b/157429867
Change-Id: I111a74faefc0e77b5c98921804d6625cba1b84af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176126
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreasson <henrika@chromium.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31368}
This adds priority to the API configuration of datachannels,
and passes the value in the OPEN message.
It does not yet influence SCTP prioritization of messages.
Bug: chromium:1083227
Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31287}
These two types need to be exported in order to access the stats report
from an ObjC / Swift codebase.
Bug: webrtc:11158
Change-Id: Ibb2f81f289b56f824f02df70971c28accd5a1350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31257}
This is a reland of 1b8ef63876. It was
previously reverted (https://webrtc-review.googlesource.com/c/src/+/175008)
but the revert was found to be unnecessary.
Original change's description:
> Add an optional override for AudioRecord device
>
> This is important when we have multiple named devices connected over
> USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
> choose a specific input device to route from.
>
> Bug: b/154440591
> Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
> Commit-Queue: Robin Lee <rgl@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31130}
TBR=henrika@webrtc.org,sakal@webrtc.org,rgl@google.com
Bug: b/154440591, b/155256727
Change-Id: Ic9bf8305c85552a0dc0d2cde6190988423e7fc70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175084
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31255}
This reverts commit 1b8ef63876.
Reason for revert: Breaks downstream projects. b/155256727
Original change's description:
> Add an optional override for AudioRecord device
>
> This is important when we have multiple named devices connected over
> USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
> choose a specific input device to route from.
>
> Bug: b/154440591
> Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
> Commit-Queue: Robin Lee <rgl@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31130}
TBR=henrika@webrtc.org,sakal@webrtc.org,rgl@google.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: b/154440591, b/155256727
Change-Id: I6836676096d47d9da5702a40b9d127569ad50dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175008
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31238}
This field trial will be used to rollout the cellular types added
in https://webrtc-review.googlesource.com/c/src/+/174500 in
a controlled fashion.
Bug: webrtc:11473
Change-Id: I371d13d6935f6e0273a023657ce1b11b32bef346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174831
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31224}
This patch adds detection of 5G to andoird_network_monitor
using the TelephonyManager.NETWORK_TYPE_NR.
It also adds
- TelephonyManager.NETWORK_TYPE_GSM as 2G
- TelephonyManager.NETWORK_TYPE_TD_SCDMA as 3G
- TelephonyManager.NETWORK_TYPE_IWLAN as 4G
note: AdapterTypeFromNetworkType still return rtc::ADAPTER_TYPE_CELLULAR
for all cellular connections (changing that is a next step).
Bug: webrtc:11473
Change-Id: If2e681e10b24f46ea0071db0cdba758a8c4e7ee2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174500
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31171}
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:
- RTC_OBJC_TYPE_PREFIX:
Macro used to prepend a prefix to the API types that are exported with
RTC_OBJC_EXPORT.
Clients can patch the definition of this macro locally and build
WebRTC.framework with their own prefix in case symbol clashing is a
problem.
This macro must only be defined by changing the value in
sdk/objc/base/RTCMacros.h and not on via compiler flag to ensure
it has a unique value.
- RCT_OBJC_TYPE:
Macro used internally to reference API types. Declaring an API type
without using this macro will not include the declared type in the
set of types that will be affected by the configurable
RTC_OBJC_TYPE_PREFIX.
Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10
The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.
Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
Currently, ScopedJavaGlobalRef can only be set at creation and never
changed. This CL makes it possible to re-set these.
Bug: b/153389044
Change-Id: I6be92dae83a9f5f3d87aa44dde226b874f4ca0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31145}
This is important when we have multiple named devices connected over
USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
choose a specific input device to route from.
Bug: b/154440591
Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
Commit-Queue: Robin Lee <rgl@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31130}
We have observed an internal deadlock in libGLESv2_adreno where one
thread is in eglCreateContext and another thread in glUseProgram. We
have observed similar deadlocks before and started to synchronize all
access to the offending GL functions. Calls to eglCreateContext are
already synchronized, and this CL synchronizes calls to glUseProgram as
well.
Bug: b/153513005
Change-Id: I576e564aab44c9e429f2b1407105ed72942c309e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173742
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31118}
We have seen crashes originating from derefencing nullptrs in this code,
for unknown reasons. This CL adds null checks to protect against this.
