admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/AUTHORS

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

217 lines
7.3 KiB
Plaintext
Raw Permalink Normal View History

authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
# Names should be added to this file with this pattern:
#
# For individuals:
# Name <email address>
#
# For organizations:
# Organization <fnmatch pattern>
#
# See python fnmatch module documentation for more information.
#
# Please keep the list sorted.
Adding author git-svn-id: http://webrtc.googlecode.com/svn/trunk@869 4adac7df-926f-26a2-2b94-8c16560cd09d
2011-11-02 09:48:06 +00:00
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
# BEGIN individuals section.
Adds missing header to fix compilation error when compiling with use_custom_libcxx set to false. Fixed: webrtc:12584 Change-Id: I8830095f887e7ee8887bc37106da847b60c1e996 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211762 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33557}
2021-03-14 15:37:35 +00:00
Aaron Clauson <aaron@sipsorcery.com>
Add a GN target for unit tests, get them working again and added a test. BUG=webrtc:3417 Review-Url: https://codereview.webrtc.org/2050153003 Cr-Commit-Position: refs/heads/master@{#14959}
2016-11-07 14:53:28 -08:00
Adam Fedor <adam.fedor@gmail.com>
Fixed that RTCCameraPreviewView did not rotate the video on device rotation. BUG=webrtc:6749 Review-Url: https://codereview.webrtc.org/2798993002 Cr-Commit-Position: refs/heads/master@{#17742}
2017-04-18 07:12:05 -07:00
Akshay Shah <meetakshay99@gmail.com>
Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast. Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000 Hiding snprintf definition if building with Visual Studio 2015 Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc. BUG=webrtc:5183 Review URL: https://codereview.webrtc.org/1412653006 Cr-Commit-Position: refs/heads/master@{#11434}
2016-01-30 14:40:44 -08:00
Alexander Brauckmann <a.brauckmann@gmail.com>
Protect APM in webkit builds. Update libwertc AudioRtpSender::SetAudioSend with WEBRTC_WEBKIT_BUILD This only introduces the WEBRTC_WEBKIT BUILD, inspired by WEBRTC_CHROMIUM_BUILD macro. It is only defined by Webkit libwebrtc build system. https://trac.webkit.org/changeset/210977 BUG=webrtc:7039 Review-Url: https://codereview.webrtc.org/2651273003 Cr-Commit-Position: refs/heads/master@{#16432}
2017-02-03 06:37:05 -08:00
Alexandre Gouaillard <agouaillard@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Alex Henrie <alexhenrie24@gmail.com>
So long and thanks for all the code reviews! - Remove myself from OWNERS. - Add myself to AUTHORS (I signed a CLA). - Add minyue to audio_conference_mixer which would otherwise be empty. - Add missing comma in WATCHLISTS. Review URL: https://codereview.webrtc.org/1458763002 Cr-Commit-Position: refs/heads/master@{#10686}
2015-11-18 03:07:39 -08:00
Andrew MacDonald <andrew@webrtc.org>
Fixed crash on iOS13, methods beginGeneratingDeviceOrientationNotifications and endGeneratingDeviceOrientationNotifications. 1. On iOS13 the implementation of methods begin- and endGeneratingDeviceOrientationNotifications changed and now are looks like "not threadsafe" (in specific sence) - they should be called only on the main thread. This fact is not documented. And may be a mistake. 2. By the Apple official documentation methods begin- and endGeneratingDeviceOrientationNotifications should be balanced. (Each begin- method should be balanced with end- method.) By the reason two above facts they consequences merged and produced the "floating" NSInternalInconsistencyException crash. Bug: webrtc:11216 Change-Id: Ibedd5bba7476cc687de3b9b04be49e3cceac1d27 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162205 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Commit-Queue: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30121}
2019-12-17 23:36:27 +03:00
Andrey Efremov <yoklmnprst@ya.ru>
Fix bug with the sps-pps-idr-in-keyframe fmtp parameter. RtpVideoStreamReceiver was forked to RtpVideoStreamReceiver2 recently, so the code that checks for this parameter needs to be present in the forked location, but it wasn't. This also enables RtpVideoStreamReceiver2TestH264.InBandSpsPps test on MSAN, which was another already fixed bug that wasn't ported over to the recently forked RtpVideoStreamReceiver2. See webrtc:11595 for information about the fork. See webrtc:11769 for information about this fmtp parameter. See webrtc:11376 for the original MSAN issue. Bug: webrtc:11957, webrtc:11595, webrtc:11769, webrtc:8423 Change-Id: I3734d077b2883c2f747ad35a0189b83c1915c3ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184524 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32144}
2020-09-18 22:35:59 -07:00
Andrew Johnson <ajohnson@draster.com>
Fix race condition on ObjCNetworkMonitor::network_monitor_ There is a race condition on reading `ObjCNetworkMonitor::_network_monitor` field. The `ObjCNetworkMonitor::OnPathUpdate()` checks its nullability on the org.webrtc.RTCDispatcherNetworkMonitor thread BEFORE the `ObjCNetworkMonitor::Start()` assigns it on the network_monitor thread. In addition, this field is neither atomic nor protected by mutex so its last assigned value is not guaranteed to be visible to another [reading] thread. Bug: webrtc:355238623 Change-Id: I1a05215111cc873b7d4931824e18f281aebfb91f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357880 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42778}
2024-07-26 10:49:03 +03:00
Andy Tamilo <andytamilo@gmail.com>
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
Anil Kumar <an1kumar@gmail.com>
Fix hw decoder rendering delay after frame resize Need to clear frameInfos in case of reinit, as outdated items produce incorrect decode time. This leds to render timestamps 'in future' (VCMTiming::RenderTime) and rendering delays (low fps). Bug: None Change-Id: Iee569ff74fe3e0ff3610877472756cbbd59aba7a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374680 Auto-Submit: Anna Lemehova <anna.lemehova@gmail.com> Reviewed-by: Zoé Lepaul <xalep@webrtc.org> Commit-Queue: Zoé Lepaul <xalep@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43772}
2025-01-17 19:28:53 +02:00
Anna Lemehova <Anna.Lemehova@gmail.com>
Fix RTCMTLNSVideoView undefined symbol error Bug: webrtc:15861 Change-Id: I43ab06b0935e3894b3c14ea44d77d287169013c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341920 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42517}
2024-03-05 15:24:24 +02:00
Anton Barkov <anton.barkov@macpaw.com>
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
Ben Strong <bstrong@gmail.com>
changed src\modules\audio_device\win\audio_device_core_win.cc , and it is working Bug: webrtc:12384 Change-Id: Ie9fddc3fa8016eb6a0bcc4c6757f30c4b087c10a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203821 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33158}
2021-02-04 09:33:19 +01:00
Berthold Herrmann <bertholdherrmann08@googlemail.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
Bob Withers <bwit@pobox.com>
Enable Multithreaded H264 Encoding For OpenH264 Re-enabled multithreaded encoding using OpenH264, as the issue described in crbug.com/583348 no longer applies. Bug: webrtc:14368 Change-Id: I5ae768a6edf3b40d99c13fb4ee4662626c993a66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271820 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37837}
2022-08-18 11:32:45 +00:00
Brett Hebert <brett@quebic.com>
Fix Event Log For Video Receiver Resolves an issue where, in Chrome, WebRTC event logs do not capture outgoing packets for video receivers because no reference to the event log was passed to the video receiver. Bug: webrtc:14338 Change-Id: Ia33ce6f2d69a0e341530648b10a08516dc53abf3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271080 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37746}
2022-08-09 18:25:04 +00:00
Brett Hebert <hebert.brett@pm.me>
AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial. R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16379004 Patch from Bridger Maxwell <bridgeyman@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-30 00:17:47 +00:00
Bridger Maxwell <bridgeyman@gmail.com>
Add missing header to fix build error when using linux system libraries Change-Id: I4fc04563c2cfe36fa2352f72f2ae61d47972f025 Bug: webrtc:11226 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268194 Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Auto-Submit: Bruno Pitrus <brunopitrus@hotmail.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37555}
2022-07-08 19:27:33 +02:00
Bruno Pitrus <brunopitrus@hotmail.com>
Fix _hRecThread,_hPlayThread RTC_DCHECK reverse bug. Bug: webrtc:6779 Change-Id: I030ec010c39ba3755f70b16a64a5163d0857c256 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228721 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34793}
2021-08-14 16:50:08 +08:00
Cheng Qian <chengqian1521@qq.com>
Add Christophe Dumez to AUTHORS. Copied from Chromium's AUTHORS. R=ch.dumez@samsung.com Review URL: https://webrtc-codereview.appspot.com/5559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5341 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-06 19:43:21 +00:00
Christophe Dumez <ch.dumez@samsung.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Chris Tserng <tserng@amazon.com>
Fix WebRtc ninja x86 build using Visual Studio 2015 (set GYP_MSVS_VERSION=2015). Visual Studio 2015 balks at the implicit truncation of values. Easily fixed with an explicit cast. Fixed redefinition of CLOCKS_PER_SEC when using Visual Studio 2015 and the Windows 10 SDK. CLOCKS_PER_SEC is also defined in "<WIN10 SDK DIR>\include\10.0.10240.0\ucrt\time.h" and also has the value of 1000 Hiding snprintf definition if building with Visual Studio 2015 Fixed C4573 compiler complaint in audio_processing_impl_locking_unittest.cc. BUG=webrtc:5183 Review URL: https://codereview.webrtc.org/1412653006 Cr-Commit-Position: refs/heads/master@{#11434}
2016-01-30 14:40:44 -08:00
Cody Barnes <conceptgenesis@gmail.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
Colin Plumb
