This cl/ fixes a race condition with the recent additions
to NetworkMonitorAutoDetect (getAllNetworksFromCache).
The getAllNetworksFromCache-feature uses the by the Android team preferred way of
enumerating networks, i.e to register network listeners.
This however introduces a unpleasant race condition like this:
1) network.cc discover rmnet0
2) BasicPortAllocator tries to create UDP port on rmnet0
- This fails as BindSocketToNetwork requires a android handle.
3) NetworkMonitorAutoDetect gets callback with rmnet0
4) BasicPortAllocator tries to create TCP port on rmnet0
- This succeeds.
5) Since rmnet0 has one working port, there will not be
any new ports created on that network
=> We end up with a TCP only connection :(
---
By impl. IsAdapterAvailable, we make sure that the network
will not be used by BasicPortAllocator (or anyone else!)
until we support binding to it.
The IsAdapterAvailable was implemented for IOS,
and has test coverage using FakeNetworkManager.
This cl/ is default enabled with the kill-switch
WebRTC-AndroidNetworkMonitor-IsAdapterAvailable.
Bug: webrtc:13741
Change-Id: I7c2cb7789660fd2e082c214d00e50c894666b07c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36406}
* tools_webrtc/PRESUBMIT.py is only checking the licence which is already done here:
38f35db4d4:PRESUBMIT.py;l=913;bpv=1;bpt=0;drc=4fc9bd9f69a0d88889d86d0cc9f8e27406e8a342
* sdk/android/PRESUBMIT.py was added before 'git cl format' was required from the root PRESUBMIT.py:
https://codereview.webrtc.org/2377113003
Bug: webrtc:13895
Change-Id: Ia5ea2529c36ceebfd7d4e6a6a72352bd30c573b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257280
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36391}
Ironic :( The "field trial guy" constructing a invalid string,
if only there would have been a builder instead...
I tested the code several times...but not with debug build...
Bug: webrtc:13741
Change-Id: If3caad0f5533fc150ffd6a34a89ab84f3f0264aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256979
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36370}
convert rtc_base/network and collateral.
This also remove last usage of system_wrappers/field_trials
in p2p/...Yay!
Bug: webrtc:10335
Change-Id: Ie8507b1f52bf7f3067e9b4bf8c81a825e4644fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36357}
The only caveat is that a name attribute for the `@Parameters`
annotation is required, as otherwise the test infrastructure
doesn’t find test results.
Bug: webrtc:13662
Change-Id: Ib3e2a6671d1045b0e19746ce78dd434fbee78b87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256462
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Auto-Submit: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36322}
This migrates all tests that work by just changing their runner.
This excludes tests using `@RunWith(ParameterizedRunner.class)`, and a
few other non-parameterized tests that fail with the default runner.
Bug: webrtc:13662
Change-Id: Ia0b7c80e04a6a6b7a51348b3a7f587d10061b58e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256367
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36298}
This is part of a large-scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL converts the majority of "const std::string&"s in function
parameters under rtc_base/ to absl::string_view.
Bug: webrtc:13579
Change-Id: I2b1e3776aa42326aa405f76bb324a2d233b21dca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254081
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Anders Lilienthal <andersc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36239}
This is part of a large scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL adds absl::string_view versions of the OnLogMessage functions in
rtc::LogSink. The const std::string& versions are kept for now since
downstream clients use subclasses of LogSink and need to be migrated
before these are removed.
Bug: webrtc:13579
Change-Id: I57bb72ad503805ff0ca16f1d7aece2d44c65cb73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Owners-Override: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36189}
This is in preparation to introduce new java buildtargets that will use the `libaom_av1_encoder` buildtarget instead.
bug: webrtc:13573
Change-Id: I23e80653943ede576657acc17bcc5602cb0a4d5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254540
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36171}
When loading the library fails, the user will be faced with this error:
java.lang.UnsatisfiedLinkError: No implementation found for void org.webrtc.PeerConnectionFactory.nativeInitializeAndroidGlobals()
With no context, however.
