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}
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}
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}
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}
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}
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}
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}
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}
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 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}
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}
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}
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}
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}
