The performance cost is not trivial but according to my profiling,
it is acceptable.
Bug: b/139745386
Change-Id: I0e63221ccf22e9f6fb32c630ff63a279e765994a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150539
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28973}
Bug: webrtc:10419
Change-Id: I18528bf2526e933568bf052de76a434f012161da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148320
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28838}
The GetImplementations function is similar to the GetSupportedFormats function, but instead of providing one SdpVideoFormat per codec it provides one per codec implementation. These SdpVideoFormats can then be tagged so that a certain implementation can be instantiated when CreateVideoEncoder is called.
Bug: webrtc:10795
Change-Id: I79f2380aa03d75d5f9f36138625abf3543c2339d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145215
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28553}
Encountering GL_OUT_OF_MEMORY is relatively common and we should give
clients a chance to deal with it in a non-fatal way.
Bug: webrtc:8154
Change-Id: Ifa9ca74392f21083692b02a5144dc5632a88d34d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144561
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28495}
When provided, these thresholds will be used instead of WebRTC default
limits specified in DropDueToSize() and GetMaxDefaultVideoBitrateKbps().
Bug: none
Change-Id: Ida45ea832041963b8b8475d69114b5c60a172fb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142170
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28390}
Preparation for adding a release() method on java's EncodedImage, and
call that from C++.
Bug: webrtc:9378
Change-Id: I301f64b16684c535f45a3fc9cd9ae1543df59d92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141861
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28268}
This is a reland of 11dfff0878
Now that I am sure that WebRTC code is not calling the obsolete
versions, I will just remove the NOT_REACHED and call the
new version from the old ones, so as not to trip up downstream
projects.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,sprang@webrtc.org
Bug: webrtc:10720
Change-Id: I46c69e45c190805c07f7e51acbe277d7eebd1600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141412
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28236}
This reverts commit 11dfff0878.
Reason for revert: Downstream import failure.
Original change's description:
> Inform VideoEncoder of negotiated capabilities
>
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
>
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}
TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org
Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().
Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
SurfaceTextureHelper currently crashes if an OES texture is produced
before setTextureSize() has been called. This is annoying if the texture
size is not easily known beforehand. A real world example is MediaPlayer
that provides the video size with an asynchronous call to
setOnVideoSizeChangedListener(), but that might happen after the first
texture is produced on some devices.
This CL waits with delivering frames until the size has been sent,
rather than crashing.
Bug: webrtc:10709
Change-Id: I5d9ce542e0edaafe1153fd5fe7d64dba86d7e33c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140080
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28151}
from a field trial to RTCConfiguration.
The test coverage is also expanded for the underlying feature.
Bug: None
Change-Id: Ic9c1362867e4a956c5453be7a9355083b6a442f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28143}
This CL adds the callback on changes of the ICE connection state
following the standardized transitions
(https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate) to the
Android and the iOS SDKs.
Bug: None
Change-Id: I6133391fa54dd4e09016f29dddb85e4a0e270878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138181
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28127}
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.
Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
This is a reland of 177670afd6
Fixing failing tests.
TBR=magjed@webrtc.org
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
Bug: webrtc:10464
Change-Id: I95fac3967217c20a9fdddb490aea30eca2061ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130362
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27402}
This reverts commit 177670afd6.
Reason for revert: Fails android_instrumentation_test_apk:
https://ci.chromium.org/p/webrtc/builders/ci/Android64%20(M%20Nexus5X)/11553
Original change's description:
> Add bindings for simulcast and RIDs in Android SDK.
>
> This adds the bindings for rid in RtpParameters.Encoding and bindings
> for send_encodings in RtpTransceiverInit to allow creating a transceiver
> with multiple send encodings.
>
> Bug: webrtc:10464
> Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27323}
TBR=magjed@webrtc.org,shampson@webrtc.org,amithi@webrtc.org
Change-Id: Id6c4e2d41c3c2fbfad31baed907cfa73d82ef14a
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10464
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130466
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27354}
This adds the bindings for rid in RtpParameters.Encoding and bindings
for send_encodings in RtpTransceiverInit to allow creating a transceiver
with multiple send encodings.
Bug: webrtc:10464
Change-Id: I4c205dc0f466768c63b7efcb3c68e93277236da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128960
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27323}
Even if neither frame height nor frame width is <=0 we can end up
with <=0 dimensions in renderHeight or renderWidth. With this change,
we perform the check on the latter.
Bug: webrtc:10367
Change-Id: I9672672659ad7d12cf1e7ccab5b5cde583ae7e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129760
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27307}
We may sometimes want to override only input or only output, or
override them with different values. This change allows setting the
overrides separately.
Change-Id: Ib0c44cb7a3cfa834f997fb6cd54f7cad68705f41
Bug: webrtc:10441
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128763
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27236}
This CL adds a way to extract the underlying android.opengl.EGLContext
and javax.microedition.khronos.egl.EGLContext for EglBase14 and
EglBase10 respectively. The reason is that clients can't be expected to
use only WebRTC's OpenGL code and might need to integrate with their
own GL code.
Bug: None
Change-Id: Ie00a564de45a090683542a52005da7e43c586ced
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127888
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27205}
The video decoder thread is the pilot user.
For now this is an Android-only feature, since that's the only
platform we can print stack traces on.
Bug: webrtc:9987
Change-Id: Ie638c619673b5f159d91a32683fd787baf46479a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126222
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27127}
Previously, we have created a Legacy ADM when no ADM is supplied.