Bug: webrtc:13619
Change-Id: I88565f085773ad1e8c2f5742d7fdba96fb6043d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253960
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36150}
WebRTC’s minSdk is 21, so all those checks are dead code.
Change-Id: I26497fd92259b66d9e5ac6afbb393adf4d904c77
Bug: webrtc:13780
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253124
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Linus Nilsson <lnilsson@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36140}
This change adds a cache for networks in the SimpleNetworkCallback that
is already registered, allowing the cache to be used preferentially as
opposed to the deprecated getAllNetworks call.
This is a fork of https://webrtc-review.googlesource.com/c/src/+/251401
- adds field trials for new behavior
- removes test that did not work
- add (poor) test of field trials
- remove the "network_monitor_java" build target (that I could
not find any reference to...)
Bug: webrtc:13741
Change-Id: I2829c2f1940d4b42455d8e1a2217cf15c133e22b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252284
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36121}
On Android, MediaCodec can request a specific layout of the input buffer.
One can use the stride and slice height to calculate the layout from
the Encoder's MediaFormat. The current code assumes
a specific layout, which is a problematic in Android 12.
Fix this by honoring the stride and slice-height.
Bug: webrtc:13427
Change-Id: I2d3e429309e3add3ae668e0390460b51e6a49eb9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/240680
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#36033}
These two files turn out to be entangled with each other. Keeping
them together for now.
This is a simpler approach than the one attempted in
https://webrtc-review.googlesource.com/c/src/+/251060
but leaves cleanup of the relationship to a later work item.
Bug: webrtc:13634
Change-Id: I2b38f86c0c510332dc24a6b83531aee143a5df10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35942}
Add timestamps to audio_record_jni DataIsRecorded() function, and make
WebRtcAudioRecord find and send the time stamp to that function.
This CL is an continuation of
https://webrtc-review.googlesource.com/c/src/+/249085
Bug: webrtc:13609
Change-Id: I63ab89f1215893cbe1d11d9d8948f5639fc5cdfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249951
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35933}
This CL removes even more top-level const from parameters in function
declarations. This change is safe because top-level const in function
declarations (not function definitions) are ignored by the compiler
and so change is just a no-op cleanup.
Bug: webrtc:13610
Change-Id: Icf6868c27b1fdb9d9915b3a7020eb34bdcf07a09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249989
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35866}
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will
be used to store audio timestaps in future changes.
This is a part of the A/V sync metric metric feature for mobile. The metric
have already launched for web clients.
Bug: webrtc:13609
Change-Id: I0031843476ff1b573b262308fca52d587fae30b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Minyue Li <minyue@google.com>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#35851}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
Because rtc::Thread inherits from TaskQueueBase, it already implements
a pair of PostTask/PostDelayedTask methods that we want to keep. But in
addition to those, rtc::Thread defines its own PostTask/PostDelayedTask
using templates. These are the versions that we want to deprecate.
They were originally implemented prior to rtc::Thread inheriting from
TaskQueueBase. We want to deprecate them because...
- We don't want to have multiple code paths that do the same thing.
- We want to move away from rtc::Thread to TaskQueueBase long-term.
- These versions are not overridable in Chromium.
- These versions don't have high/low precision versions of PDT.
Helper methods are added to rtc::Thread so that callers don't have to
wrap every lambda in webrtc::ToQueuedTask() and update dependencies.
Bug: webrtc:13582
Change-Id: I58702c53f4cb3705681bd9f1ea16b7aaa5052c18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35750}
Updates all webrtc code, to have a small followup cl to just add the
"explicit" keyword. Patchset #24 passed all webrtc tests, with explicit.
Bug: webrtc:13464
Change-Id: I39863d3752f73209b531120f66916dc9177bf63a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242363
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35718}
The default value of sdpSemantics is about to change from PLAN_B to
UNIFIED_PLAN. In order not to cause subtle bugs by applications that
depend on the default value being PLAN_B, we are temporarily making the
default NOT_SPECIFIED. Constructing with NOT_SPECIFIED causes the C++
layer to crash (https://webrtc-review.googlesource.com/c/src/+/242968).