Fixed issue with missing network interfaces on iOS Fixed issue with network interfaces due to a missing return value in the "nw_path_enumerate_interfaces(...)" block. Exposed in iOS 18, RTCNetworkMonitor::initWithObserver will only enumerate the first interface, instead of all device interfaces Bug: webrtc:359245764 Change-Id: Ifb9f28c33306c0096476a4afb0cdb4d734e87b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359541 Auto-Submit: Corby <corby.hoback@gmail.com> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42818}
2024-08-14 08:40:50 -04:00
Corby Hoback <corby.hoback@gmail.com>
SetStreams API of RtpSender wrapped for iOS and Android Bug: webrtc:10129 Change-Id: I36ea0110de655bbffa2bd18a024abd15a2136838 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155983 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29405}
2019-10-08 13:06:15 +00:00
Cyril Lashkevich <notorca@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
CZ Theng <cz.theng@gmail.com>
Danail Kirov <dkirovbroadsoft@gmail.com>
Update Apple device list Added new apple devices to corresponding enumeration. Added H264 profile level information. Previous update was done as part of: https://webrtc-review.googlesource.com/c/src/+/158744 Device machine names obtained from: https://gist.github.com/adamawolf/3048717 Change-Id: Ibe71ec525679d34494b579f6da851c2b45b0cd86 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202743 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33119}
2021-01-19 15:29:11 -06:00
Dave Cowart <davecowart@gmail.com>
Fixed issue with BGRA RTCCVPixelBuffer scale and crop BGRA RTCCVPixelBuffers were cropped and scaled incorrectly. Libyuv’s `ARGBScale` method is used in RTCCVPixelBuffer to scale and crop the pixel buffer. To crop by `cropX` and `cropY` pixels, pointer arithmetic is used to offset the src pointer of the original pixel buffer bytes. There is a bug in how this offset is calculated. The offset is done by `src += srcStride * _cropY + _cropX`. Libyuv expects that the src pointer will point to the start of a new pixel. However, if _cropX is a not a multiple of 4 (4 bytes for BGRA), the src pointer will point to a byte in the middle of a pixel and thus libyuv will incorrectly treat the data as the start of pixel (incorrectly treating the first byte as red when it is actually green, etc...). To fix this, the src pointer needs to be offset to always point to the start of a new pixel. Before this change: Original Test Gradient image with a cropX of 2: https://i.imgur.com/gSIgwGV.jpg Scaled image (notice the colors are incorrect): https://i.imgur.com/oPxbTEK.jpg After this change: Scaled image (notice the colors are correct): https://i.imgur.com/dqBsmsH.jpg A new unit test which tests scaling with cropX and cropY values has been added. The test fails without this change and now passes with the correct src pointer offsetting. Bug: webrtc:9555 Change-Id: I87cbd7b91bc139d51fb4e11cc50ccb014cfa8051 Reviewed-on: https://webrtc-review.googlesource.com/89220 Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24076}
2018-07-23 12:50:33 -07:00
David Porter <david@porter.me>
Fixed typos, candiate -> candidate Bug: None Change-Id: I2dee549aa79f1eb6bddd58cfc6c9f67eb6ba3663 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256147 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36273}
2022-03-19 01:43:10 -07:00
David Sanders <dsanders11@ucsbalum.com>
Added support for changing the volume of AudioTrack as discussed in BUG=webrtc:6533 This is a short term solution to change the volume of an AudioTrack until applyConstraints for MediaStreamTracks has been implemented. This CL adds 1 new Java method & the relevant JNI file update: AudioTrack.java: public void setVolume(double volume); BUG=webrtc:6533 Review-Url: https://codereview.webrtc.org/2710683009 Cr-Commit-Position: refs/heads/master@{#17682}
2017-04-12 16:58:48 -07:00
Dax Booysen <dax@younow.com>
fix of a compilation error in Visual Studio 2022 due to a warning C4244. Bug: webrtc:353294048 Change-Id: I8c40a9df06464b9ffd7112fd7d070dee2d0e5ca4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360140 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42825}
2024-08-21 12:17:15 +02:00
Denis Genestier <denis.genestier@shadow.tech>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Dennis Angelo <dennis.angelo@gmail.com>
Explicitly set presentation style to full screen, because, otherwise it is showing in page sheet. Bug: webrtc:11786 Change-Id: I66e24127c249d342dc3cc3df20a78dcde0429da3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179040 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31751}
2020-07-11 08:47:45 +05:30
Dharmesh Chauhan <dharmesh.r.chauhan@gmail.com>
Populate jitter stats for video RTP streams Trying to take my first stab at contributing to WebRTC and I chose to populate jitter stats for video RTP streams. Please yell at me if this isn't something I'm not supposed to pick up. Appreciate a review, thanks! Bug: webrtc:12487 Change-Id: Ifda985e9e20b1d87e4a7268f34ef2e45b1cbefa3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208360 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33325}
2021-02-19 17:34:22 -08:00
Di Wu <meetwudi@gmail.com>
Add ',' between elements in RTCStatsReport::ToJson There was no ',' between array-elements the stats-report, yielding invalid JSON. Let's bring back the ",", which was lost during recent refactoring. Bug: webrtc:10173 Change-Id: Ib58025d56c4895c6af33b9777cb2ebdb94a678ea Reviewed-on: https://webrtc-review.googlesource.com/c/116483 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26161}
2019-01-07 11:41:05 +00:00
Dirk-Jan C. Binnema <djcb@djcbsoftware.nl>
Fix iOS8 crash in background mode. Add system version check functionality in UIDevice+RTCDevice category. Check for iOS system version when handle capture session interruption. BUG=webrtc:7201 Review-Url: https://codereview.webrtc.org/2733773003 Cr-Commit-Position: refs/heads/master@{#17079}
2017-03-06 16:42:19 -08:00
Dmitry Lizin <sdkdimon@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Eike Rathke <erathke@redhat.com>
Update AUTHORS file. BUG= Review URL: https://codereview.webrtc.org/1173403008 Cr-Commit-Position: refs/heads/master@{#9625}
2015-07-23 11:40:20 -07:00
Eric Rescorla, RTFM Inc. <ekr@rtfm.com>
Don't round the computed time deltas to nearest ms. Resolving https://bugs.chromium.org/p/webrtc/issues/detail?id=14023 At the moment, in DelayBasedBwe the time deltas are rounded to the nearest millisecond. This change makes sure the numbers are passed as doubles as expected by the TrendlineEstimator. Change-Id: I68882547fb19af0e67e7b5d8de4159083a54d7eb Bug: webrtc:14023 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261320 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36806}
2022-05-06 01:35:28 -07:00
Filip Hlasek <filip@orcamobility.ai>
Added support for changing the volume of RTCAudioSource as discussed in BUG=webrtc:6533 This is a short term solution to change the volume of a RTCAudioTrack (which contains an RTCAudioSource property) until applyConstraints for RTCMediaStreamTracks has been implemented. This CL adds one new Objective-C method to AudioSourceInterface's wrapper: -(void)setVolume:(double)volume BUG=webrtc:6533, webrtc:6805 This is my first CL for Chromium/WebRTC, so please let me know if I did something wrong. Review-Url: https://codereview.webrtc.org/2534843002 Cr-Commit-Position: refs/heads/master@{#16809}
2017-02-23 13:57:00 -08:00
Frederik Riedel, Frogg GmbH <frederik.riedel@frogg.io>
Revert "h264: fix first_packet_in_frame logic for multislice in a single rtp packet" This reverts commit 3753c8190e3f0aca6758a5521e33f8b5d4f09ab4. Reason for revert: Break assembling of hardware encoded h264 P frame on weak network condition. Original change's description: > h264: fix first_packet_in_frame logic for multislice in a single rtp packet > > a frame must be (or should be) first when it contains either SPS (but not just PPS), > is an IDR or is a slice with first_mb_in_slice == 0. > > Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit > into a single RTP packet which can happen with small 320x196 frames > > BUG=webrtc:352379280,webrtc:346608838 > > Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5 > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#42652} Bug: webrtc:368335257 Change-Id: I07725c78be628bff71b79b8b9369677e39f5f5ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363080 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philipp Hancke <phancke@meta.com> Cr-Commit-Position: refs/heads/main@{#43062}
2024-09-20 08:20:30 +00:00
Gao Chun <gaochun.dev@gmail.com>
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 - SDP sctpmap attribute replaced with fmtp attribute - SDP sctp-port attribute is newly added BUG=3592 R=jiayl@webrtc.org, juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:31:56 +00:00
Giji Gangadharan <giji.g@samsung.com>
Added support for an Origin header in STUN messages. For WebRTC there are instances where it may be desirable to provide information to the STUN/TURN server about the website that initiated a peer connection. This modification allows an origin string to be included in the MediaConstraints object provided by the browser, which is then passed as a STUN header in communications with the server. A separate change will be submitted to the Chromium project that uses and is dependent on this change, implementing IETF draft http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02 Originally a patch from skobalt@gmail.com. (https://webrtc-codereview.appspot.com/12839005/edit) R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 00:47:02 +00:00
Graham Yoakum <gyoakum@skobalt.com>
Fixes a linear interpolation bad access Bug: webrtc:353425611 Change-Id: I9c38428d2463d9d76047ad5be84ed57cba9ebb72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367981 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43408}
2024-11-14 15:28:03 +01:00
Guillaume Petit <guillaume.petit@shadow.tech>
Fix do not unregister bluetooth receiver if it was not registered Bug: webrtc:7890 Change-Id: Ib46b4a4407fa030500930ed03a093b26c71f8963 Reviewed-on: https://chromium-review.googlesource.com/550617 Commit-Queue: Henrik Andreasson <henrika@webrtc.org> Reviewed-by: Henrik Andreasson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18892}