With this change we will start creating a Java ADM instead.
The end goal is to make injection mandatory, and never creating ADMs.
This is one step on the way, and will allow us to clean up the Legacy
ADM code.
Bug: webrtc:7452
Change-Id: Ib99adc50346fe6b748f9435d2fc6321a50c3ee4e
Reviewed-on: https://webrtc-review.googlesource.com/c/123887
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26949}
There has been some crashes due to frames having illegal sizes, most
likely 0x0. Probably these frames are created as a workaround for
something.
It would be best to stop 0x0 frames from being created in the first
place, but a reasonable quick fix is to just not draw those frames.
Bug: webrtc:10367
Change-Id: Ib93057c4de7285773c99614b4e7d9bd4b099c4dc
Reviewed-on: https://webrtc-review.googlesource.com/c/124988
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26897}
Added audio format field and set method to Builder. - WebRTCAudioRecord. Added audio format field, added to constructor. Default audio format value AudioFormat.ENCODING_PCM_16BIT. initRecord(), added how to calculate bytesPerFrame, depends on audioFormat.
First commit and contribution, updated AUTHORS file
Bug: None
Change-Id: I16f660d42350ec9ce2e329b239bd7f6324e76dfe
Reviewed-on: https://webrtc-review.googlesource.com/c/122302
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26775}
This CL adds an API for injecting video processing after the WebRTC
CPU and QP scaling step.
Bug: webrtc:10247
Change-Id: I776498e1e9113f50e953ee411bbb31f181863312
Reviewed-on: https://webrtc-review.googlesource.com/c/119953
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26740}
AndroidVideoTrackSource::OnFrameCaptured currently does adaptation
before passing frames on. We want to add video processing between
adaptation and delivering the frame to the rest WebRTC C++. This
CL prepares for that by splitting OnFrameCaptured() into a separate
adaptation step and delivery step.
Bug: webrtc:10247
Change-Id: Iab759bac7f3072d4552ece80d0b81fc3e634c64c
Reviewed-on: https://webrtc-review.googlesource.com/c/119952
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26571}
This CL attempts to do separation of concerns by introducing a simple
class that only handles JNI wrapping of a C++ AndroidVideoTrackSource.
This layer can be easiliy mocked out in Java unit tests.
Bug: webrtc:10247
Change-Id: Idbdbfde6d3e00b64f3f310f76505801fa496580d
Reviewed-on: https://webrtc-review.googlesource.com/c/121562
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26556}
This layer is not needed since the methods are thread safe, and the
classes those method touches (VideoBroadcaster, cricket::VideoAdapter)
are thread safe.
Bug: webrtc:10247
Change-Id: Id4e309de4ac1b9669052aaa60d3bd1ed965aaa29
Reviewed-on: https://webrtc-review.googlesource.com/c/120801
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26543}
This CL hooks up the recently added native stack trace functionality to
the existing PeerConnectionFactory API.
Bug: webrtc:10168
Change-Id: I16189d2988b1359fc53f9a4d0b3d06f34e2a8fd5
Reviewed-on: https://webrtc-review.googlesource.com/c/118600
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26344}
This CL prepares for adding stack trace capability to the native part of
the Android PeerConnectionFactory code. The main blocker this CL removes
is the static printStackTrace() function. We need this function to be
non-static since the C++ counterpart of PCF is non-static. This Cl also
performs various other cleanups in surrounding code.
This CL:
* Removes static thread references from PeerconnectionFactory and turns
them into non-static member variables.
* Adds a non-static alternative to
PeerconnectionFactory.printStackTraces().
* Removes the rtc::Thread::Invoke() calls, and turns them into
asynchronous posts.
* Consolidates the two different Java PCF ctors into one, so that there
is one shared path used by both native API and Java API.
Bug: webrtc:10168
Change-Id: I05dbf5b17069d4a115d9adafc25faa121f23b945
Reviewed-on: https://webrtc-review.googlesource.com/c/115961
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26329}
eglDestroyContext has been observed to deadlock with other GL threads
unless the GL program is detached beforehand.
TBR=sakal
NO_TRY=TRUE
Bug: b/120481228
Change-Id: Ie256e745828997b6fee0d62e681f5ef953aa0fe7
Reviewed-on: https://webrtc-review.googlesource.com/c/114164
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25999}
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.
Original comment from upstream change:
> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.
Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
Stack traces usually get printed when an error occur and we want this
to be included in release versions.
Bug: None
Change-Id: I17fdbc58393f5b4d597b14e95240bdb04473b4ad
Reviewed-on: https://webrtc-review.googlesource.com/c/112133
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25821}
Ability to provide user defined predicate to disable particular
codec in particular circumstances was added. This could help
addressing mysterious crashes on specific Android devices.
Bug: webrtc:10029
Change-Id: I7ad81f4b1351aa68f036c0ee3b6d32fbf0f697ed
Reviewed-on: https://webrtc-review.googlesource.com/c/111781
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25820}
This change removes the ability to set CryptoOptions through the PeerConnection
Factory in both Java and IOS. Native will be removed after the Chromium change
lands. The semantics have been changed such that these options should only be
set on individual PeerConnections and not directly on the Factory itself. This
allows for more flexibility in setting CryptoOptions for PeerConnections which
are created as part of a factory.
Bug: webrtc:10020
Change-Id: I9ef3d431e728927b9ced5de6188cedeb2671254b
Reviewed-on: https://webrtc-review.googlesource.com/c/111560
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25736}
