Skip to content
This repository has been archived by the owner on Oct 25, 2024. It is now read-only.

Fix HEVC depacketizer issues. #185

Merged
merged 1 commit into from
Jan 31, 2023

Conversation

jianjunz
Copy link
Collaborator

  • Incorrect VPS ID.
  • FindFrames doesn't return packets.

- Incorrect VPS ID.
- FindFrames doesn't return packets.
@jianjunz jianjunz added the bug Something isn't working label Jan 31, 2023
@@ -447,7 +447,7 @@ std::vector<std::unique_ptr<PacketBuffer::Packet>> PacketBuffer::FindFrames(
}
}
#endif
if (is_h264 || full_frame_found) {
if (is_h264 || is_h265 || full_frame_found) {
Copy link
Collaborator

Choose a reason for hiding this comment

The reason will be displayed to describe this comment to others. Learn more.

Fine with that but why 108 explicitly introduce the full_frame_found flag for non hevc/avc streams?

Copy link
Collaborator Author

Choose a reason for hiding this comment

The reason will be displayed to describe this comment to others. Learn more.

@jianjunz jianjunz merged commit 39e3528 into open-webrtc-toolkit:108-sdk Jan 31, 2023
dvrogozh added a commit to intel/cloud-streaming that referenced this pull request Feb 7, 2023
Use --fake_audio on OWT build since we don't need real audio support inside OWT.

The built was done from this commit:
* open-webrtc-toolkit/owt-client-native@e629093

Corresponding build command:
  python3 scripts/build_linux.py --gn_gen --sdk --cloud_gaming --output_path dist \
    --scheme release --arch x64 --use_gcc --shared --fake_audio

Requires: open-webrtc-toolkit/owt-client-native#654
Requires: open-webrtc-toolkit/owt-client-native#655
Requires: open-webrtc-toolkit/owt-client-native#656
Requires: open-webrtc-toolkit/owt-client-native#666
Requires: open-webrtc-toolkit/owt-client-native#667
Requires: open-webrtc-toolkit/owt-client-native#668
Requires: open-webrtc-toolkit/owt-client-native#669
Requires: open-webrtc-toolkit/owt-deps-webrtc#182
Requires: open-webrtc-toolkit/owt-deps-webrtc#185
Requires: open-webrtc-toolkit/owt-client-native#672
Requires: open-webrtc-toolkit/owt-client-native#679
Requires: open-webrtc-toolkit/owt-deps-webrtc#187
Signed-off-by: Dmitry Rogozhkin <dmitry.v.rogozhkin@intel.com>
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 <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 to WebPlatformForEmbedded/WPEWebKit 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 to WebPlatformForEmbedded/WPEWebKit 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
jacek-manko-red pushed a commit to LibertyGlobal/WPEWebKit that referenced this pull request Aug 1, 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
Sign up for free to subscribe to this conversation on GitHub. Already have an account? Sign in.
Labels
bug Something isn't working
Projects
None yet
Development

Successfully merging this pull request may close these issues.

2 participants