The stacktraces will be missing or truncated when this happens.
Bug: b/147338449
Change-Id: Ieb006f0f8dec4f9621e4df2e2c1a9641f086df86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173593
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31079}
network_priority was already exposed, but without the ability to set
enable_dscp, it wasn't actually doing anything.
Bug: webrtc:5658
Change-Id: I092bc3dd46e3e7be363313203428bccfccccf3c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171641
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30951}
This reverts commit 86e0ea5711.
Reason for revert: The reasons for removing bitratePriority are unclear. Aside from the fact that you can't yet use it for the relative bitrate of simulcast streams, only the relative bitrate of entire tracks, it's working as intended. It differs from the standard, but only in that it's more flexible; the web standard only allows values of 0.5, 1.0, 2.0, and 4.0 while for the native API we allow any ratio.
Original change's description:
> Remove bitratePriority from the Obj-C RTCRtpEncodingParameters wrapper.
>
> This was added in CL 135122, but the bitratePriority parameter is not
> standard and not implemented in a way users would expect. So it should
> actually not be exposed in the Obj-C SDK.
>
> Bug: webrtc:10438
> Change-Id: I801ce940a32701d2703e951ef2b601c606aa2111
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135287
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27861}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10438
Change-Id: Ibc16b6054a1583de43a868d98683ea114bd89435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171140
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30863}
This is a reland of de86381161
Original change's description:
> Leverage dispatch_queue_create_with_target when possible.
>
> Replacing dispatch_queue_create followed by
> dispatch_set_target_queue with dispatch_queue_create_with_target
> is claimed to be source of GCD performance improvement:
> https://developer.apple.com/videos/play/wwdc2017/706/
> Video since 40 min. Slides since 199.
>
> Bug: webrtc:9055
> Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#30781}
Bug: webrtc:9055
Change-Id: I36b0b6423c81c0497f66f7c993741c33ff6ec5ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170443
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30821}
Removes support for parsing and serializing
a=ssrc:1 mslabel:stream
a=ssrc:1 label:track
which have been superceeded by
a=ssrc:1 msid:stream track
a long time ago.
Bug: webrtc:7110
Change-Id: I3aca47728098b6e7e049b82ed34c59426d411c41
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168244
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30801}
Replacing dispatch_queue_create followed by
dispatch_set_target_queue with dispatch_queue_create_with_target
is claimed to be source of GCD performance improvement:
https://developer.apple.com/videos/play/wwdc2017/706/
Video since 40 min. Slides since 199.
Bug: webrtc:9055
Change-Id: I0136f7faaef0951a7ad243bc8772f3ee952d5470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168491
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/master@{#30781}
to avoid collission and confusion with VideoCodeType based on
c++ enum with the same name.
Bug: b/148146536
Change-Id: I049cce21d59f454c7ce507fdfc3a85d168f96223
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170048
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30728}
The degradation preference is now based on the content hint of the track
if it's unspecified.
Bug: webrtc:11164
Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30691}
It can only be one of four possible values, so it never made sense
for it to be a double. Other than the fact that its neighbor
bitrate_priority is a double, and they're both defined as the same enum
in the web spec. However, while bitrate_priority being a double
offers more flexibility than the web spec, network_priority being a
double is only confusing.
Bug: webrtc:5658
Change-Id: I0784c116f3260c4b3a8b99a3cd85c8d66017e46f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168840
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30685}
This patch exposes webrtc::PeerConnectionDependencies c++-object
and makes it possible to supply one when creating a PeerConnection.
This makes it possible to e.g inject a VideoBitrateAllocatorFactory.
Bug: webrtc:10547
Change-Id: Ib7431bdcec1380e7903dc5f66f3583501aeab0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168307
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30480}
This makes it safe to deliver frames to the sink from VideoProcessor
even after setSink has been called with null reference without danger
of use after free.
Bug: b/148063550
Change-Id: Ib78f75ac49fc6117f744c55da1a4e671bbdcdf22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168160
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30455}
This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This is the first step to move //:android_junit_tests to the righ
package (the target is triggering presubmit errors every time //BUILD.gn
gets updated).
Next steps:
* Update recipes
* Remove //:android_junit_tests
Issues with GN formatting, introduced by [1] will be addressed
separately in a "format all" CL.