This is in accordance to the publically announced plans:
https://groups.google.com/u/1/g/discuss-webrtc/c/SdoVP02eUIk
While we're at it, we're upgrading almost all unit tests to use Unified
Plan. However there are still two tests using Plan B for which I added
TODO comments to be dealt with later; not having an Android setup makes
it impossible to debug these efficiently.
Bug: webrtc:11121
Change-Id: Ib086186bee947d18d31b413e3aeba0cb247b377d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246000
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35700}
This also removes all internal usage of RemoveTrack, and changes
the replacement function to RemoveTrackOrError rather than RemoveTrackNew.
Bug: webrtc:9534
Change-Id: Idf7bb17495686de77c70428dcbfb12278328ce59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244094
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35624}
After changing the way libunwind is built in https://crrev.com/c/3297439,
this test should work fine.
Bug: webrtc:13383
Change-Id: I5da7bf27ce3041c934d4ab91367a26c076fac0c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35558}
This CL adds the callback on ICE Candidate Error to the Android and
the iOS SDKs.
Spec: https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-onicecandidateerror
Bug: webrtc:13446
Change-Id: I6e511aaa80f1aa8f4310d8518d1144d97470cd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239460
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35531}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This is part of the removal of support for SDES.
Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}
It seems the Android CTS tests only verify that 16x16 aligned resolutions
are supported.
This change checks the validity of input frame's size when initialing
or encoding processes are about to start using H/W MediaCodec.
This change has additional APIs to retrieve
|requested_resolution_alignment| and |apply_alignment_to_all_simulcast_layers|
from JAVA VideoEncoder class and its inherited classes. HardwareVideoEncoder
using MediaCodec has values of 16 and true for above variables.
Bug: webrtc:13089
Change-Id: I0c4ebf94eb36da29c2e384a3edf85b82e779b7f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229460
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35169}
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".
Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
at org.webrtc.Camera2Session$CaptureSessionCallback.chooseStabilizationMode(Camera2Session.java:234)
at org.webrtc.Camera2Session$CaptureSessionCallback.onConfigured(Camera2Session.java:172)
at android.hardware.camera2.impl.CallbackProxies$SessionStateCallbackProxy.lambda$onConfigured$0(CallbackProxies.java:53)
at android.hardware.camera2.impl.-$$Lambda$CallbackProxies$SessionStateCallbackProxy$soW0qC12Osypoky6AfL3P2-TeDw.run(-.java:4)
at android.os.Handler.handleCallback(Handler.java:873)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:193)
at android.os.HandlerThread.run(HandlerThread.java:65)
Bug: webrtc:13032
Change-Id: Ifb6ef920b700ca03d37c64803c0b34230785846f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227292
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34874}
StreamConfigurationMap.getOutputSizes() may return null:
https://developer.android.com/reference/android/hardware/camera2/params/StreamConfigurationMap#getOutputSizes(java.lang.Class%3CT%3E)
Fixes:
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
at org.webrtc.Camera2Enumerator.convertSizes(Camera2Enumerator.java:234)
at org.webrtc.Camera2Enumerator.getSupportedSizes(Camera2Enumerator.java:147)
at org.webrtc.Camera2Session.findCaptureFormat(Camera2Session.java:325)
at org.webrtc.Camera2Session.start(Camera2Session.java:313)
at org.webrtc.Camera2Session.<init>(Camera2Session.java:296)
at org.webrtc.Camera2Session.create(Camera2Session.java:274)
at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:237)
at android.os.HandlerThread.run(HandlerThread.java:67)
Bug: webrtc:13032
Change-Id: I9154be567cd12c066087818ba22e9cd69e75a22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227291
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34872}
Remove android.support.test.rule.UiThreadTestRule as chromium did in [1] and
Replace android.support.test.annotation.UiThreadTest
with org.chromium.base.test.UiThreadTest.
