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Fixing memory leak for Westeros backend #47

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merged 1 commit into from Apr 26, 2016

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vertexodessa
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@philn philn merged commit 23cda9e into WebPlatformForEmbedded:comcast-master Apr 26, 2016
@philn
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philn commented Apr 26, 2016

Next time please submit changes for the master branch.

@igor-borovkov
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Hi Philippe, we use comcast-master for our daily QA builds, could you and other Metrological devs not push changes to this branch please. @kraj

@philn
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philn commented Apr 26, 2016

Then no need for pull-requests, just push directly to that branch?

@igor-borovkov
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hi,
traditionally all our builds used our own hosted git repos. for the purpose of simplified upstreaming Khem wants to use github, and for even faster/simpler upstreaming Khem wanted to use your repo. We can't just pull master branch into our builds, so we created comcast-master... traditionally we also have code reviews for every commit, thus no direct pushing... I think it's better if Azam, Eugene, Ihor would create pull request directly to your "master" then we'll pull to comcast-master

@philn
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philn commented Apr 27, 2016

Why can't you pull master into your builds?

philn pushed a commit that referenced this pull request Feb 26, 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. (#185)
Enalbing low latency mode for RTC (#169)
Enable HEVC support. (#165)
Fix out-of-bounds write in H265VpsSpsPpsTracker (#163)
Apply fix bitstream logic to RtpVideoStreamReceiver2 (#142)
Add missing CODEC_H265 switch case (#136)
Add HEVC support for iOS/Android (#68)
H265 packetization_mode setting fix (#53)
Add H.265 QP parsing logic (#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 <jianlin.qiu@intel.com>
jianjunz <jianjun.zhu@intel.com>
Cyril Lashkevich <notorca@gmail.com>
Piasy <xz4215@gmail.com>
ShiJinCheng <874042641@qq.com>
Andreas Unterhuber <andreas.unterhuber@keepinmind.info>
dong-heun <63987238+dong-heun@users.noreply.github.com>

* 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
philn pushed a commit that referenced this pull request Mar 11, 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. (#185)
Enalbing low latency mode for RTC (#169)
Enable HEVC support. (#165)
Fix out-of-bounds write in H265VpsSpsPpsTracker (#163)
Apply fix bitstream logic to RtpVideoStreamReceiver2 (#142)
Add missing CODEC_H265 switch case (#136)
Add HEVC support for iOS/Android (#68)
H265 packetization_mode setting fix (#53)
Add H.265 QP parsing logic (#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 <jianlin.qiu@intel.com>
jianjunz <jianjun.zhu@intel.com>
Cyril Lashkevich <notorca@gmail.com>
Piasy <xz4215@gmail.com>
ShiJinCheng <874042641@qq.com>
Andreas Unterhuber <andreas.unterhuber@keepinmind.info>
dong-heun <63987238+dong-heun@users.noreply.github.com>

* 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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3 participants