It can happen that SFU will resend the frame which was before
considered as dropped during stream switching.
Bug: b/197740434
Change-Id: I95a67e6e637f6005a24df15875b50133a6e8eaaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230423
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34865}
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}
With this change, we catch audio unit start errors and pipe them to the
audio session. The audio session notifies its delegate, which can then
take appropriate action based on the error code.
The signal follows the same path as the playout glitch detection.
Bug: webrtc:13119
Change-Id: I8c9f9d2a1e3457447d0ce61ad197f7e1c6392837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230240
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34862}
This updates the VP8 payload diagrams to be compliant with RFC 7741. It
also fixes some minor inconsistencies with PID, previously referred to
as PartID.
Bug: None
Change-Id: I33eb57d96f3d95b01ef5f0afa21a9dc54b41db2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230243
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34859}
This class is accessed by Electron for its desktop capture support,
but it breaks with component builds on Windows because the symbols
aren't exported by the dll.
No behavior change at runtime, only modifies the generated .lib
when building as a shared library (static builds are unchanged).
Bug: None
Change-Id: I5dc606846de990c1bf4d375ddbb1c73dfc512762
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230200
Reviewed-by: Joe Downing <joedow@chromium.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/main@{#34858}
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$5.run(CameraCapturer.java:272)
at android.os.Handler.handleCallback(Handler.java:883)
at android.os.Handler.dispatchMessage(Handler.java:100)
at android.os.Looper.loop(Looper.java:237)
at android.os.HandlerThread.run(HandlerThread.java:67)
Bug: webrtc:13032
Change-Id: I30b6d6da40bc90a94c0c3c79f9dff523182d3da4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227289
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34854}
This patch adds support for manually setting subnets that
should be handled as VPN, i.e be subject to VpnPreference,
in case webrtc fails to auto-detect VPNs.
Bug: webrtc:13097
Change-Id: I42514f0677a35cfe30ad053570fa9c2a5b4a856b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230122
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34852}
Fatal Exception: java.lang.NullPointerException: Attempt to read from field 'int org.webrtc.CameraEnumerationAndroid$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 reverts commit 9d0730942677a520ce7e184d081b4c5a2469fc48.
Reason for revert: Speculative revert due to failing downstream test. If the test recovers, I'll assign the issue to the tests owners.
Original change's description:
> red: generate and parse the red fmtp format
>
> generates a fmtp line like
> a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
> and matches the incoming redundant payload types against the
> send codec one. Offers without an FMTP line will not use RED.
> Redundancy levels of 1 (plus main packet ) to 32 are accepted but
> this is not wired up to the encoder since the O/A semantic of
> RFC 2198 is not clear.
>
> This decreases the chance of a collision with the SATIN codec
> which also runs on 48khz (but so far does not specify a channelCount of 2)
>
> BUG=webrtc:11640
>
> Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34848}
TBR=henrik.lundin@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I5a0816a22a2a213679ab047c61e3b1dda40c4f59
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230140
Reviewed-by: Björn Terelius <terelius@google.com>
Commit-Queue: Björn Terelius <terelius@google.com>
Cr-Commit-Position: refs/heads/main@{#34850}
Also change return type of FinalRefCountedObject::Release() to
RefCountReleaseStatus, for consistency with other refcount classes.
Bug: webrtc:12701
Change-Id: I37c325e78ba7ae3e220b618da02cb243604ca4cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229590
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34849}
generates a fmtp line like
a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype>
and matches the incoming redundant payload types against the
send codec one. Offers without an FMTP line will not use RED.
Redundancy levels of 1 (plus main packet ) to 32 are accepted but
this is not wired up to the encoder since the O/A semantic of
RFC 2198 is not clear.
This decreases the chance of a collision with the SATIN codec
which also runs on 48khz (but so far does not specify a channelCount of 2)
BUG=webrtc:11640
Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34848}
Deletes the helper methods SdpSetObserver and SdpCreateObserver,
replaced with observer classes where used, in peer_scenario_client.cc.
Deletes the class webrtc_sdp_obs_impl::SdpSetObserversInterface, which
indirectly inherits rtc::RefCountInterface twice. Migrates this code
to use rtc::make_ref_counted, and migrates away from deprecated
versions of SetLocalDescription and SetRemoteDescription that use raw
pointers and SetSessionDescriptionObserver.
Bug: webrtc:12701, webrtc:11798
Change-Id: I18ea3fb51f533d7454a6dc75292b1827b1c80ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229981
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34843}
This patch adds a vp preference field to RTCConfig.
DEFAULT, // No VPN preference.
ONLY_USE_VPN, // only use VPN connections.
NEVER_USE_VPN, // never use VPN connections
PREFER_VPN, // use a VPN connection if possible, i.e VPN connections sorts higher than all other connections.
AVOID_VPN, // only use VPN if there is no other connections, i.e VPN connections sorts last.
Bug: webrtc:13097
Change-Id: I3f95bdfa9134e082c7d389f803bd08facfb70262
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229591
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34842}
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}
Instead offer getters for the sync_group and rtp struct. Both are
a part of the config but expose much less of the config, which has
mutable parts.
Bug: none
Change-Id: Icc8007246e9776a5d20f30cda1a2df3fb7252ffc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229980
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34838}
Having ScopedFieldTrials at class scope might introduce some hard
to understand lifetime patterns. Keeping them in scope only for the
Run method simplifies that, reducing the risk of problems.
Bug: b/197053062
Change-Id: I1c1239757387443552a7b5f83f68014ee56e4248
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229920
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34836}
The config struct is big and in order to control access to its state,
some of which isn't always const, we need to limit raw unlocked access
to it from other classes.
Bug: none
Change-Id: I4513c41486e79ef6c5cfd6376122ab338ad94642
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229921
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34835}
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}
This CL adds an RTC_CHECK in both ctor and dtor to ensure field trials
are valid. Even if the check in the ctor is done already in debug mode,
having it done always is fine because ScopedFieldTrials are testonly.
The check in the dtor should catch issues like reverting to another
ScopedFieldTrial which has already been destroyed.
Bug: None
Change-Id: I53a8680c3ff4fd0e2cbb3055af726a9023b45ac7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229861
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34828}
This patch adds small cost to Vpn connections
so that a "raw" connection identical to a vpn connection
will be chosen first.
The feature is gated by a field trial WebRTC-AddNetworkCostToVpn
for safe roll out.
Bug: webrtc:13097
Change-Id: I4ad40fa00780a6d7f89cacf6f85f3db4ecd0988c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229585
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34822}
Don't DCHECK that network send is successful, it may fail, e.g., EPIPE
if remote end has disconnected.
Bug: None
Change-Id: I7ccff072420498b60fe16598110da91b01bfe7cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229384
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34821}