Also remove unused uiThreadHandler from NetworkMonitorTest.
[1] https://crrev.com/c/2332301
Bug: webrtc:11962
Change-Id: I8f3781d43d4d53d8158c39c81568d8b09b2bec6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230220
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#34864}
Specifically, defer getting the camera index so the error can be
reported instead of crashing:
Fatal Exception: java.lang.IllegalArgumentException: No such camera: Camera 1, Facing front, Orientation 270
at org.webrtc.Camera1Enumerator.getCameraIndex(Camera1Enumerator.java:170)
at org.webrtc.Camera1Capturer.createCameraSession(Camera1Capturer.java:31)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:790)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:214)
at android.os.HandlerThread.run(HandlerThread.java:65)
Bug: webrtc:13032
Change-Id: Ida6bc65046770c11c2b3ee832906e8454cec10df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227290
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34855}
getCameraCharacteristics() may throw IllegalArgumentException:
Fatal Exception: java.lang.IllegalArgumentException: supportsCameraApi:2569: Unknown camera ID 1
at android.hardware.camera2.CameraManager.throwAsPublicException(CameraManager.java:1119)
at android.hardware.camera2.CameraManager.getCameraCharacteristics(CameraManager.java:531)
at org.webrtc.Camera2Session.start(Camera2Session.java:304)
at org.webrtc.Camera2Session.<init>(Camera2Session.java:296)
at org.webrtc.Camera2Session.create(Camera2Session.java:274)
at org.webrtc.Camera2Capturer.createCameraSession(Camera2Capturer.java:35)
at org.webrtc.CameraCapturer$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:237)
at android.os.HandlerThread.run(HandlerThread.java:67)
Bug: webrtc:13032
Change-Id: I30b6d6da40bc90a94c0c3c79f9dff523182d3da4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227289
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34854}
Fatal Exception: java.lang.NullPointerException: Attempt to read from field 'int org.webrtc.CameraEnumerationAndroid$CaptureFormat.width' on a null object reference
at org.webrtc.Camera2Session$CameraStateCallback.onOpened(Camera2Session.java:122)
at android.hardware.camera2.impl.CameraDeviceImpl$1.run(CameraDeviceImpl.java:151)
at android.os.Handler.handleCallback(Handler.java:938)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:246)
at android.os.HandlerThread.run(HandlerThread.java:67)
Fix NPE when setting the camera2 stabilization mode
Fatal Exception: java.lang.NullPointerException: Attempt to get length of null array
at org.webrtc.Camera2Session$CaptureSessionCallback.chooseStabilizationMode(Camera2Session.java:234)
at org.webrtc.Camera2Session$CaptureSessionCallback.onConfigured(Camera2Session.java:172)
at android.hardware.camera2.impl.CallbackProxies$SessionStateCallbackProxy.lambda$onConfigured$0(CallbackProxies.java:53)
at android.hardware.camera2.impl.-$$Lambda$CallbackProxies$SessionStateCallbackProxy$soW0qC12Osypoky6AfL3P2-TeDw.run(-.java:4)
at android.os.Handler.handleCallback(Handler.java:873)
at android.os.Handler.dispatchMessage(Handler.java:99)
at android.os.Looper.loop(Looper.java:193)
at android.os.HandlerThread.run(HandlerThread.java:65)
Bug: webrtc:13032
Change-Id: I6edd9f0061c445f90ab0881d78183077f89e391f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227294
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34851}
This change does not affect downstream dependencies as androidx.annotation
is fully compatible with android.support.annotation.
Bug: webrtc:11962
Change-Id: I714b473df8d0fee8000ddf3a9beca7c5613db5ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226881
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34839}
PeerConnectionFactory to break off the dependency.
- This is required so that Android app that doesn't use the
peerconnection_java as dependency can include android monitor
directly without incurring size bloat.