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11289
No-Presubmit: True
Change-Id: I70c0927d722911f82dd971c30c7ffb581aed69c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166603
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30328}
This will avoid getting into an inconsistent state where isInterrupted==YES while isActive==YES.
Bug: webrtc:11112
Change-Id: Ia4db85483e1e7a339f520d52a2feb475a73c262e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160140
Commit-Queue: Joe Chen <jsphchn@google.com>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30306}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.
Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I4a1bcd44c9523b6402b3f05b50597bdc2e6615e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30216}
The current camera switch API sequentially cycles through each
camera name for each method invocation. This policy provides
reasonable behavior for devices with 2 or 3 cameras, but
presents challenges with devices that contain several cameras.
For example in a scenario where the current camera is oriented
on the same side as the next camera name, a developer would need to
call switchCamera multiple times to capture from a camera oriented on
a different side of the device.
This commit allows a developer to specify a camera name when switching
cameras. This flexibility allows developers to have more control over
which device they switch to in cases where a device contains several cameras.
Bug: webrtc:11261
Change-Id: I93d46d70b2c7cf735a411a4ef4f33e926bf3a5ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165040
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30199}
This wires the current degradation preference in the SDK, it will later
be nullable in a follow up change once the native API supports it.
Bug: webrtc:11164
Change-Id: I8324e6e0af996dfddfa07e3aff4ba242d9533388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161321
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30170}
Replace all usages of java_files with sources in gn files, and
automatically format.
This is in preparation for java_files being completely removed upstream
in favor of sources.
NOPRESUBMIT=true
Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
1. On iOS13 the implementation of methods begin- and endGeneratingDeviceOrientationNotifications changed and now are looks like "not threadsafe" (in specific sence) - they should be called only on the main thread. This fact is not documented. And may be a mistake.
2. By the Apple official documentation methods begin- and endGeneratingDeviceOrientationNotifications should be balanced. (Each begin- method should be balanced with end- method.)
By the reason two above facts they consequences merged and produced the "floating" NSInternalInconsistencyException crash.
Bug: webrtc:11216
Change-Id: Ibedd5bba7476cc687de3b9b04be49e3cceac1d27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162205
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30121}
This reverts commit f22af3cca7.
Reason for revert: Downstream tests have been updated.
Original change's description:
> Revert "Distinguish between send and receive video codecs"
>
> This reverts commit 18314bd8d2.
>
> Reason for revert: Breaks downstream test.
>
> Original change's description:
> > Distinguish between send and receive video codecs
> >
> > Even though send and receive codecs are the same,
> > they might have different support in HW.
> > Distinguish between send and receive codecs to be able to keep
> > track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30041}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30042}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1029737
Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30078}
Both cameraThreadHandler and surfaceHelper shouldn't be null.
Bug: None
Change-Id: I3c239c4275c53b836bbc2e9d6af71bf2b1b65387
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161480
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30047}
This reverts commit 18314bd8d2.
Reason for revert: Breaks downstream test.
Original change's description:
> Distinguish between send and receive video codecs
>
> Even though send and receive codecs are the same,
> they might have different support in HW.
> Distinguish between send and receive codecs to be able to keep
> track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30041}
TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30042}
Even though send and receive codecs are the same,
they might have different support in HW.
Distinguish between send and receive codecs to be able to keep
track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30041}
If the task to call OnEncodedImage is posted to the encoder task queue
just after VideoStreamEncoder::Stop post the task to release the
encoder, the destruction sequence of java HardwareVideoEncoder
deadlocks in outputBuffersBusyCount.waitForZero();
Encoders are generally allowed to call OnEncodedImage on any internal
encoder thread, so posting to the encoder task queue seems unnecessary.
Bug: webrtc:9378
Change-Id: Iee14f151d9efdc5ab348f9c86069fdb762e6a0dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161447
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30035}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889
Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
The NetEqFactory is currently expected to wrap the AudioDecoderFactory,
but this turns out not to be a good idea. Instead, it makes more sense
to pass the AudioDecoderFactory through the CreateNetEq method.
