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[git-webkit] Open pre-populated GitHub page for new tokens#185

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[git-webkit] Open pre-populated GitHub page for new tokens#185
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JonWBedard:eng/git-webkit-Open-pre-populated-GitHub-page-for-new-tokens

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@JonWBedard JonWBedard commented Mar 3, 2022

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43e98ff

[git-webkit] Open pre-populated GitHub page for new tokens
https://bugs.webkit.org/show_bug.cgi?id=237436
<rdar://problem/89763053 >

Reviewed by Ryan Haddad.

GitHub gives us a way to pre-populate its token generation web page. We should
open this page with appropriate arguments during setup.

* Tools/Scripts/libraries/webkitbugspy/setup.py: Bump version.
* Tools/Scripts/libraries/webkitbugspy/webkitbugspy/__init__.py: Ditto.
* Tools/Scripts/libraries/webkitbugspy/webkitbugspy/github.py:
(Tracker.credentials): Make 'prompt' into a function, allowing us to open a url
and change the prompt based on the success of opening the url.
* Tools/Scripts/libraries/webkitcorepy/setup.py: Bump version.
* Tools/Scripts/libraries/webkitcorepy/webkitcorepy/__init__.py: Ditto.
* Tools/Scripts/libraries/webkitcorepy/webkitcorepy/credentials.py:
(credentials): Allow 'prompt' to be callable.


Canonical link: https://commits.webkit.org/248033@main
git-svn-id: https://svn.webkit.org/repository/webkit/trunk@290793 268f45cc-cd09-0410-ab3c-d52691b4dbfc

@JonWBedard JonWBedard self-assigned this Mar 3, 2022
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JonWBedard force-pushed the eng/git-webkit-Open-pre-populated-GitHub-page-for-new-tokens branch 2 times, most recently from e408b0f to a31260f Compare March 3, 2022 19:02

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Do we want to provide an option to users to skip this, and let them know that they will need to provide it next time they run a command which needs these credentials?

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The trouble in git-webkit setup is that failing to provide credentials here will fail setup, because we can't create a user-owned fork.

@aj062 aj062 Mar 3, 2022

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ok. Maybe update the message being printed above to indicate that fork is being created as well. currently it only talks about pull requests.

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We do, just not here because the webkitbugspy code is a bit more generic:

if repository.owner == username or args.defaults or Terminal.choose(

@JonWBedard
JonWBedard force-pushed the eng/git-webkit-Open-pre-populated-GitHub-page-for-new-tokens branch from a31260f to 5355b36 Compare March 3, 2022 21:32
@JonWBedard JonWBedard closed this Mar 3, 2022
@JonWBedard
JonWBedard force-pushed the eng/git-webkit-Open-pre-populated-GitHub-page-for-new-tokens branch from 5355b36 to 43e98ff Compare March 3, 2022 21:51
@JonWBedard
JonWBedard merged commit 43e98ff into WebKit:main Mar 3, 2022
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JonWBedard deleted the eng/git-webkit-Open-pre-populated-GitHub-page-for-new-tokens branch March 3, 2022 21:52
@JonWBedard JonWBedard added Tools / Tests Tools in the Tools directory, build issues, test infrastructure, and bugs in test cases WebKit Nightly Build labels Mar 29, 2022
webkit-commit-queue pushed a commit to xingri/WebKit that referenced this pull request Sep 6, 2023
https://bugs.webkit.org/show_bug.cgi?id=258794

Reviewed by Youenn Fablet.

The current WebRTC HEVC is using generic packetization instead of RFC 7789 Packetization.
So this is about implementing the RFC 7798 Packetization.

Fix HEVC depacketizer issues. (WebKit#185)
Enalbing low latency mode for RTC (WebKit#169)
Enable HEVC support. (WebKit#165)
Fix out-of-bounds write in H265VpsSpsPpsTracker (WebKit#163)
Apply fix bitstream logic to RtpVideoStreamReceiver2 (WebKit#142)
Add missing CODEC_H265 switch case (WebKit#136)
Add HEVC support for iOS/Android (WebKit#68)
H265 packetization_mode setting fix (WebKit#53)
Add H.265 QP parsing logic (WebKit#47)

