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}
Users of webrtc generally should be able to choose own task queue implementation.
Poison avoids accidental dependency of a low level component on the default implementation
Android and ios apis are still de-facto forced to use the default implementation.
Bug: webrtc:10284
Change-Id: I67ecf2317f43ee32b0c9e8a6e69f1e0987cf1914
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144786
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28524}
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}
Using relative paths for JNI includes is causing build failures in chromium.
WebRTC already uses full include paths for generated JNI headers, so this CL
just removes the "jni_package" parameter from WebRTC generate_jni() targets
and removes the "jni/" portion of includes. The "jni_package" variable will be
removed from the generate_jni() template shortly.
To fix includes:
find . -name *.cc -exec sed -i -E 's@(#include.+generated.+jni)/jni/(.+_jni.h)@\1/\2@' {} \;
See https://groups.google.com/a/chromium.org/forum/?#!topic/java/MEovGrAwbqI
for discussion on naming scheme.
No-Try: True
TBR: kwiberg@webrtc.org
Bug: chromium:964169
Change-Id: I758c1b41bf6f5005587e55b82f14065fe251baad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143521
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28380}
In https://chromium-review.googlesource.com/1650265 attributes like minSdkVersion were moved from AndroidManifest.xml to GN files. For WebRTC there were a few problems with that.
* We don't want to suppress UsesMinSdkAttributes lint but now there are these "invalid" manifest files that we can't exclude or discern. So disable this lint error.
https://chromium-review.googlesource.com/c/chromium/src/+/1650265/14/build/android/AndroidManifest.xml
* We should specify the versions in GN files, so I did that here (by exactly copying the versions that are already in the targets' corresponding XML files), but we never want to get rid of them in the XML files. For now this information will just be duplicated (without any synchronicity check!) so there should be followup to this.
Change log: 6ae0f0cd4c..bf62d746a4
Full diff: 6ae0f0cd4c..bf62d746a4
Changed dependencies
* src/base: 9e5e9332df..e5a1d1f652
* src/build: 5a031748ec..2ef566e990
* src/buildtools: 6ae683be2f..6f3775ad6e
* src/buildtools/linux64: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/mac: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/buildtools/win: git_revision:8c7f49102234f4f4b9349dcb258554675475e596..git_revision:81ee1967d3fcbc829bac1c005c3da59739c88df9
* src/ios: 2f5c817266..7f1a97d593
* src/testing: 1d4247de57..b1b36ff0d4
* src/third_party: 6f7cbf7c46..42e96c4074
* src/third_party/android_sdk/public: ki7EDQRAiZAUYlnTWR1XmI6cJTk65fJ-DNZUU1zrtS8C..xhyuoquVvBTcJelgRjMKZeoBVSQRjB7pLVJPt5C9saIC
* src/third_party/android_sdk/public: iIwhhDox5E-mHgwUhCz8JACWQCpUjdqt5KTY9VLugKQC..ppQ4TnqDvBHQ3lXx5KPq97egzF5X2FFyOrVHkGmiTMQC
* src/third_party/android_sdk/public: 4Y2Cb2LGzoc-qt-oIUIlhySotJaKeE3ELFedSVe6Uk8C..MSnxgXN7IurL-MQs1RrTkSFSb8Xd1UtZjLArI8Ty1FgC
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ed9fcf3f70..9e5dbd8b46
* src/tools: f58f33bca1..a9a4b8fc7b
DEPS diff: 6ae0f0cd4c..bf62d746a4/DEPS
No update to Clang.
Bug: chromium:891996
Change-Id: I773d6fa90e8083d934c84eecc1cb9d7d4496eca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142235
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28311}
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 11dfff0878c949f2e19d95a0ddc209cdad94b3b4
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 change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.
Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
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}
Since many tests rely on rtc::Thread::Current(), add an
explicit rtc::AutoThread in the main() function used by tests.
Bug: webrtc:9714
Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28000}
This is a reland of e779847fb6499ac2dc4757de8c625ac377e9d0d4
Original change's description:
> Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
>
> Also add explicit includes of rtc_base/string_utils.h in files depending on it.
>
> Bug: webrtc:6424
> Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27903}
Tbr: kwiberg@webrtc.org
Bug: webrtc:6424
Change-Id: Ic08d5d7fbc25ff89e4182d7c9cb3b0e8e356339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135946
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27957}