Bug: webrtc:11005
Change-Id: I8027ff6593f40c92072e7e88157631dcf329a984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160644
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29918}
Implementers of Java wrappers for native encoders need to have the same
implementation of all the unsupported methods, as mentioned in the
documentation of VideoEncoder.createNativeVideoEncoder (and its decoder
equivalent).
This simplifies implementation of such encoders/decoders, and also make sure
they don’t override unsupported methods, as they are guaranteed not to be
called.
Bug: None
Change-Id: Iaa8499eda1b52cc14b04622bea2766cd09ba43e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160186
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#29866}
This CL effectively expands the zone of influence of
https://webrtc-review.googlesource.com/64160,
forcing 16-byte stack alignment of generated JNI methods
for the Android x86 platform.
Bug: webrtc:9085
Change-Id: Idc40c00ea3fb52dbbbeac7b58ceda2a9a44733d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159928
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29858}
Those are preventive annotations to prepare for incoming android update
(coming with Chromium roll).
Currently the roll is blocked partly because errorprone complains!
Bug: webrtc:11095, chromium:1003532
Change-Id: If4e2879a522e895ce7fb1f2a9ad36d06f98f2a61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29830}
This method is called on every GetStats call and fills up log output on iOS
with three log lines per cycle at INFO+ (the not-supported one is LS_ERROR):
[181:040] [82471] (audio_device_module_ios.mm:646): GetPlayoutUnderrunCount
[181:040] [82471] (audio_device_generic.cc:48): GetPlayoutUnderrunCount: Not supported on this platform
[181:040] [82471] (audio_device_module_ios.mm:649): output: -1
Alternatively, we could remove the error logging in the base class, or (better) log it once the first time it is called, but this is the simpler change.
Bug: None
Change-Id: Ibaa1d176f10cdc92f2ba1a6bf15aaa580da6edb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159672
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29797}
This CL ensures that webrtc can work with an already-connected Wi-Fi
Direct network on Android Q.
Bug: None
Change-Id: Icf98c2f029fe0a92f95266310e6304268c2d9c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157504
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29579}
LogMessage::streams_ is a global and thus should have trivial destructor
Bug: None
Change-Id: Ie6a8029602f50b2bc5bab546ffc0365ef0954024
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157042
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29552}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
Merge GlobalLock and GlobalLockPod, make member private.
annotate creation of all GlobalLocks with ABSL_CONST_INIT
Bug: None
Change-Id: I29abcc86796ec0e45b15df7d26392309d1bf7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29447}
This happend because sdk_unittests is not built on arm/arm64 iOS build.
Bug: webrtc:11022
Change-Id: I8f9adfd48e11c8512c27992804cc9b69dff15ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156100
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29407}
Intended to be used for holding on to references to the java
EncodedImage and call its release method when no longer used by C++.
Bug: webrtc:9378
Change-Id: I40d917c2bb4217419ef2d609e517566c8466a274
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154740
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29347}
Before this change we always logged false in WebRTC.Audio.SourceMatchesRecordingSession
even when a test had not been executed (happens e.g. for Android < N).
This issue is now fixed and we only update WebRTC.Audio.SourceMatchesRecordingSession
if a valid test has been performed.
No-Try: True
TBR: glaznev
Bug: webrtc:10971
Change-Id: I907197476f00b812c67bb71e8fdcd6f297cfbdee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154563
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29324}
Callback set by HardwareVideoEncoder, and wired to the codec's
releaseOutputBuffer. Intention is to move call of this method to the
destructor of a corresponding C++ class in a followup cl, and
eliminate an allocation and memcpy in the process.
Bug: webrtc:9378
Change-Id: I578480b63b68e6ac7a96cdde36379b3c50f05c3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142160
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29283}
Goal is to be able to retrieve more details about possible microphone conflicts in
cases where Init/Start of audio recording fails.
Only supported on Android N and higher.
Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession.
Its value is stored after the recording session has been stopped.
Does not affect the media flow or functionality of the ADM. Time to start audio should
not be affected either since the new check and logging takes place on a separate
ExecutorService thread.
See go/webrtc-adm-android for more details and examples.
Bug: webrtc:10971
Change-Id: Ia80c1534e326907a1582824225d5f58caa016922
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29236}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
Now fixed issue which caused http://b/140707892
First version was reverted in https://webrtc-review.googlesource.com/c/src/+/152526.