2017-06-29 11:25:16 +03:00
Gustavo Garcia <gustavogb@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Hans Knoechel <hans@hans-knoechel.de>
remove deprecated <codecvt> Bug: chromium:358567090 Change-Id: I0480e8962ee38425406ad167f43d666b57418f5b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360142 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Helmut Januschka <helmut@januschka.com> Cr-Commit-Position: refs/heads/main@{#42827}
2024-08-21 22:31:52 +02:00
Helmut Januschka <helmut@januschka.com>
Re-add author accidentally removed in https://codereview.webrtc.org/2534843002. BUG=None Review-Url: https://codereview.webrtc.org/2785453002 Cr-Commit-Position: refs/heads/master@{#17422}
2017-03-28 06:05:00 -07:00
Hugues Ekra <hekra01@gmail.com>
Adding ChannelStatistics Logs Bug: webrtc:363353566 Change-Id: I187432794d173175b83efd4a8899199916306dcd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361127 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tim Na <natim@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42916}
2024-09-02 15:49:43 -04:00
Hyungjoo Na <element11sodium@gmail.com>
Add jakehilton@gmail.com to AUTHORS BUG=3918 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34889004 Cr-Commit-Position: refs/heads/master@{#8172} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8172 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-27 19:15:11 +00:00
Jake Hilton <jakehilton@gmail.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
James H. Brown <jbrown@burgoyne.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Jan Grulich <grulja@gmail.com>
Call callback in IDLE state In current state, if you want to do something with the capturer (eg. switch to next camera again) it fails with an exception that camera switch is already in progress. Change-Id: I908eb590b54fdf3346441097b39f1f2a2eb56ce8 Bug: webrtc:9527 Change-Id: I908eb590b54fdf3346441097b39f1f2a2eb56ce8 Reviewed-on: https://webrtc-review.googlesource.com/88700 Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23995}
2018-07-16 13:50:32 +02:00
Jan Kalab <pitlicek@gmail.com>
Support 4 channel mic in Windows Core Audio BUG=webrtc:7220 Review-Url: https://codereview.webrtc.org/2712743004 Cr-Commit-Position: refs/heads/master@{#16940}
2017-03-01 05:11:22 -08:00
Jens Nielsen <jens.nielsen@berotec.se>
Added `PeerConnectionObserverJni::OnRemoveTrack()` 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}
2021-05-25 10:35:10 +02:00
Jesús Leganés-Combarro <piranna@gmail.com>
Fix a bug in computing audio delay on ios device. Converts seconds to milliseconds by multiplying 1000 instead of dividing 1000. BUG= R=tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1265823003 . Patch from Jiawei Ou <jiawei.ou@gmail.com>. Cr-Commit-Position: refs/heads/master@{#9693}
2015-08-10 13:24:49 -07:00
Jiawei Ou <jiawei.ou@gmail.com>
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
Jie Mao <maojie0924@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Jiwon Kim <jwkim0000@gmail.com>
Fix potential crash during SimulcastEncoderAdapter tear down. On the Android and iOS platforms, occasionally crash when using the SimulcastEncoderAdapter. The Android platform reverted, In function `SimulcastEncoderAdapter::EncoderContext::Release`, After executing `encoder_->RegisterEncodeCompleteCallback(nullptr)` before execute `encoder_->Release()` If the encoder thread is executed here, ``` // out/xxx/xxx/gen/sdk/android/generated_video_jni/VideoEncoderWrapper_jni.h JNI_GENERATOR_EXPORT void Java_org_webrtc_VideoEncoderWrapper_nativeOnEncodedFrame( JNIEnv* env, jclass jcaller, jlong nativeVideoEncoderWrapper, jobject frame) { VideoEncoderWrapper* native = reinterpret_cast<VideoEncoderWrapper*>(nativeVideoEncoderWrapper); CHECK_NATIVE_PTR(env, jcaller, native, "OnEncodedFrame"); return native->OnEncodedFrame(env, base::android::JavaParamRef<jobject>(env, frame)); // HERE } ``` it will cause `native` to nullptr. iOS also. Bug: webrtc:13156 Change-Id: Id5563b3fa2c11606ae7b35de56bbaa6adba59b14 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231780 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34989}
2021-09-14 11:28:17 +08:00
Johnny Wong <hellojinqiang@gmail.com>
Remove unused var in device info bits from video capture module for Linux Bug: None Change-Id: Icea40fe58e7f65cd1eb311c456ce3cdc802f88a8 Reviewed-on: https://webrtc-review.googlesource.com/97421 Commit-Queue: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24700}
2018-09-03 16:48:58 +02:00
Jose Antonio Olivera Ortega <josea.olivera@gmail.com>
fix: h26x packet buffer video artifacts This change resolves an issue that arises when there is a gap in the sequence numbers of packets associated with a single frame. Before this change, the H26x packet buffer could potentially assemble a frame using only a subset of the packets in the buffer if a packet was missing in the middle and a packet with a marker bit arrived. To address this, the change introduces a check before assembling a frame. This ensures that all packets belonging to a single frame are correctly collected by iterating backward until the first packet in the frame is identified. Bug: webrtc:384391181 Change-Id: I4d09a3d6d569624ece204264cb32e5076ed090a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374183 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Jianlin Qiu <jianlin.qiu@intel.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43793}
2025-01-22 18:45:34 +01:00
Kacper Waśniowski <kacp.was@gmail.com>
Expose setCodecPreferences/getCapabilities for iOS Bug: webrtc:15749 Change-Id: I92f5d5dc5d9eb4d0a60c33ed724a0d3e8b4fa1a8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333402 Auto-Submit: Karim Ham <karim@karhm.com> Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41601}
2024-01-04 09:28:49 +04:00
Karim Hammache <karim@karhm.com>
Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits) https://chromium.googlesource.com/chromium/deps/libjpeg_turbo.git/+log/fa0de07678c9..7b4981b6500c $ git log fa0de0767..7b4981b65 --date=short --no-merges --format='%ad %ae %s' 2021-02-23 enm10k Include jpeglibmangler.h 2021-02-02 ehmaldonado Move metadata in OWNERS files to DIR_METADATA files Created with: roll-dep src/third_party/libjpeg_turbo Bug: none Change-Id: I034188911ccf8cee1e864cf156d0dd4c35759992 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208600 Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33329}
2021-02-24 08:20:39 +09:00
Keiichi Enomoto <enm10k@gmail.com>
Fix: rename ms_per_buffer to buffer_duration Buffer duration is in seconds, not milliseconds. No-Try: True Bug: webrtc:11430 Change-Id: Ib03c2002f2dc6c43e01e50d745d709c2644c8b1e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170500 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30798}
2020-03-16 02:00:26 -07:00
Kiran Thind <kiran.thind@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Korniltsev Anatoly <korniltsev.anatoly@gmail.com>
Fix bugs in discarding stun address If mdns obfuscation is enabled, the stun address "MUST NOT be considered redundant" when it is equal to local socket address. Reference: https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-mdns-ice-candidates-03#section-3.1.2.2 This fixes Bug: webrtc:13426 Change-Id: I28e981fd1b8818f6f8294bc3fedf23b8d9146819 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285421 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38774}
2022-11-30 21:29:02 +09:00
Kyutae Lee <gorisanson@gmail.com>
stdc++: remove unneeded absl::optional wrapping std::optional<T>::emplace() without an initializer is broken on clang++ with gnu libstdc++. this workarounds the bug by removing the absl::optional wrapping, which is actually pointless. https://gcc.gnu.org/bugzilla/show_bug.cgi?id=101227 Bug: chromium:41455655 Change-Id: I05354e57cc4cdda3fa6d3cd23f46462b69cc3bee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355900 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Diep Bui <diepbp@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42561}
2024-06-28 14:55:39 +02:00
lauren n. liberda <lauren@selfisekai.rocks>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Lennart Grahl <lennart.grahl@gmail.com>
Update AUTHORS file. BUG= Review URL: https://codereview.webrtc.org/1173403008 Cr-Commit-Position: refs/heads/master@{#9625}
2015-07-23 11:40:20 -07:00
Luke Weber <luke.weber@gmail.com>
Fixed video capturing on Mac. On specific Macbooks (no exact pattern, unfortunately), video from an integrated camera is not captured. Changed AVCaptureVideoDataOutput pixel format configuration as in Chromium which solved the problem. https://chromium.googlesource.com/chromium/src/media/+/master/capture/video/mac/video_capture_device_avfoundation_mac.mm FourCharCode best_fourcc = kCVPixelFormatType_422YpCbCr8; Tested with external cameras as well. Bug: webrtc:8958 Change-Id: Ib99382b38d1914e2963761a33df310024524c9a4 Reviewed-on: https://webrtc-review.googlesource.com/58880 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22709}
2018-02-28 17:57:23 +03:00
Maksim Khobat <maksimkhobat@gmail.com>
Delete wrapper API ConvertToI420 for YUV conversion to I420 Directly use the libyuv API for YUV conversion to I420 Bug: None Change-Id: Iea6e8fa8f7179c800ea850305170002398cb00dc Reviewed-on: https://webrtc-review.googlesource.com/17260 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Alex Glaznev <glaznev@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20681}
2017-11-15 14:35:56 +05:30
Mallikarjuna Rao V <vm.arjun@samsung.com>
Unit tests for SSLAdapter R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17309004 Patch from Manish Jethani <manish.jethani@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 05:56:44 +00:00
Manish Jethani <manish.jethani@gmail.com>
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
Martin Storsjo <martin@martin.st>
