This reverts commit da2fd2a2b25ee4bd7b383424cb26d51fb6cc7716,
as well as follow-up b7227a5a10f233cec04642f15a0233e7355bd340,
"Fix handling of partial match for GetVpnUnderlyingAdapterType".
Reason for revert: Breaks downstream test.
First change's description:
> Fix problem with ipv4 over ipv6 on Android
>
> This patch fixes a problem with using ipv4 over ipv6
> addresses on Android. These addresses are discovered
> using 'getifaddr' with interfaces called 'v4-wlan0' or
> 'v4-rmnet' but the Android API does not report them.
>
> This leads to failure when BasicPortAllocator tries
> to bind a socket to the ip-address, making the ipv4
> address unusable.
>
> This solution does the following
> 1) Insert BasicNetworkManager as NetworkBinderInterface
> rather than AndroidNetworkManager.
>
> 2) When SocketServer calls BindSocketToNetwork,
> BasicNetworkManager first lookup the interface name,
> and then calls AndroidNetworkManager.
>
> 3) AndroidNetworkManager will then first try to bind
> using the known ip-addresses, and if it can't find the network
> it will instead match the interface names.
>
> The patch has been tested on real android devices, and works fine.
> And everything is disabled by default, and is enabled by field trial.
>
> My plan is to rollout the feature, checking that it does not introduce
> any problems, and if so, enabled for all.
>
> Bug: webrtc:10707
> Change-Id: I7081ba43d4ce17077acfa5fbab44eda127ac3971
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211003
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33422}
Second change's description:
> Fix handling of partial match for GetVpnUnderlyingAdapterType
>
> This is a followup to https://webrtc-review.googlesource.com/c/src/+/211003
> and fixes the problem pointed out by deadbeef@, thanks!
>
> Bug: webrtc:10707
> Change-Id: I8dea842b25ba15416353ce4002356183087873c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211344
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33436}
TBR=hta@webrtc.org,jonaso@webrtc.org
NOTRY=True
Bug: webrtc:10707
Change-Id: Ib13127fbf087c7f34ca0ccc6ce1805706f01d19d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211740
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33453}
Use PendingTaskSafetyFlag for safe Stop. Followup to
https://webrtc-review.googlesource.com/c/src/+/209181.
Also fix rtc::scoped_refptr to work with RTC_PT_GUARDED_BY.
Bug: webrtc:12339
Change-Id: Ic0e3ecb17049f1a0e6af887ba5f97a5b48a32d98
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211351
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33447}
Removes use of AsyncInvoker, replaced with PendingTaskSafetyFlag. The
latter is extended to support creation on a different thread than
where it will be used, and to support stop and restart.
Bug: webrtc:12339
Change-Id: I28b6e09b1542f50037e842ef5fe3a47d15704b46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211002
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33432}
This patch fixes a problem with using ipv4 over ipv6
addresses on Android. These addresses are discovered
using 'getifaddr' with interfaces called 'v4-wlan0' or
'v4-rmnet' but the Android API does not report them.
This leads to failure when BasicPortAllocator tries
to bind a socket to the ip-address, making the ipv4
address unusable.
This solution does the following
1) Insert BasicNetworkManager as NetworkBinderInterface
rather than AndroidNetworkManager.
2) When SocketServer calls BindSocketToNetwork,
BasicNetworkManager first lookup the interface name,
and then calls AndroidNetworkManager.
3) AndroidNetworkManager will then first try to bind
using the known ip-addresses, and if it can't find the network
it will instead match the interface names.
The patch has been tested on real android devices, and works fine.
And everything is disabled by default, and is enabled by field trial.
My plan is to rollout the feature, checking that it does not introduce
any problems, and if so, enabled for all.
Bug: webrtc:10707
Change-Id: I7081ba43d4ce17077acfa5fbab44eda127ac3971
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211003
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33422}
This is consistent with other Notify methods in this class, which
handle callbacks from java using blocking invokes to the network
thread.
This eliminates the use of the deprecated AsyncInvoker class.
Bug: webrtc:12339
Change-Id: Ib2d19b37b8f669df5b97e89d720f6eb6fc9e5517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209181
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33398}
Some of the PCF and PC methods are actually return nil, but was by
default annotated as nonnull via NS_ASSUME_NONNULL_BEGIN.
Bug: None
No-Presubmit: True
Change-Id: Ib8b9263452a61241c9e7a16c1807d87bd597c093
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/209180
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33384}
It is now possible to use AV1 encoder and decoder on iOS and test
them in apps like AppRTCMobile.
Bug: None
Change-Id: Ifae221020e5abf3809010676862eecd9ffeec5e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208400
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33378}
Sometimes c2.qti.vp8.decoder reports format updates with zero frame
width / height right after initialization, that leads to the
precondition check failure made by SurfaceTextureHelper.setTextureSize.