The mistake I had done in the original version was that I missed that the new
builder could throw a different type of exception and it was never caught.
TBR: glaznev@webrtc.org
Bug: webrtc:10942
Change-Id: I0e11511936d2d25681a1ffae3bbd367095fee7a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152664
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29164}
This reverts commit 24b945d605.
Reason for revert: Caused http://b/140707892
Original change's description:
> Add support of AudioRecord.Builder in the ADM for Android
>
> Use the latest builder class for AudioRecord instead of the old
> constructor. AudioTrack has been updated for a while now.
>
> Bug: webrtc:10942
> Change-Id: Ia68b12e5aaf1525cfa630650fbaaa02d70ada15f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151305
> Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29072}
TBR=henrika@webrtc.org,glaznev@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10942
Change-Id: Idbc487cf8d42e76f6a3435be6fef6634aa0cd62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152526
Reviewed-by: Daixiang Mou <dmou@webrtc.org>
Commit-Queue: Daixiang Mou <dmou@webrtc.org>
Commit-Queue: Hari Molabanti <harimb@google.com>
Cr-Commit-Position: refs/heads/master@{#29159}
Use the latest builder class for AudioRecord instead of the old
constructor. AudioTrack has been updated for a while now.
Bug: webrtc:10942
Change-Id: Ia68b12e5aaf1525cfa630650fbaaa02d70ada15f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151305
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29072}
The new target does not depend on libjingle_peerconnection_api, and to
do this, the named "audio" and "video" string literals had to be moved from
media_stream_interface.cc to media_types.cc.
In this cl, the dependency on libjingle_peerconnection_api can be
dropped from a few targets.
No-Presubmit: True
Bug: webrtc:8733
Change-Id: Icc675280d5c3c537f2255a9389ff18a482049921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/53861
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28998}
This patch adds support for setting the TURN_LOGGING_ID
in RTCConfig using the android SDK.
TURN_LOGGING_ID was added to webrtc in
https://webrtc-review.googlesource.com/c/src/+/149829
The intended usage of this attribute is to correlate client and
backend logs.
bug: webrtc:10897
Change-Id: Ifd62e0f1dac396942c76a794bf7a75553d3244b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150538
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28996}
This reverts commit 44bd29a3b0.
Reason for revert:
Going for an alternative implementation that makes this unnecessary
https://webrtc-review.googlesource.com/c/src/+/150649
Original change's description:
> Detect leaks of TextureBufferImpl objects.
>
> The performance cost is not trivial but according to my profiling,
> it is acceptable.
>
> Bug: b/139745386
> Change-Id: I0e63221ccf22e9f6fb32c630ff63a279e765994a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150539
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28973}
TBR=sakal@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic6266e5fd24389d41a6d5dbfe51de6505b861b12
Bug: b/139745386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150650
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28983}
The performance cost is not trivial but according to my profiling,
it is acceptable.
Bug: b/139745386
Change-Id: I0e63221ccf22e9f6fb32c630ff63a279e765994a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150539
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28973}
Some devices have issues decoding the resolutions that result when using 4
as a factor.
Bug: webrtc:9381
Change-Id: I5055923ca318a1bde62bcefb452cae8f33165e43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150102
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28945}
This silences a warning that appeared with iOS 13, and is more efficient
in general.
Bug: webrtc:10866
Change-Id: I23db6b78af36e59b1d825d3f0cccc6008f9b626a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149808
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28911}
Bug: webrtc:10419
Change-Id: I18528bf2526e933568bf052de76a434f012161da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148320
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28838}
Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.
References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/https://stackoverflow.com/a/2524673
Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
This CL removes code related to the usage of the delay agnostic and
extended filter modes in AEC2.
Bug: webrtc:8671
Change-Id: I1a2c7a9eba54b03f5a015df3adb617785f52a939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133912
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28789}
1. Prevents deadlocks from AsyncInvoker destructor
2. Makes future state() calls are guaranteed to return the new state after
SetState() completes.
I am not sure if it is allowed to call FireOnChanged from non-signaling
threads so I will leave the post for now.