This fixes a small memory leak (found using Xcode/Instruments on iOS) in the ObjC bindings of PeerConnection. The generated session description has to be released by the recipient BUG=3985 R=tkchin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28959004 Patch from Matthias Liebig <matthias.gcode@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 22:01:53 +00:00
Matthias Liebig <matthias.gcode@gmail.com>
rename use_x11 to ozone_platform_x11 Ozone is default now in Chromium and non-Ozone/X11 (aka use_x11) is deprecated. During the transition period use_x11 == ozone_platform_x11. Bug: chromium:1096425 Change-Id: Ie3643360ec6607796533054bdedf8e2bb8e7e749 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231650 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35040}
2021-09-20 09:02:01 +03:00
Maksim Sisov <msisov@igalia.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Maxim Pavlov <pavllovmax@gmail.com>
iOS: add type to peer connection local streams BUG= Review-Url: https://codereview.webrtc.org/2249173002 Cr-Commit-Position: refs/heads/master@{#13825}
2016-08-18 14:21:20 -07:00
Maxim Potapov <vopatop.skam@gmail.com>
Add certificate generate/set functionality to bring iOS closer to JS API The JS API supports two operations which have never been implemented in the iOS counterpart: - generate a new certificate - use this certificate when creating a new PeerConnection Both functions are illustrated in the generateCertificate example code: - https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate Currently, on iOS, a new certificate is automatically generated for every PeerConnection with no programmatic way to set a specific certificate. Work sponsored by |pipe| Bug: webrtc:9498 Change-Id: Ic1936c3de8b8bd18aef67c784727b72f90e7157c Reviewed-on: https://webrtc-review.googlesource.com/87303 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24276}
2018-07-05 15:28:24 +02:00
Michael Iedema <michael@kapsulate.com>
Add Opus FEC options to rtp_encode tool Bug: None Change-Id: I7be70951c20069207963b0fa43564c4008eda870 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318220 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40668}
2023-08-29 14:50:49 -04:00
Michael Klingbeil <klingm@amazon.com>
[DesktopCapture]: Allow toggling the visibility of the cursor This is a change needed to implement (cursor: 'never') https://developer.mozilla.org/en-US/docs/Web/API/MediaTrackSettings/cursor constraint. Bug: chromium:1007177 Change-Id: Id7fae62de180d46a3874856978a3fda559aa6477 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282861 Commit-Queue: Alexander Cooper <alcooper@chromium.org> Reviewed-by: Alexander Cooper <alcooper@chromium.org> Cr-Commit-Position: refs/heads/main@{#38770}
2022-11-24 17:32:45 +01:00
Michał Zarach <michalzaq12@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Michel Promonet <michel.promonet.1@gmail.com>
Miguel Paris <mparisdiaz@gmail.com>
Allow GCC 4.9 to compile Chromium In order to implicit cast an lvalue to an rvalue when returning from a function, the return type and type of variable in the return statement previously had to be exactly the same. When this was not the case, std::move was required. For instance, when returning a std::unique_ptr<Derived> variable in a function with a std::unique_ptr<Base> return type, std::move is required. DR 1579 changed this, and allows for implicitly converting to the return type, if the return type has a constructor(T&&), where T is the type of the local variable being returned. DR 1579 was implemented in GCC 5, but not in GCC 4.9 and below. By explicitly qualifying the local variable with std::move, we allow for compiling with GCC 4.9 and incur no performance penalty. The code is still absolutely correct to the word of C++11. BUG=chromium:682965 See also: * https://bugs.gentoo.org/show_bug.cgi?id=600288 * https://stackoverflow.com/questions/22018115/converting-stdunique-ptrderived-to-stdunique-ptrbase#comment33375875_22018521 * http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2014/n3833.html#1579 Review-Url: https://codereview.webrtc.org/2642053003 Cr-Commit-Position: refs/heads/master@{#16175}
2017-01-19 20:20:45 -08:00
Mike Gilbert <floppymaster@gmail.com>
Fixed mistake in DelayBasedBwe. Use audio_inter_arrival_delta_ when separate audio is enabled. Bug: None Change-Id: Ie657126a89d3a9177d2b76e876afa5da8cac9d48 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266490 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Diep Bui <diepbp@webrtc.org> Reviewed-by: Ying Wang <yinwa@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37638}
2022-06-27 18:04:43 +08:00
Mike Wei <Mike.WeiB@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Min Wang <mingewang@gmail.com>
fixes crash caused by race derefing pixelbufferpool ivar while being destroyed and replaced by format change removes cached pixelbufferpool and instead retrieves current pool from the compressionSession each time (as recommended by apple docs) Bug: webrtc:14688 Change-Id: I2244e69e7f32b912021db0905b9d5867d0bf6357 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284240 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38920}
2022-11-19 22:46:00 -08:00
Mike Woodworth <mike@divergentmedia.com>
Allow any unsignalled SSRC changes on default video receive channel. The first unsignalled SSRC creates a default receive channel. Any unsignalled SSRC changes after that replace the default SSRC. Add unit tests for changing unsignalled SSRCs. BUG=webrtc:5208 Review-Url: https://codereview.webrtc.org/2692993009 Cr-Commit-Position: refs/heads/master@{#16682}
2017-02-17 15:46:43 -08:00
Mo Zanaty <mzanaty@cisco.com>
Exposed Adapter types for Android This avoids the need of hard coding the values to use networkIgnoreMask on Android platform. Bug: None Change-Id: Ib5e860913cec2c6d41cfa1b778cb122d0bfe1300 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311780 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40541}
2023-07-13 13:02:14 +05:30
Mohamed Risaldar UT <mohamed.ut@travancoreanalytics.com>
Fix sharing full screen PowerPoint pres on macOS Before this change the full screen application handler was failing to detect PowerPoint going into presentation mode, resulting in the editor window continuing to be shared rather than the intended behavior of sharing the presentation itself. Fix this by always looking for the PowerPoint full screen presentation window, regardless of whether the editor window is still open. In the current version of PowerPoint, the editor stays open during presentation. Bug: chromium:1231437 Change-Id: I1b21e263d25320cc236d127d22d4d64bb52fcbda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269560 Reviewed-by: Mark Foltz <mfoltz@chromium.org> Commit-Queue: Mark Foltz <mfoltz@chromium.org> Cr-Commit-Position: refs/heads/main@{#37632}
2022-07-26 20:00:18 -04:00
Nico Schlumprecht <me@github.nico.onl>
Added main profile to supported H264 codecs This adds the Main 3.1 profile to the list of supported H264 codecs. This unifies the output of WebRTC codecs among macOS/Windows (which both have Main 3.1 codecs) and headless Linux browsers. Bug: None Change-Id: Ife2fe8c1827be9368fabccc5f24ba316671b1b8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236600 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35328}
2021-11-08 03:58:32 -05:00
Niek van der Maas <mail@niekvandermaas.nl>
Ignore v4l2 select() return when stopping capture With some slightly broken webcams, it's possible that the select() returns with a timeout or no event. In that case, the v4l2 thread never returns. To fix this, just check if quit_ is set and exit unconditionally in that case. https://bugzilla.mozilla.org/show_bug.cgi?id=1752326 Bug: None Change-Id: Ic07ce15afd0016ff9f967c2cf64e646c20127457 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251540 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36200}
2022-02-10 14:00:48 -05:00
Olivier Crête <olivier.crete@ocrete.ca>
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
Pali Rohar
Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() R=henrike@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-29 17:11:07 +00:00
Paul Kapustin <pkapustin@gmail.com>
Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet Bug: webrtc:9370 Change-Id: I59606ef6ea9bbf26de844a2fd3f597856271a86a Reviewed-on: https://webrtc-review.googlesource.com/81700 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23555}
2018-06-07 19:20:55 +08:00
Peng Yu <yupeng323@gmail.com>
Fixed counting of index 'send_codec_position' For-loop has a 'continue' statement that skips increment of the index. Added such an increment before 'continue' for the index to keep up with the for-loop. I guess current implementation will bug on codecs sequence like: 'red, unknown, opus' since the subsequent for-loop (the 'red_codec' one) will not be able to find 'opus'. Seems like adding second increment statement is the easiest way to fix it. Bug: None Change-Id: Iab9cc66cf569458af9fd9ba5b938d83186c78c73 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369700 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43488}
2024-12-02 17:06:05 +03:00
Pete Makeev <guuud.morning@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Philipp Hancke <philipp.hancke@googlemail.com>
Piasy Xu <xz4215@gmail.com>
The designated initializer method declaration in the Objective-C headers for RTCICEServer does't match its implementation. R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/6019004 Patch from Rafael Lopez Diez <rafalopezdiez@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5309 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 21:33:27 +00:00
Rafael Lopez Diez <rafalopezdiez@gmail.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
Ralph Giles <giles@ghostscript.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Raman Budny <budnyjj@gmail.com>
Ramprakash Jelari <ennajelari@gmail.com>
Add Riku Voipio to AUTHORS. Contributed this openmax patch: crrev.com/4636f5bb744c0828ac853e1a513e375886fcd424 R=riku.voipio@linaro.org Review URL: https://codereview.webrtc.org/1430973006 . Cr-Commit-Position: refs/heads/master@{#10543}
2015-11-06 10:51:59 -08:00
Riku Voipio <riku.voipio@linaro.org>
