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Progressive tests #136

Merged
merged 1 commit into from
Nov 29, 2016
Merged

Progressive tests #136

merged 1 commit into from
Nov 29, 2016

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ghost
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@ghost ghost commented Nov 28, 2016

Progressive tests stuck/timeout after 21. currentSrc

@eocanha
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eocanha commented Nov 29, 2016

Looks good, but if you don't mind I'm going to squash the two commits into a single one before merging.

@eocanha eocanha merged commit dae12f5 into master Nov 29, 2016
@philn philn deleted the Progressive_tests branch December 5, 2016 17:39
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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