Bug: webrtc:10813
Change-Id: I5712a45f71431765898037867382397d537570a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147727
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28741}
This is a reland of 63741c7fa1
It's possible to reland after the downstream fix in
https://chromium-review.googlesource.com/1730415
Original change's description:
> Don't use all_dependent_configs for sdk frameworks
>
> libs should be propagated to the final binary even without that:
> https://gn.googlesource.com/gn/+/master/docs/reference.md#var_libs
>
> But add some missing SDK framework dependencies:
>
> * RTCNativeI420Buffer.mm uses CGBitmapContextGetBytesPerRow.
> * socketrocket uses SecCertificateCopyData.
>
> Bug: None
> Change-Id: Iba38a5dfaf470a5a790d494cbec8ade44b1d16ba
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146082
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28605}
Bug: None
Change-Id: I6a1cd80c5177ef3a3b92ee55fc91e187b202d864
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147720
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28740}
libs should be propagated to the final binary even without that:
https://gn.googlesource.com/gn/+/master/docs/reference.md#var_libs
But add some missing SDK framework dependencies:
* RTCNativeI420Buffer.mm uses CGBitmapContextGetBytesPerRow.
* socketrocket uses SecCertificateCopyData.
Bug: None
Change-Id: Iba38a5dfaf470a5a790d494cbec8ade44b1d16ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146082
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28605}
The GetImplementations function is similar to the GetSupportedFormats function, but instead of providing one SdpVideoFormat per codec it provides one per codec implementation. These SdpVideoFormats can then be tagged so that a certain implementation can be instantiated when CreateVideoEncoder is called.
Bug: webrtc:10795
Change-Id: I79f2380aa03d75d5f9f36138625abf3543c2339d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145215
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28553}
Users of webrtc generally should be able to choose own task queue implementation.
Poison avoids accidental dependency of a low level component on the default implementation
Android and ios apis are still de-facto forced to use the default implementation.
Bug: webrtc:10284
Change-Id: I67ecf2317f43ee32b0c9e8a6e69f1e0987cf1914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144786
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28524}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
Encountering GL_OUT_OF_MEMORY is relatively common and we should give
clients a chance to deal with it in a non-fatal way.
Bug: webrtc:8154
Change-Id: Ifa9ca74392f21083692b02a5144dc5632a88d34d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144561
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28495}
When provided, these thresholds will be used instead of WebRTC default
limits specified in DropDueToSize() and GetMaxDefaultVideoBitrateKbps().
Bug: none
Change-Id: Ida45ea832041963b8b8475d69114b5c60a172fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142170
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28390}
Using relative paths for JNI includes is causing build failures in chromium.
WebRTC already uses full include paths for generated JNI headers, so this CL
just removes the "jni_package" parameter from WebRTC generate_jni() targets
and removes the "jni/" portion of includes. The "jni_package" variable will be
removed from the generate_jni() template shortly.
To fix includes:
find . -name *.cc -exec sed -i -E 's@(#include.+generated.+jni)/jni/(.+_jni.h)@\1/\2@' {} \;
See https://groups.google.com/a/chromium.org/forum/?#!topic/java/MEovGrAwbqI
for discussion on naming scheme.
No-Try: True
TBR: kwiberg@webrtc.org
Bug: chromium:964169
Change-Id: I758c1b41bf6f5005587e55b82f14065fe251baad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143521
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28380}
instead of using components that rely on GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Icf7d1758b7f3ff6277b6a6d1b152715f0ab50969
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142800
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28367}
In https://chromium-review.googlesource.com/1650265 attributes like minSdkVersion were moved from AndroidManifest.xml to GN files. For WebRTC there were a few problems with that.
* We don't want to suppress UsesMinSdkAttributes lint but now there are these "invalid" manifest files that we can't exclude or discern. So disable this lint error.
https://chromium-review.googlesource.com/c/chromium/src/+/1650265/14/build/android/AndroidManifest.xml
* We should specify the versions in GN files, so I did that here (by exactly copying the versions that are already in the targets' corresponding XML files), but we never want to get rid of them in the XML files. For now this information will just be duplicated (without any synchronicity check!) so there should be followup to this.