Fix to allow small negative jumps due to out of order packets in packet buffer This resolves an issue where when packets appear out of order at the beginning of a stream, packet_buffer.cc might drop the entire packet buffer because it detects a "large negative jump" even though the difference in sequence numbers is very minor and is caused by network congestion / packet re-ordering. Currently, when the issue occurs, this can cause video corruption/artifacts. More details and reproduction is available on the attached webrtc bug report 390329776. Bug: webrtc:390329776 Change-Id: Idb56eb2e066d596d8afd7ec904359baf0cb3feef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374540 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43753}
2025-01-16 19:41:42 +05:30
Rishit Bansal <rishitbansal0@gmail.com>
Always call ConvertToI420 with positive crop_height Source height may be negative, causing libyuv to invert the image. However the height of the destination buffer specified by crop_height should be positive. Remaining calls in common_video_unittests are valid. Bug: webrtc:9447 Change-Id: I6d398909ae80a99d228ccbbd8c1d7ae804e5bf8d Reviewed-on: https://webrtc-review.googlesource.com/c/86540 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25427}
2018-10-28 00:16:33 +00:00
Robert Bares <robert@bares.me>
Video Capture PipeWire: Filter out non-camera nodes This can be helpful in various situations, such as debugging with an unrestricted Pipewire socket or for downstream projects like B2G/Capyloon. Additionally it will help once we move from the camera portal to the more generic device portal. Original patch by Fabrice Desré <fabrice@desre.org> Bug: webrtc:15464 Change-Id: Iae6802f242d68244bca85947cb15ef3eee923ab0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318642 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40706}
2023-09-06 11:57:00 +02:00
Robert Mader <robert.mader@collabora.com>
Pipewire: Use xdg-portal provided file descriptor The documentation for `OpenPipeWireRemote()` says: > Open a file descriptor to the PipeWire remote where the camera nodes > are available. The file descriptor should be used to create a > pw_core object, by using pw_context_connect_fd. In `InitPipeWire()` we already successfully requested the FD, but then went on and used the unrestricted default socket. This does not matter in non-sandboxed environments, as the stream we want to use is available from both FDs. In flatpak sandboxes, however, this requires to give full Pipewire access to the application. Fix this by simply using the right, restricted FD, and while on it, also make sure to not leak it. This change has already landed in downstream in Firefox, see https://phabricator.services.mozilla.com/D122904 https://phabricator.services.mozilla.com/D124508 Bug: webrtc:13152 Change-Id: I3f8995c54c797e1a90a980f231e496a13cbe65b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230803 Reviewed-by: Joe Downing <joedow@chromium.org> Commit-Queue: Joe Downing <joedow@chromium.org> Cr-Commit-Position: refs/heads/main@{#34983}
2021-09-06 19:33:17 +02:00
Robert Mader <robert.mader@posteo.de>
Update AUTHORS file. BUG= Review URL: https://codereview.webrtc.org/1173403008 Cr-Commit-Position: refs/heads/master@{#9625}
2015-07-23 11:40:20 -07:00
Robert Nagy <robert.nagy@gmail.com>
Added support for an Origin header in STUN messages. For WebRTC there are instances where it may be desirable to provide information to the STUN/TURN server about the website that initiated a peer connection. This modification allows an origin string to be included in the MediaConstraints object provided by the browser, which is then passed as a STUN header in communications with the server. A separate change will be submitted to the Chromium project that uses and is dependent on this change, implementing IETF draft http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02 Originally a patch from skobalt@gmail.com. (https://webrtc-codereview.appspot.com/12839005/edit) R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41409004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-10 00:47:02 +00:00
Ryan Yoakum <ryoakum@skobalt.com>
fix: do not use dispatch_queue_global_t directly Referencing this type directly is not allowed when building with the macOS 14.0 SDK. Other usages in webrtc follow this inline pattern too so going with this instead of "auto" which also works. Change-Id: I84a0ba9c78e83843bc73c71c642caca75750f127 Bug: chromium:1454356 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308560 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40313}
2023-06-12 15:07:14 -07:00
Samuel Attard <samuel.r.attard@gmail.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
Sarah Thompson <sarah@telergy.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Satender Saroha <ssaroha@yahoo.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
Saul Kravitz <Saul.Kravitz@celera.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Allow maxptime equal to minptime in the SDP. The DCHECK crashes debug builds running some applications such as Webex. Bug: None Change-Id: I0061286c4c1d04964678a00014896f1fccd4685d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276460 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38644}
2022-09-25 21:50:56 +08:00
Shaofan Qi <vshaqi@gmail.com>
Associate payload_type with rid When a value is set in RtpEncodingParameters::codec, the corresponding payload_type will be set in the SDP a=rid: line. a=rtpmap:96 VP8/90000 ... a=rtpmap:97 VP9/90000 ... a=rid:r0 send pt=96 a=rid:r1 send pt=97 Bug: webrtc:362277533 Change-Id: Ia9688a5fc83c53cf46621d97e87f8dd363a4d7f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361240 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43049}
2024-09-06 21:44:08 +09:00
Shigemasa Watanabe <shigemasa7watanabe@gmail.com>
video: Implement bandwidth based scaler The |slice_qp_detla| reported by the hardware is not credible, which causing the quality scaler cannot work properly,the resolution cannot be adjusted correctly. To fix this issue, this CL implements a bandwidth scaler which is used for adjust resolution, this scaler will be used when QP based quality scaler is not working due to untrusted QP reported by HW AVC encoder. Bug: webrtc:12942 Change-Id: I2fc5f07a5400ec7e5ead2c2c502faee84d7f2a76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228860 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35120}
2021-09-29 17:19:44 +08:00
Shuhai Peng <shuhai.peng@intel.com>
Fix video renderer slowdown by wrong RenderTime This commit fixes the issue of video playback in slow motion caused by VCMTiming being unable to provide the correct rendering time in scenarios of continuous network packet loss WANT_LGTM=mbonadei Bug: webrtc:376183208 Change-Id: I63617068506e536c4b812215ea084eec18e8ee06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367000 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43392}
2024-11-12 10:37:46 +08:00
Shunbo Li <lishunbo@agora.io>
Fix regression caused by default action changed for h264:Nalu:kFiller This commit fixes the issue of discontinuous RTP sequence numbers caused by improper discarding of these nalu types: kFiller/kEndofSequence/kEndOfStream. Bug: webrtc:368335257 Change-Id: Id7a2d34b22ee1c6e1523d8279d9838c57fdeb97f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366501 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43299}
2024-10-24 17:06:48 +08:00
Shunbo Li <lishunbo@shengwang.cn>
Changed OutputToDebug to create a CFString at compile-time, rather than runtime CFStringFromCString is useful, however if we have a static C string, we can use the defined macro CFSTR to create a CFString at runtime instead. Bug: None Change-Id: I54b3f590b3ab07858409af27b817013c98556ded Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Seija K. <doremylover123@gmail.com> Cr-Commit-Position: refs/heads/main@{#39341}
2023-02-10 13:12:11 -05:00
Seija <doremylover123@gmail.com>
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
Silviu Caragea <silviu.cpp@gmail.com>
Fix invalid output buffer usage This patch fixes the internal AudioCoder output buffer setting to be set prior it will be used within callback from ACM BUG=webrtc:7462 Review-Url: https://codereview.webrtc.org/2806933002 Cr-Commit-Position: refs/heads/master@{#17800}
2017-04-20 15:19:10 -07:00
Stefan Gula <steweg@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Stephan Hartmann <stha09@googlemail.com>
Update AUTHORS file. BUG=N/A R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/25509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-11 21:12:59 +00:00
Steve Reid <sreid@sea-to-sky.net>
Fix 'Screen flickering on ScreenCapturerWinDirectx' Bug: webrtc:15718 Change-Id: I230a0a7d196a4a3aea3b3e47cdf4f47c437e7196 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330800 Commit-Queue: Mark Foltz <mfoltz@chromium.org> Reviewed-by: Zijie He <zijiehe@chromium.org> Reviewed-by: Mark Foltz <mfoltz@chromium.org> Cr-Commit-Position: refs/heads/main@{#42177}
2023-12-10 14:27:06 +08:00
Tao chen <meemetao@gmail.com>
Both build options for symbol export can be specified Bug: None Change-Id: I26c421811952ded7382b48cbe970b97f4ed4d9e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234640 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35176}
2021-10-09 22:16:01 +09:00
Takaaki Suzuki <takaakisuzuki.14@gmail.com>
Adding capture device selection capability for video_loopback. It will help to select any capture device to test the utility. In future we can add screen share as capture device. BUG=webrtc:7719 Change-Id: Iddc66188341c0c90e96766dff671ac3863bf3f5d Reviewed-on: https://chromium-review.googlesource.com/517523 Commit-Queue: Peter Boström <pbos@webrtc.org> Reviewed-by: Peter Boström <pbos@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18392}
2017-05-31 19:20:57 +05:30
Tarun Chawla <trnkumarchawla@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Todd Wong <todd.wong.ndq@gmail.com>
Tomas Popela <tomas.popela@gmail.com>