Bug: None
Change-Id: I7b3453f268467550c0a4b3a0bbf858d55d2fd8a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229322
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34829}
Before this change HardwareVideoEncoder used capture time as frame
timestamp passed to HW encoder. That led to buffer overshoots with
HW encoders which infer frame rate from timestamps when frames were
dropped before encoding (i.e., frame rate decreases according to frame
timestamps) or when FramerateBitrateAdjuster was used.
Fixed this by using synthetic monotonically increasing timestamps
calculated based on target frame rate provided by bitrate adjuster.
Bug: webrtc:12982
Change-Id: I2454cd4e574bbea1cb9855ced4d998104845415c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228902
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34810}
This allows to get encoder implementation name and other properties
without the need of initializing encoder.
Bug: none
Change-Id: I263a358d562a65a31c420ddb7c4b195316fa5ec8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226867
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34559}
Need someone from video team devs to be in the list. Working on projects
related to Android media codecs for couple of years and have enough
experience to review the changes. A concrete short-term motivation is
the need to land https://webrtc-review.googlesource.com/c/src/+/226867
Bug: none
Change-Id: I1d0a672f6b497bbe1e2d446386284d568f84664a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226951
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34556}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
Just like the C++ API, add a method in Java VideoFrame.Buffer that
describes the underlying implementation.
Use this method to properly select AndroidVideoBuffer
or AndroidVideoI420Buffer in Java -> C++ Video Frame Conversion.
Also, add a test case for WrappedNativeI420Buffer
in VideoFrameBufferTest for consistency.
Bug: webrtc:12602
Change-Id: I4c0444e8af6f6a1109bc514e7ab6c2214f1f6d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223080
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34545}
This is a step to ensure that the Java to C++ Video Frame Buffer
conversion respects its types.
Bug: webrtc:12602
Change-Id: I1b688b1f421f44474e022b433f9075e75744d86f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223082
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34487}
Also changed the logging of exceptions to give more details
Bug: webrtc:10804
Change-Id: Ifba6dee3d1c8ba4ecab408ca7715c3b792d9c004
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222641
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34325}
This method has been deprecated since 2018-07:
https://webrtc-review.googlesource.com/c/src/+/88368/
It is never called by WebRTC itself.
Custom `VideoDecoderFactory` implementations overriding this method can
switch to the overload accepting a `VideoCodecInfo` object.
This is also adding a `toString()` implementation to `VideoCodecInfo`,
to make logging of the value more useful.
Bug: webrtc:7925
Change-Id: I70ec07a0cd4ffa07d165c9851e393439fcc5870b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221960
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34302}
AV1X->AV1 mapping added to SdpVideoFormatToVideoCodecInfo in
https://webrtc-review.googlesource.com/c/src/+/215586 results in
discrepancy of codec name between SDP and VideoCodecInfo. That violates
VideoCodecInfo design and breaks downstream projects.
This CL moves the mapping from VideoCodecInfoToSdpVideoFormat and
SdpVideoFormatToVideoCodecInfo to VideoCodecTypeMime.
Bug: b/181690054
Change-Id: I2a76524c29b082241f2ec72a60a209ce9b0c7c5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221205
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34230}
Change-Id: I0880caa77a1097f56c560152e85c9ca29242f825
This PR add support for the `PeerConnectionObserverJni::OnRemoveTrack()`
event on Java, allowing to be notified when a remote track has been
removed. It's a very thing JNI wrapper on top of C++ API, being mostly
similar to other already available events like `track` and `addTrack`.
In Javascript API, tracks are not "removed" explicitly from the
PeerConnection, but instead receiver PeerConnection gets notified that
they have been removed from the streams they are associated to, and when
no `MediaStream` object has that track, it's considered that the track
has been removed from the PeerConnection. In Java and C++ APIs there's no
`MediaStreamObserver` class, so there's no way to listen to the
`removeTrack` event the same way happens in Javascript API, but instead
C++ API has a `removeTrack` event at PeerConnection level. This patchset
just only wraps and expose this `removeTrack` event from the C++ API to
the Java API.