Change log: 6ae0f0cd4c..bf62d746a4
Full diff: 6ae0f0cd4c..bf62d746a4
Changed dependencies
* src/base: 9e5e9332df..e5a1d1f652
* src/build: 5a031748ec..2ef566e990
* src/buildtools: 6ae683be2f..6f3775ad6e
* src/buildtools/linux64: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/mac: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/win: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/ios: 2f5c817266..7f1a97d593
* src/testing: 1d4247de57..b1b36ff0d4
* src/third_party: 6f7cbf7c46..42e96c4074
* src/third_party/android_sdk/public: ki7EDQRAiZAUYlnTWR1XmI6cJTk65fJ-DNZUU1zrtS8C..xhyuoquVvBTcJelgRjMKZeoBVSQRjB7pLVJPt5C9saIC
* src/third_party/android_sdk/public: iIwhhDox5E-mHgwUhCz8JACWQCpUjdqt5KTY9VLugKQC..ppQ4TnqDvBHQ3lXx5KPq97egzF5X2FFyOrVHkGmiTMQC
* src/third_party/android_sdk/public: 4Y2Cb2LGzoc-qt-oIUIlhySotJaKeE3ELFedSVe6Uk8C..MSnxgXN7IurL-MQs1RrTkSFSb8Xd1UtZjLArI8Ty1FgC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ed9fcf3f70..9e5dbd8b46
* src/tools: f58f33bca1..a9a4b8fc7b
DEPS diff: 6ae0f0cd4c..bf62d746a4/DEPS
No update to Clang.
Bug: chromium:891996
Change-Id: I773d6fa90e8083d934c84eecc1cb9d7d4496eca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142235
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28311}
Preparation for adding a release() method on java's EncodedImage, and
call that from C++.
Bug: webrtc:9378
Change-Id: I301f64b16684c535f45a3fc9cd9ae1543df59d92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141861
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28268}
This is a partial reland of
https://webrtc-review.googlesource.com/c/src/+/39680,
including only the (hopefully) non-problematic parts of it, but
postponing actual deletion of automatic thread wrapping.
Bug: webrtc:9714
Change-Id: I9b79dd073f0e945cbb62f3b54cff05eaaea9b06c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141664
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28265}
This is a reland of 11dfff0878
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
SurfaceTextureHelper currently crashes if an OES texture is produced
before setTextureSize() has been called. This is annoying if the texture
size is not easily known beforehand. A real world example is MediaPlayer
that provides the video size with an asynchronous call to
setOnVideoSizeChangedListener(), but that might happen after the first
texture is produced on some devices.
This CL waits with delivering frames until the size has been sent,
rather than crashing.
Bug: webrtc:10709
Change-Id: I5d9ce542e0edaafe1153fd5fe7d64dba86d7e33c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140080
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28151}
from a field trial to RTCConfiguration.
The test coverage is also expanded for the underlying feature.
Bug: None
Change-Id: Ic9c1362867e4a956c5453be7a9355083b6a442f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28143}
This CL adds the callback on changes of the ICE connection state
following the standardized transitions
(https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate) to the
Android and the iOS SDKs.
Bug: None
Change-Id: I6133391fa54dd4e09016f29dddb85e4a0e270878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138181
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28127}
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.
Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
Since many tests rely on rtc::Thread::Current(), add an
explicit rtc::AutoThread in the main() function used by tests.
Bug: webrtc:9714
Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28000}
This is a reland of e779847fb6
Original change's description:
> Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
>
> Also add explicit includes of rtc_base/string_utils.h in files depending on it.
>
> Bug: webrtc:6424
> Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27903}
Tbr: kwiberg@webrtc.org
Bug: webrtc:6424
Change-Id: Ic08d5d7fbc25ff89e4182d7c9cb3b0e8e356339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135946
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27957}
This fixes a potential crash if interrupted before the audio unit has been initialized.
Bug: None
Change-Id: Ib9f5ea305c98a172f8df52af5767c8543e59701c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136800
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27937}
Also add explicit includes of rtc_base/string_utils.h in files depending on it.
Bug: webrtc:6424
Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27903}
This will allow call audio to function when audio session attributes like `preferredInputNumberOfChannels` cannot be set due to intermittent OS errors.
Bug: webrtc:10602
Change-Id: Ie9f3e58a6ab54a26a9bd795575d16c3a9fe5c65f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135440
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27871}
This was added in CL 135122, but the bitratePriority parameter is not
standard and not implemented in a way users would expect. So it should
actually not be exposed in the Obj-C SDK.