Reland "Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""""" This is a reland of c8496e9814ad2681b372946f143d1acb45475c7e Original change's description: > Reland "Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."""" > > This is a reland of 703a5d76d9ba8e7984509cc7bf70fb4ed84ef6be > > Original change's description: > > Reland "Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5.""" > > > > This is a reland of af51be7869994a299451e22e6382ae641767b26d > > > > Original change's description: > > > Reland "Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5."" > > > > > > This is a reland of a0adf3d4409036d095480e9bfa0fc06990362f84 > > > > > > Original change's description: > > > > Reland "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate extension version 1.5." > > > > > > > > This is a reland of e7153012682ccd3d1eacc18f802cab7820e3bad3 > > > > > > > > Original change's description: > > > > > Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5. > > > > > > > > > > Bug: chromium:396091 > > > > > Change-Id: Ia1b36c771632c536bb8d15322461b479fabc409e > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148768 > > > > > Commit-Queue: Sergey Ulanov <sergeyu@chromium.org> > > > > > Reviewed-by: Sergey Ulanov <sergeyu@chromium.org> > > > > > Cr-Commit-Position: refs/heads/master@{#29083} > > > > > > > > Bug: chromium:396091 > > > > Change-Id: I0d9171ae5f340e0489e4b45ce5d97bc52b0a4904 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156067 > > > > Commit-Queue: Tommi <tommi@webrtc.org> > > > > Reviewed-by: Tommi <tommi@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#29655} > > > > > > Bug: chromium:396091 > > > Change-Id: I47525911095fabc6cee613d03b0d83134b95b084 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158900 > > > Reviewed-by: Tomas Gunnarsson <tommi@chromium.org> > > > Reviewed-by: Tommi <tommi@webrtc.org> > > > Commit-Queue: Tommi <tommi@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30032} > > > > Bug: chromium:396091 > > Change-Id: I03702c8ea935bb5fe1797defda1ba6b279b95217 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165724 > > Reviewed-by: Jamie Walch <jamiewalch@chromium.org> > > Commit-Queue: Jamie Walch <jamiewalch@chromium.org> > > Cr-Commit-Position: refs/heads/master@{#30461} > > TBR=jamiewalch@chromium.org,tommi@webrtc.org > > Bug: chromium:396091 > Change-Id: If9bd5e7b35240acc4dd528397926ba663fe2affc > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168760 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30548} Bug: chromium:396091 Change-Id: I6892d4bb49cdffe655c238c99e981c4927c9e6fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169200 Reviewed-by: Jamie Walch <jamiewalch@chromium.org> Commit-Queue: Jamie Walch <jamiewalch@chromium.org> Cr-Commit-Position: refs/heads/master@{#30622}
2020-02-26 05:42:59 -05:00
Trevor Hayes <trevor.axiom@gmail.com>
Add IPv6 configuration parameters to iOS API Adds |disableIPV6| and |disableIPV6OnWiFi| properties to RTCConfiguration Bug: None Change-Id: Id59fb2002afadd7817f7caeaa62231bf90ecb274 Reviewed-on: https://webrtc-review.googlesource.com/c/109280 Commit-Queue: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Anders Carlsson <andersc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25496}
2018-11-05 12:48:35 +03:00
Uladzislau Susha <landby@gmail.com>
Implement command line flags for peerconnection client example on Windows Adding the flags and functionality for 'autoconnect', 'autocall', 'server', 'port', and 'help' like in the linux example. BUG=3459 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/13609004 Patch from Vicken Simonian <vsimon@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@6573 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-01 16:28:13 +00:00
Vicken Simonian <vsimon@gmail.com>
Fix compilation errors on Fedora 20. peerconnection_jni.cc: syscall() comes from <unistd.h> RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it rtp_payload_registry_unittest.cc: avoid narrowing int to uint32. BUG=2700 R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5019004 Patch from Victor Costan <costan@gmail.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-09 21:34:30 +00:00
Victor Costan <costan@gmail.com>
Fix build errors on GCC w/ libstdc++ 13.2.1 missing cstdint Bug: webrtc:15870 Change-Id: Id91f6d603c777312eda6d3bc9f03c78109737372 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343000 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42150}
2024-03-14 00:54:32 +01:00
Vinzenz Feenstra <evilissimo@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Vladimir Beloborodov <VladimirTechMan@gmail.com>
This CL adds RTCMTLVideoView.h and RTCCameraVideoCapturer.h to WebRTC.h in order to fix a build issue that comes up when using WebRTC.framework from swift code. BUG=webrtc:7488 Review-Url: https://codereview.webrtc.org/2832803002 Cr-Commit-Position: refs/heads/master@{#17909}
2017-04-27 06:26:19 -07:00
Xiaohong Xu <freemine@yeah.net>
Move rendering code in SurfaceViewRenderer to a separate class. The new SurfaceEglRenderer helper class extends EglRenderer and implements rendering on a SurfaceView. Bug: webrtc:8242 Change-Id: Ic532fe487755d3b54c6bd03f239d714e1ecb10ad Reviewed-on: https://webrtc-review.googlesource.com/2940 Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20562}
2017-11-03 07:55:01 +08:00
Xiaolei Yu <dreifachstein@gmail.com>
Fix crash in peerconnection_client example code Use rtc::scoped_refptr to keep object pointers valid Bug: webrtc:13993 Change-Id: Iffb6bf47e97606bfa3c2e9d8814a56cf509abbd6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259940 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36645}
2022-04-24 18:02:35 +08:00
Xinchao Tian <tianxinchao@360.cn>
Make the code conform to the code style. Make the jsep_transport.cc code conform to WebRTC code style. Bug: None Change-Id: I9142c1e83d4006988e206aeb0bbe52dfffafda60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244600 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35631}
2022-01-05 17:48:33 +08:00
Yaowen Guo <albertguo88@gmail.com>
fix: Handle out-of-range device index after GetDevicesInfo When the specified device was not found in GetDevicesInfo, SetPlayoutDevice/SetRecordingDevice will never return a (-1) error. Bug: None Change-Id: I9ac71cf72f7876c1c54ee593f184aa4007dba22f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320500 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40768}
2023-09-17 16:23:31 +08:00
Youfa <vsyfar@gmail.com>
Fixed NPE inside org.webrtc.Camera1Session.create On some devices `android.hardware.Camera.open` returns null instead of raising exception. It causes `NPE` inside `Camera1Session.create` when method `setPreviewTexture` is invoked on local variable `camera`, which is `null`. Bug: webrtc:8658 Change-Id: Ic65b4aef2c0b8b65735a9db02433b536bfe92ddd Reviewed-on: https://webrtc-review.googlesource.com/33620 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21352}
2017-12-19 12:18:41 +03:00
Yura Yaroshevich <yura.yaroshevich@gmail.com>
Use default RTCConfiguration on iOS With "aggressive" preset the default bundlePolicy is set to "maxBundle" when it shoud be "balanced" according to spec. Bug: webrtc:9458 Change-Id: Ifbdd76be3a6d9968574cba857f178d5f859dcb87 Reviewed-on: https://webrtc-review.googlesource.com/88567 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24650}
2018-09-07 16:43:31 +02:00
Yuriy Pavlyshak <yuriy@appear.in>
Use range-based-for instead of std::for_each and std::mem_fun std::mem_fun is deprecated in C++11, and removed in C++17. Using C++17 option for building libwebrtc causes build failure. This is found during upgrading WebKit tree from C++14 to C++17. This patch replaces std::for_each and std::mem_fun with range-based-for. We also merge loops for streams_ into one. Bug: webrtc:9277 Change-Id: I44a7e44ea21fc33ffa9a586ddfea570f97dfacb6 Reviewed-on: https://webrtc-review.googlesource.com/77280 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23285}
2018-05-17 22:01:13 +09:00
Yusuke Suzuki <utatane.tea@gmail.com>
Fix a dead loop in tools_webrtc/libs/generate_licenses.py Bug: webrtc:13975 Signed-off-by: hanpfei <hanpfei@gmail.com> Change-Id: I3943f66b1a80e3f1e4b05f4436dbeff23f3d6285 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259280 Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Christoffer Jansson <jansson@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@google.com> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#36596}
2022-04-20 10:14:45 +08:00
Pengfei Han <hanpfei@gmail.com>
Video: add new metric for VP9/AV1 hw encoding with softwareBRC Record the RMSE of encoding bitrate diff from output stream to target bitrate set by WebRTC when the AV1/VP9 stream is removed. Design doc: https://docs.google.com/document/d/1tE7Fquj1U6CwZ01QxxjoCaJS2o0LFYyIxvstLDx0qB0/edit#heading=h.c11ebhaja9tg Bug: chromium:1358750 Change-Id: Ifa6f5aacfa8f6b0835f4609f102ae3ae1ee2897e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298684 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39907}
2023-04-20 14:28:01 +08:00
Yingying Ma <yingying.ma@intel.com>
fix h264 encoder don't generate template_structure after first keyframe Bug: webrtc:345993676 Change-Id: Ie71c08d9a29b33c5f5d74d3e0779084ead9b5505 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365962 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43265}
2024-10-17 07:27:14 +00:00
Hailin Zhao <lynnworld7@gmail.com>
Reduce redundant memory allocation when capturing a single monitor. When capturing only one display, it is unnecessary to use DoDuplicateAll, which allocates bitmap image memory for a rectangle encompassing all screens and captures all displays. In this case, I switch to DoDuplicateOne. I have added an int parameter to GetNumFrameCaptured and EnsureFrameCaptured to distinguish which display's skip behavior is currently being executed. After the modification, when capturing a single monitor in a multi-monitor environment, only the bitmap image memory of the size of the captured monitor will be allocated, rather than for all monitors. Bug: webrtc:391914255 Change-Id: Iee56796c50282beaf1dbeef47f5937fe7fe94a05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/375181 Reviewed-by: Joe Downing <joedow@chromium.org> Commit-Queue: Alexander Cooper <alcooper@chromium.org> Reviewed-by: Alexander Cooper <alcooper@chromium.org> Cr-Commit-Position: refs/heads/main@{#43822}