This patch makes AndroidVideoDecoder.reformat to ignore such format
updates so as to continue to use this HW decoder.
It seems to be safe because this decoder singals one more format update
with valid dimensions soon and continue to operate in normal mode.
Bug: webrtc:12492
Change-Id: I5155166637bd2d4247d31e608d714e687e0ad1df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208222
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33332}
- Use matching config to avoid EGL_BAD_MATCH.
- Use the same display in both eglMakeCurrent calls to avoid
EGL_BAD_ACCESS on subsequent calls because the context was not
successfully unbound.
Bug: webrtc:12471
Change-Id: Ifdf4bd94cdfd14b683959b8703d75a2a46ec1226
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207861
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33279}
Some locations in the WebRTC codebase RTC_LOG the value of the
__FUNCTION__ macro which probably is useful in debug mode. Moving
these instances to RTC_DLOG saves ~10 KiB on arm64.
Bug: webrtc:11986
Change-Id: I5d81cc459d2850657a712b9aed80c187edf49a3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203981
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33086}
In AudioDeviceIOS, when we call Stop() on the VoiceProcessingAudioUnit,
we do not always detach the I/O thread checker in preparation for a new
start. This means that if we start up the VoiceProcessingAudioUnit - and
subsequently a new AURemoteIO thread to deal with I/O operations - the
DCHECK in OnDeliverRecordedData and OnGetPlayoutData will fail. Note
that we want to detach the I/O thread checker regardless of whether
Stop() returns with a success status or not. The success status is
dictated by the iOS function AudioOutputUnitStop. The documentation of
this function does not guarantee that the audio unit will not stop in
the case the function returns with an error code. That is to say, it is
possible the audio unit stops even if the function Stop() returns false.
Therefore, it is safer to prepare the I/O thread checker for a new start
in either case.
Change-Id: Iee50a2457959aff2e6089e9a664c649dc4dbbbd6
Bug: webrtc:12382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202945
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33063}
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a
Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
Ensures the state of the class remains correct even if an unhandled
exception is thrown from this method.
Bug: b/176214704
Change-Id: I94504bb8aa4bd2dba45d116d5fa13da070a3b60f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201621
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32963}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
This is just general cleanup.
The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).
Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
While RTC_EXPORT is aware of component builds (selecting "default"
visibility only when WebRTC is built as a shared library),
RTC_OBJC_EXPORT (which predates RTC_EXPORT) was always marking symbols
as "default" visible.
This CL fixes the problem but on the other hand it will require
standalone builds of the WebRTC.framework to set the GN argument
`rtc_enable_symbol_export` to true.
No-Presubmit: True
Bug: chromium:1159620
Change-Id: I4a16f9bd3c1564140a5a30f03b3e77caed1df591
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198082
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32856}
This CL makes it possible to use a low-latency mode on Android O and later. This should help to reduce the audio latency. The feature is disabled by default and needs to be enabled when creating the audio device module.
Bug: webrtc:12284
Change-Id: Idf41146aa0bc1206e9a2e28e4101d85c3e4eaefc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196741
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@{#32854}
AudioAttributes::getAllowedCapturePolicy was added in API Level 29.
Update WebRtcAudioTrack to add API Level check before using the API.
Bug: webrtc:12250
Change-Id: Ica6604eb1d7fa736a0e64729a022eefcfb7b3020
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195941
Commit-Queue: Gaurav Vaish <gvaish@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32735}
This CL renames webrtc guava dependencies from
third_party/guava:guava_android_java to
//third_party/android_deps:guava_android_java
This is in preparation for deleting third_party/guava:guava_android_java
BUG=chromium:2560401
No-Presubmit: True
Change-Id: If9227f4ac4d24386896c47eeb38142a76a27a4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195720
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32730}
This patch adds support for setting the TURN_LOGGING_ID
in RTCConfig using the ios 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.
This change was tested out with duo via wireshark.
Bug: webrtc:10897
Change-Id: Iedbefdc6392c4df203aca08cf750028b450a11ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191340
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Brad Pugh <bradpugh@google.com>
Cr-Commit-Position: refs/heads/master@{#32626}
In particular move end_of_picture flag out of vp9 specific information
since VP9 is not the only codec that can use spatial scalability and
thus need to distinguish layer frame and picture (aka temporal unit).
Bug: webrtc:12167
Change-Id: I0d046d8785fbea55281209ad099738c03ea7db96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192542
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32588}
WebRtcAudioTrack is hardcoded to configure AudioAttributes with
1. usage=USAGE_VOICE_COMMUNICATIOON
2. contentType=CONTENT_TYPE_SPEECH
This change allows AudioAttributes to be configured via the
JavaAudioDeviceModule.
Bug: webrtc:12153
Change-Id: I67c7f6e572c5a9f3a8fde674b6600d2adaf17895
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191941
Commit-Queue: Gaurav Vaish <gvaish@chromium.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32583}
