New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
Speech recognition #53
Conversation
…WebKitForWayland into speech-recognition To rebase to master
b4a4f67
to
3f4ede2
Compare
The base work looks fine. New source files do need license headers though. |
#if ENABLE(SPEECH_RECOGNITION) | ||
|
||
#include "URL.h" | ||
#include "wtf/RefCounted.h" |
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
WTF headers are usually included via angle brackets: <wtf/RefCounted.h>
endif () | ||
|
||
if (SPHINX_FOUND) | ||
if (NOT SPHNIX_FIND_QUIETLY) |
There was a problem hiding this comment.
Choose a reason for hiding this comment
The reason will be displayed to describe this comment to others. Learn more.
SPHNIX -> SPHINX, same for the following two lines.
Thanks Zan for the comments. |
@Haseenamol Is this still on your todo list? Doing some cleaning of old PRs :) |
This was created long back and now there is lot of changes in the master. So I think, we can close the pull request for the time being and create new once we resume work on this |
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
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
SpeechRecogntition Work-in progress version for review:
Following are the status: Recognition start/stop/abort/events are implemented
Pending: Attributes lang, maxAlternatives, serviceURI, Grammar correction, Match string identification, speech training with more words and log message to WebKit log formats