2025-01-24 16:47:11 +08:00
Fizz Fang <fangkb555@gmail.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
# END individuals section.
Libjingle in webrtc needs updated AUTHORS, COPYING, LICENSE_THIRD_PARTY AND README. BUG=1935 R=andrew@webrtc.org, fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1805005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4356 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 18:04:56 +00:00
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
# BEGIN organizations section.
red: modify the encoder to send RFC 2198 modifies the RED encoder to send the actual RFC 2198 format described in https://tools.ietf.org/html/rfc2198 Decoding is handled in neteq, see red_payload_splitter.h BUG=webrtc:11640 Change-Id: Ib3005882a3ceee49d2b05c43357f552432a984ac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176371 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31560}
2020-06-11 10:57:58 +02:00
8x8 Inc. <*@8x8.com>
Add *.jitsi.org to AUTHORS bug: None Change-Id: If231097bd02d4038edd6f18402c1b2bbca660ebc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230242 Reviewed-by: Christoffer Jansson <jansson@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Christoffer Jansson <jansson@google.com> Cr-Commit-Position: refs/heads/main@{#35271}
2021-08-26 11:15:49 +02:00
8x8 Inc. <*@jitsi.org>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
8x8 Inc. <*@sip-communicator.org>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
Agora IO <*@agora.io>
ARM Holdings <*@arm.com>
BroadSoft Inc. <*@broadsoft.com>
modules/video_capture: add NV12 support on Linux Add native NV12 support on Linux v4l2 video_capture module. Bug: webrtc:14650 Change-Id: I97e2010be4f15168b218da4855be8b0e985008a5 Signed-off-by: Dimitri John Ledkov <dimitri.ledkov@canonical.com> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282841 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38753}
2022-11-09 12:04:28 +01:00
Canonical Ltd <*@canonical.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
Remove dimension check in SimulcastUtility::ValidSimulcastParameters We found that the legacy assumption for H264 which assumed that simulcast streams would use 2x width ratios in unnecessary as the encoder has since been fixed to handle multiple ratios. H264 encoder still works even if this assumption is invalid Bug: None Change-Id: I9caacf78d26c8215b94858a2d8674ec4cd64e96e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286940 Reviewed-by: Mirta Dvornicic <mirtad@google.com> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39072}
2023-01-10 15:15:10 -08:00
Discord Inc. <*@discordapp.com>
Fix typo in the include path of ooura_fft.h Bug: None Change-Id: Iaac4a80f75dcd81ab0d2665cb20f27f0342cb17d Reviewed-on: https://webrtc-review.googlesource.com/38441 Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21565}
2018-01-09 09:15:37 -08:00
Facebook Inc. <*@fb.com>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
Google Inc. <*@google.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Highfive, Inc. <*@highfive.com>
Expose peer connection's `getStats` by RtpSender/Receiver in Android SDK Currently if you want to obtain the stats for a specific sender/receiver in Android, you need to call peerConnection.getStats() and filter manually the result by sender. pc.getStats(receiver/sender) exists in c++ and ios but was not exposed in Android Bug: webrtc:14547 Change-Id: I9954434880f0f93821fcd2e2de24a875e8d136ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275880 Reviewed-by: Xavier Lepaul‎ <xalep@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38428}
2022-10-07 10:50:46 +02:00
Hopin Ltd. <*@hopin.to>
Use IFA_LOCAL instead of IFA_ADDRESS over IPv4 network on ANDROID IFA_ADDRESS gives DESTINATION address in case of point-to-point connection, which is not able to create ports for candidate gathering. Use IFA_LOCAL to avoid this problem. Bug: webrtc:9189 Change-Id: Ifcb1955b1b4011dc69c93d99b4e223b370dc16eb Reviewed-on: https://webrtc-review.googlesource.com/69620 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23055}
2018-04-27 09:47:07 +09:00
HyperConnect Inc. <*@hpcnt.com>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
Intel Corporation <*@intel.com>
GCC: Avoid symbol clash in RenderBuffer GCC fails to resolve getter for RenderBuffer::Block() because its return type has the same name with the getter method. Rename getter method with the prefix "Get" as guided in the https://chromium.googlesource.com/chromium/src/+/main/styleguide/c++/blink-c++.md#Precede-setters-with-the-word-Set_use-bare-words-for-getters "If a getter’s name collides with a type name, prefix it with “Get”." Bug: chromium:819294, webrtc:14089 Change-Id: Ieaa3af27415eb8c39806aa8480897b47fd07baa8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263420 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36982}
2022-05-24 10:49:50 +03:00
LG Electronics, Inc. <*@lge.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Life On Air Inc. <*@lifeonair.com>
Expose setCodecPreferences/getCapabilities for android Bug: webrtc:15177 Change-Id: If61ef9a87bc4f68d73cef6e681461682ca48f034 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304880 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40071}
2023-05-15 11:33:38 +00:00
LiveKit <*@livekit.io>
Fix bug where RTCTransportStats.dtlsCipher was missing when using OpenSSL A bug in the id being searched for inside OpenSSLStreamAdapter::SslCipherSuiteToName prevented the lookup from ever succeeding. This resulted in this stat being unavailable when calling PeerConnection::GetStats(). To fix the problem, look for (0x03000000L | cipher_suite) which matches what the BoringSSL codepath is doing. Bug: webrtc:14596 Change-Id: Ic36d77dbc4c2378fbde1e2f21a9f5bd735b36741 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280100 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38460}
2022-10-21 12:08:22 -07:00
Meta Platforms, Inc. <*@meta.com>
Adding Microsoft Corporation (*@microsoft.com) to WebRTC AUTHORS Bug: None Change-Id: I7c425b7ca48580d87757db7a70db30fcbe259adb Reviewed-on: https://webrtc-review.googlesource.com/c/110360 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25642}
2018-11-08 09:53:36 -08:00
Microsoft Corporation <*@microsoft.com>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
MIPS Technologies <*@mips.com>
Mozilla Foundation <*@mozilla.com>
Fix "control reaches end of non-void function" warnings "warning: control reaches end of non-void function [-Wreturn-type]" Reported by gcc (8.3) In all the reported cases, the end of function is never actually reached. Add RTC_CHECK(false) to ensure the compiler is aware that this path is a dead-end. Bug: webrtc:12008 Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com> Cr-Commit-Position: refs/heads/master@{#32503}
2020-10-22 10:08:50 +02:00
Netgem S.A. <*@netgem.com>
Fix for payload type id collision This collision can occur when we have asymetrical send and receive codecs. This is the case in the current code base with the VP9 codec familly but is not visible untill more codecs are added. Added Nutanix Inc. to AUTHORS. Bug: chromium:1291956 Change-Id: I09d3f76161d984d2a3edf721639753bffd4947b9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250034 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35944}
2022-02-03 04:26:22 -08:00
Nutanix Inc. <*@nutanix.com>
Import proto_library.gni when rtc_enable_protobuf is true Import proto_library.gni when rtc_enable_protobuf is true instead of when build_with_mozilla is false. Makes it maybe easier to reason about the intention (e.g. intention is to not compile any protobuf in, hence flag rtc_enable_protobuf) The build file could not work if build_with_mozilla = true but rtc_enable_protobuf = true. Bug: webrtc:10338 Change-Id: I26e5983bd1519aa46c308b11796d518de5ef7597 Reviewed-on: https://webrtc-review.googlesource.com/c/123763 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26868}
2019-02-26 11:46:17 +02:00
NVIDIA Corporation <*@nvidia.com>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
Opera Software ASA <*@opera.com>
Normalize main(..) routines for WinUWP In order to support WinUWP platform, all main(..) routines must be normalized to the formal int main(int argc, char* argv[]) form. A platform wrapper main is auto-created linking against the default main(...). This can only work if the linkage is exactly matching the proper formal definition and not a loosely defined main(...) alternative. Bug: webrtc:8608 Change-Id: I606663aaea7df1792c7c5636279617b8926fa5cc Reviewed-on: https://webrtc-review.googlesource.com/28721 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21229}