Bug: webrtc:10438
Change-Id: I801ce940a32701d2703e951ef2b601c606aa2111
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135287
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27861}
This reverts commit 1fa06041bc.
Reason for revert: Likely cause for breaking downstream projects
Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}
TBR=steveanton@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
Change-Id: Iad7b7d4e37227fa6a76ff830160ca3da9dbe4719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132761
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27599}
This CL fixes the problem of misfired negotiationneeded notifications due
to the lack of a NegotiationNeeded slot and the proper procedure to
update it.
Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27594}
This is a reland of 7ac0d5f348
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,perkj@webrtc.org
Bug: webrtc:10481
Change-Id: I2978d5c527a18e885b7845c4e53a2424e8ad5b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132551
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27593}
This reverts commit 7ac0d5f348.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Replace usage of old SetRates/SetRateAllocation methods
>
> This rather large CL replaces all relevant usage of the old
> VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
> API is unchanged to allow downstream projects to update without
> breakage.
>
> Bug: webrtc:10481
> Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27554}
TBR=brandtr@webrtc.org,sakal@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,perkj@webrtc.org
Change-Id: I576760b584e3f258013b0279c0c173c895bbb37e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132561
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27559}
This rather large CL replaces all relevant usage of the old
VideoEncoder::SetRates()/SetRateAllocation() methods in WebRTC.
API is unchanged to allow downstream projects to update without
breakage.
Bug: webrtc:10481
Change-Id: Iab8f292ce6be6c3f5056a239d26361962b14bb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131949
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27554}
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.
Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.
Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.
Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
This is a reland of 177670afd6
Fixing failing tests.
TBR=magjed@webrtc.org
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
Bug: webrtc:10464
Change-Id: I95fac3967217c20a9fdddb490aea30eca2061ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130362
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27402}
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This reverts commit 177670afd6.
Reason for revert: Fails android_instrumentation_test_apk:
https://ci.chromium.org/p/webrtc/builders/ci/Android64%20(M%20Nexus5X)/11553
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
TBR=magjed@webrtc.org,shampson@webrtc.org,amithi@webrtc.org
Change-Id: Id6c4e2d41c3c2fbfad31baed907cfa73d82ef14a
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130466
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27354}
Since it is a WebRTC-only macro, let's prefix it with WEBRTC_.
Bug: None
Change-Id: I309666858ea898dc7cd1a68c21be190f98c87b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129935
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27327}
This adds the bindings for rid in RtpParameters.Encoding and bindings
for send_encodings in RtpTransceiverInit to allow creating a transceiver
with multiple send encodings.
Bug: webrtc:10464
Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27323}
This adds bindings for RIDs in RtpEncodingParameters.
Bug: webrtc:10464
Change-Id: I3cc25db25a4d777b9d9573ba69c82127d1c9a597
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128826
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27322}
Even if neither frame height nor frame width is <=0 we can end up
with <=0 dimensions in renderHeight or renderWidth. With this change,
we perform the check on the latter.
Bug: webrtc:10367
Change-Id: I9672672659ad7d12cf1e7ccab5b5cde583ae7e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129760
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27307}
Injection is made possible through VP8Encoder::Create.
According to native-api.md, it is a defacto public API despite
not being in the api/ folder.
Bug: webrtc:10259, webrtc:10382
Change-Id: Ifc5d55aa99613cfee0fcb4f0c6690121c85b2e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27281}
After being stuck "forever" (3 seconds) waiting for an event to
trigger, log the stack trace of the current thread to aid debugging of
the deadlock.
Bug: webrtc:10308
Change-Id: I04852f191027294d7e7a7f5e63de4c6c7fdd6326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128342
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27263}
Use rtc::scoped_refptr instead of std::unique_ptr to hold the instance
of OpenSLEngineManager; this makes it safe to share it between
OpenSLESRecorder and OpenSLESPlayer.
Bug: webrtc:10436
Change-Id: Ibd0717e5410020c89a40bfdb05953a02378a6a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128651
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27253}
We may sometimes want to override only input or only output, or
override them with different values. This change allows setting the
overrides separately.
Change-Id: Ib0c44cb7a3cfa834f997fb6cd54f7cad68705f41
Bug: webrtc:10441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128763
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27236}