This PR has been sponsored by Atos Research and Innovation
(https://atos.net/en/about-us/innovation-and-research).
Bug: webrtc:12850
Change-Id: I0880caa77a1097f56c560152e85c9ca29242f825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218847
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34225}
Deprecate CreateDataChannel, and make it a simple wrapper function.
Bug: webrtc:12796
Change-Id: I053d75a264596ba87ca734a29df9241de93a80c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219784
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34130}
This is one step in getting rid of cricket::MediaType.
Bug: webrtc:12754
Fixes: webrtc:12764
Change-Id: Idee832572bdc4c0e3bfdec6fb31ec0ba9db3e995
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218346
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33994}
Applications should use CreatePeerConnectionOrError instead.
Moved fallback implementations of CreatePeerConnection into the
api/peer_connection_interface.h file, so that we do not have to
declare these methods in the proxy.
Bug: webrtc:12238
Change-Id: I70c56336641c2a108b68446ae41f43409277a584
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217762
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33964}
This reverts commit 793bac569f.
Reason for revert: rare compilation error fixed
Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> > a) Handles for detached threads are simply closed where applicable.
> > b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
But keep option to disable it.
The fix has been tested using A/B experiment with Duo
with positive results.
Bug: webrtc:10707
Change-Id: Iaac72430bc52026ab5db9dd042c5fdc3832fdce5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217380
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33941}
This reverts commit c89fdd716c.
Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
> a) Handles for detached threads are simply closed where applicable.
> b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=handellm@webrtc.org
Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
On a 32bit system, this reduces the allocation size of the flag
down from 12 bytes to 8, and removes the need for a vtable (the extra
4 bytes are the vtable pointer).
The downside is that this change makes the binary layout of the
flag, less compatible with RefCountedObject<> based reference counting
objects and thus we don't immediately get the benefits of identical
COMDAT folding and subsequently there's a slight binary size increase.
With wider use, the binary size benefits will come.
Bug: none
Change-Id: I04129771790a3258d6accaf0ab1258b7a798a55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33793}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
This denies the ability to request RTP data channels to callers.
Later CLs will rip out the actual code for creating these channels.
Bug: chromium:928706
Change-Id: Ibb54197f192f567984a348f1539c26be120903f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177901
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33740}
"enableImplicitRollback" is necessary for perfect negotiation algorithm
"offerExtmapAllowMixed" is necessary for backward compatibility with
legacy clients.
Bug: webrtc:12609
Change-Id: I30a5a01c519ca9080a346e2d36b58f7bab28f15a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212741
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33639}
PC.restartIce() is part of perfect negotiation algorithm.
Bug: webrtc:12609
Change-Id: I21a0f8637e92e13ee2653ef477d0cd22a32bf9c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212645
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33589}
The motivation is making it easier to catch exceptions for these
kind of failures only.
Bug: b/182561645
Change-Id: I09527d8665fda0fa24144cb05e9fd24c041549a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212608
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33540}
Instead of using SetAlive on the old flag (which might allow old
tasks in the queue to run).
Bug: webrtc:12339
Change-Id: Ia1a3eb6932f62881f013fd62b0e008d97d8713cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211863
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33475}
This reverts commit 1e60490ddb.
Reason for revert: Downstream project has been fixed (127.0.0.1 is not PII)
Original change's description:
> Revert "Fix problem with ipv4 over ipv6 on Android"
>
> This reverts commit da2fd2a2b2,
> as well as follow-up b7227a5a10,
> "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}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10707
Change-Id: I0a11025c366c3127e2f57cd2cd2c33cc3877d1e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211862
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33460}
This reverts commit da2fd2a2b2,
as well as follow-up b7227a5a10,
"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}
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}
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}
This is a reland of 69241a93fb
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 69241a93fb.
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}
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}
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}
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}