This patch is extracted from following Open WebRTC Toolkit code changes:
<open-webrtc-toolkit/owt-deps-webrtc#185>
<open-webrtc-toolkit/owt-deps-webrtc#169>
<open-webrtc-toolkit/owt-deps-webrtc#165>
<open-webrtc-toolkit/owt-deps-webrtc#163>
<open-webrtc-toolkit/owt-deps-webrtc#142>
<open-webrtc-toolkit/owt-deps-webrtc#136>
<open-webrtc-toolkit/owt-deps-webrtc#68>
<open-webrtc-toolkit/owt-deps-webrtc#53>
<open-webrtc-toolkit/owt-deps-webrtc#47>

co-authoured by:
taste1981 <[email protected]>
jianjunz <[email protected]>
Cyril Lashkevich <[email protected]>
Piasy <[email protected]>
ShiJinCheng <[email protected]>
Andreas Unterhuber <[email protected]>
dong-heun <[email protected]>

* Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_codec_type.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h:
(webrtc::VideoCodecH265::operator!= const):
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_common.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_common.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_sps_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_sps_parser.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_vps_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_vps_parser.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/media/base/media_constants.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/media/base/media_constants.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internal_decoder_factory.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h265.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h265.h:
(webrtc::RtpPacketizerH265::PacketUnit::PacketUnit):
(webrtc::VideoRtpDepacketizerH265::~VideoRtpDepacketizerH265): Deleted.
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h265/include/h265_globals.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h265_vps_sps_pps_tracker.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h265_vps_sps_pps_tracker.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/min_video_bitrate_experiment.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/test/video_codec_settings.h:
(webrtc::test::CodecSettings):
* Source/ThirdParty/libwebrtc/Source/webrtc/video/config/video_encoder_config.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/config/video_encoder_config.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_overshoot_detector.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver2.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver2.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc:
* Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj:

Canonical link: https://commits.webkit.org/267677@main
eocanha pushed a commit to eocanha/WebKit that referenced this pull request Jun 19, 2024
https://bugs.webkit.org/show_bug.cgi?id=258794

Reviewed by Youenn Fablet.

The current WebRTC HEVC is using generic packetization instead of RFC 7789 Packetization.
So this is about implementing the RFC 7798 Packetization.

Fix HEVC depacketizer issues. (WebKit#185)
Enalbing low latency mode for RTC (WebKit#169)
Enable HEVC support. (WebKit#165)
Fix out-of-bounds write in H265VpsSpsPpsTracker (WebKit#163)
Apply fix bitstream logic to RtpVideoStreamReceiver2 (WebKit#142)
Add missing CODEC_H265 switch case (WebKit#136)
Add HEVC support for iOS/Android (WebKit#68)
H265 packetization_mode setting fix (WebKit#53)
Add H.265 QP parsing logic (WebKit#47)

This patch is extracted from following Open WebRTC Toolkit code changes:
<open-webrtc-toolkit/owt-deps-webrtc#185>
<open-webrtc-toolkit/owt-deps-webrtc#169>
<open-webrtc-toolkit/owt-deps-webrtc#165>
<open-webrtc-toolkit/owt-deps-webrtc#163>
<open-webrtc-toolkit/owt-deps-webrtc#142>
<open-webrtc-toolkit/owt-deps-webrtc#136>
<open-webrtc-toolkit/owt-deps-webrtc#68>
<open-webrtc-toolkit/owt-deps-webrtc#53>
<open-webrtc-toolkit/owt-deps-webrtc#47>

co-authoured by:
taste1981 <[email protected]>
jianjunz <[email protected]>
Cyril Lashkevich <[email protected]>
Piasy <[email protected]>
ShiJinCheng <[email protected]>
Andreas Unterhuber <[email protected]>
dong-heun <[email protected]>

* Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_codec_type.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h:
(webrtc::VideoCodecH265::operator!= const):
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_common.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_common.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_pps_parser.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_sps_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_sps_parser.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_vps_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h265/h265_vps_parser.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/media/base/media_constants.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/media/base/media_constants.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internal_decoder_factory.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/create_video_rtp_depacketizer.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h265.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h265.h:
(webrtc::RtpPacketizerH265::PacketUnit::PacketUnit):
(webrtc::VideoRtpDepacketizerH265::~VideoRtpDepacketizerH265): Deleted.
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h265/include/h265_globals.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h265_vps_sps_pps_tracker.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h265_vps_sps_pps_tracker.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/min_video_bitrate_experiment.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/components/video_codec/nalu_rewriter.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/test/scenario/video_stream.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/test/video_codec_settings.h:
(webrtc::test::CodecSettings):
* Source/ThirdParty/libwebrtc/Source/webrtc/video/config/video_encoder_config.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/config/video_encoder_config.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_overshoot_detector.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver2.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver2.h:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc:
* Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc:
* Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj:

Canonical link: https://commits.webkit.org/267677@main
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