2017-12-03 16:45:56 -05:00
Optical Tone Ltd <*@opticaltone.com>
pipewire: handle deleting the capturer while a D-Bus call is in progress If a D-Bus call is in progress when a BaseCapturerPipeWire is deleted, then the user_data is invalid when the callback function is called. This results in memory corruption. To fix this, use a GCancellable. If it is canceled, the callback will be called with a corresponding error. Detect this error and abort before accessing the user_data. Note: The first argument is the 'source_object'. For g_dbus_proxy_call() this is the proxy object not the connection. This was not a problem before, because it was not used. Bug: None Change-Id: I8d5e3fb5c49fcc9afd61cdb8e8249f78b9434faf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149817 Reviewed-by: Jamie Walch <jamiewalch@chromium.org> Commit-Queue: Jamie Walch <jamiewalch@chromium.org> Cr-Commit-Position: refs/heads/master@{#29326}
2019-08-21 13:05:18 +02:00
Pengutronix e.K. <*@pengutronix.de>
Enable Multithreaded H264 Encoding For OpenH264 Re-enabled multithreaded encoding using OpenH264, as the issue described in crbug.com/583348 no longer applies. Bug: webrtc:14368 Change-Id: I5ae768a6edf3b40d99c13fb4ee4662626c993a66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271820 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37837}
2022-08-18 11:32:45 +00:00
Quebic Inc. <*@quebic.com>
Switch SSE2 architecture check to x86 platforms only This allows builds on non-x86 architectures such as ppc64el. Bug: webrtc:14057 Signed-off-by: Timothy Pearson <tpearson@raptorcs.com> Change-Id: Ie2c1023d2c1d041ba1d140f06af432ed9e9f7432 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262002 Commit-Queue: Alexander Cooper <alcooper@chromium.org> Reviewed-by: Mark Foltz <mfoltz@chromium.org> Cr-Commit-Position: refs/heads/main@{#36856}
2022-05-11 02:26:25 +00:00
Raptor Computing Systems, LLC <*@raptorcs.com>
Implement handling for MS PowerPoint Presentation Mode. * Introduce a FullScreenWindowDetector to manage routines for updating the list of sources being application agnostic, inspired by FullScreenChromeWindowDetector. * Introduce a FullScreenApplicationHandler to make a decision about changing window to share in application specific way, inspired by FullScreenChromeWindowDetector. * Remove FullScreenChromeWindowDetector as redundant. * Add FullScreenApplicationHandler for MS PowerPoint and Apple Keynote on MacOS. * Add FullScreenApplicationHandler for MS PowerPoint on Windows. Bug: webrtc:3852 Change-Id: I06507d929308e85b882b2f8210a025afef7f26a9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156020 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Jamie Walch <jamiewalch@chromium.org> Reviewed-by: Justin Uberti <juberti@webrtc.org> Reviewed-by: Wez <wez@google.com> Cr-Commit-Position: refs/heads/master@{#29993}
2019-10-07 11:57:01 -07:00
RingCentral, Inc. <*@ringcentral.com>
Disable remote ICE candidate DNS lookups when the IceTransportPolicy is Relay or None Bug: webrtc:11597 Change-Id: Id3884a2b5f0fc35880c7401c43ca25fee8346519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175960 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31389}
2020-05-27 09:51:05 -07:00
Signal Messenger, LLC <*@signal.org>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
Sinch AB <*@sinch.com>
struktur AG <*@struktur.de>
Telenor Digital AS <*@telenor.com>
Temasys Communications <*@temasys.io>
The Chromium Authors <*@chromium.org>
The WebRTC Authors <*@webrtc.org>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Threema GmbH <*@threema.ch>
Tuple, LLC <*@tuple.app>
Update Android camera switch API to allow specifying a name The current camera switch API sequentially cycles through each camera name for each method invocation. This policy provides reasonable behavior for devices with 2 or 3 cameras, but presents challenges with devices that contain several cameras. For example in a scenario where the current camera is oriented on the same side as the next camera name, a developer would need to call switchCamera multiple times to capture from a camera oriented on a different side of the device. This commit allows a developer to specify a camera name when switching cameras. This flexibility allows developers to have more control over which device they switch to in cases where a device contains several cameras. Bug: webrtc:11261 Change-Id: I93d46d70b2c7cf735a411a4ef4f33e926bf3a5ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165040 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30199}
2020-01-09 04:26:04 +00:00
Twilio, Inc. <*@twilio.com>
Reset VTCompressionSession when underlying CVPixelBufferPoolRef isn't valid Change-Id: If9bf4a5d0db50de36f0d14f08ec83e85dd1c69b8 Bug: webrtc:347647405 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354705 Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#42503}
2024-06-14 10:27:33 -05:00
Twitch Interactive, Inc. <*@justin.tv>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
Vewd Software AS <*@vewd.com>
Videona Socialmedia <*@videona.com>
Expose a link-local network interfaces enumeration option The bug 8432 is caused by trying to connect through a "link-local" interface (IP address 169.254.0.x/16), which is listed among the iPhone network interfaces. The bug is not happening if the link-local network interfaces are skipped in the ICE candidate gethering process. To control this behaviour an option - disable_link_local_networks - is added inside the RTCConfiguration. It is used to set the new BasicPortAllocatorSession flag - PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS. The port allocator flag is added if the configuration option is set. IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate) are refactored to work on both IPv4 and IPv6. Unit test IPIsLinkLocal. Bonus: fix a bug in IPIsLinkLocalV6: take into account just 10 network mask bits instead of 16. Bug: webrtc:8432 Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b Reviewed-on: https://webrtc-review.googlesource.com/36380 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21922}
2018-01-25 10:30:22 +01:00
Videxio AS <*@videxio.com>
Frame marking RTP header extension (PART 1: implement extension) Bug: webrtc:7765 Change-Id: I23896d121afd6be4bce5ff4deaf736149efebcdb Reviewed-on: https://webrtc-review.googlesource.com/85200 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24695}
2018-09-11 16:50:49 -04:00
Vidyo, Inc. <*@vidyo.com>
PRESUBMIT: Add authorized-authors check + AUTHORS e-mails. This check will throw a PRESUBMIT error if the author or organization is not present in the AUTHORS file. E-mail wildcard entries were also added to the organizations in the AUTHORS file. BUG=webrtc:6852 NOTRY=True Review-Url: https://codereview.webrtc.org/2564613002 Cr-Commit-Position: refs/heads/master@{#15591}
2016-12-14 00:16:21 -08:00
Vonage Holdings Corp. <*@vonage.com>
loongarch: Add loongarch architecture support. Copied from chromium's build/build_config.h No-Try: True Bug: webrtc:14433 Change-Id: I5ab4dbb49bd74aa8f6f6054776d9f1f3d606d914 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275288 Owners-Override: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Andrey Logvin <landrey@google.com> Cr-Commit-Position: refs/heads/main@{#38067}
2022-09-13 11:27:34 +08:00
Wang Qing <wangqing-hf@loongson.cn>
Changed OLA window for neteq. Old code didnt work well with 48khz fixing white spaces updated authors file Changed OLA window to use Q14 as Q5 dosnt work with 48khz. 1 ms @ 48 khz is > 2^5 BUG=webrtc:1361 Review-Url: https://codereview.webrtc.org/2763273003 Cr-Commit-Position: refs/heads/master@{#17611}
2017-04-10 02:22:46 -07:00
Wire Swiss GmbH <*@wire.com>
authors: add comments about individual / org sections following https://chromium.googlesource.com/chromium/src/+/refs/heads/master/AUTHORS style-wise and sorting the lists NOTRY=true BUG=None Change-Id: I2a2bbf959e4470827f25cf383750f53bb868b2de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183543 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> Cr-Commit-Position: refs/heads/master@{#32054}
2020-09-07 19:52:31 +02:00
&yet LLC <*@andyet.com>
# END organizations section.
Reference in New Issue Copy Permalink