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WWebRTC LUCI CQAdds reference time to webrt::VideoFrame
7b6f9963创建于 2023年10月30日历史提交
文件最后提交记录最后更新时间
test
Added AsString() function for color space for easier debugging Change-Id: I517a435769795de26502aea0dd3e01c1ed867616 Bug: chromium:1449570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320166 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40898} 2 年前
BUILD.gn
Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by - moving getters for EncodedImage fields up to EncodedImage - copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame - Removing EncodedFrame's inheritance of VCMEncodedFrame We leave VCMEncodedFrame as part of the (near) deprecated VideoCodingModule code. The only place which needs to accept either is in the generic decoder. Bug: webrtc:9378, b:296992877 Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40639} 2 年前
DEPS
Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by - moving getters for EncodedImage fields up to EncodedImage - copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame - Removing EncodedFrame's inheritance of VCMEncodedFrame We leave VCMEncodedFrame as part of the (near) deprecated VideoCodingModule code. The only place which needs to accept either is in the generic decoder. Bug: webrtc:9378, b:296992877 Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40639} 2 年前
OWNERS
Revise video owners Bug: None No-try: True Change-Id: Ibc8dcb22d0ca81897dc63d39ff13372b0fc7302d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277402 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Fanny Linderborg <linderborg@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38255} 3 年前
builtin_video_bitrate_allocator_factory.cc
Remove IsSinglecastOrAllNonFirstLayersInactive() helper. As of recent changes, we can simply look at numberOfSimulcastStreams because in the {active,inactive,inactive} case we get a single webrtc::VideoStream here[1] which results in numberOfSimulcastStreams being 1 here[2]. Looking at numberOfSimulcastStreams instead of using a helper is preferred because it is more descriptive and in the future, when {inactive,active,inactive} or {inactive,inactive,active} cases of VP9 simulcast is also supported (webrtc:15046) then this gating will work even when the first layer is not the active one. [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/config/encoder_stream_factory.cc;l=146;drc=c99753ac8f051e379ae68e281aaef04b0a5ca8f2 [2] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/video_codec_initializer.cc;l=77;drc=4baea5b07f2fd309892845cf2d1c0f4ca77862d3 # No need to wait for win chrome bot, everything else green NOTRY=True Bug: webrtc:15046 Change-Id: I8aaea2e8cc350bd01fb00cc7fd85032b7fdfe24d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299942 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39759} 3 年前
builtin_video_bitrate_allocator_factory.h
Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer" Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782 This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken. Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner. One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed. Bug: webrtc:9513 Change-Id: I38708762ff365e4ca05974b99fac71edc739a756 Reviewed-on: https://webrtc-review.googlesource.com/c/109040 Commit-Queue: Jiawei Ou <ouj@fb.com> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25574}7 年前
color_space.cc
Added AsString() function for color space for easier debugging Change-Id: I517a435769795de26502aea0dd3e01c1ed867616 Bug: chromium:1449570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320166 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40898} 2 年前
color_space.h
Added AsString() function for color space for easier debugging Change-Id: I517a435769795de26502aea0dd3e01c1ed867616 Bug: chromium:1449570 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320166 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40898} 2 年前
encoded_frame.cc
Remove VCMEncodedFrame from webrtc::EncodedFrame inheritance Remove VCMEncodedFrame from the inheritance chain of EncodedFrames by - moving getters for EncodedImage fields up to EncodedImage - copying other non-deprecated fields & Methods from VCMEncodedFrame over to EncodedFrame - Removing EncodedFrame's inheritance of VCMEncodedFrame We leave VCMEncodedFrame as part of the (near) deprecated VideoCodingModule code. The only place which needs to accept either is in the generic decoder. Bug: webrtc:9378, b:296992877 Change-Id: I60706aebbb6eacc7fd4b35ec90cc903efdbe14c8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317160 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40639} 2 年前
encoded_frame.h
Remove EncodedFrame::MissingFrame and start removing Decode() param Remove EncodedFrame::MissingFrame, as it was always false in actual in-use code anyway, and remove usages of the Decode missing_frames param within WebRTC. Uses/overrides in other projects will be cleaned up shortly, allowing that variant to be removed from the interface. Bug: webrtc:15444 Change-Id: Id299d82e441a351deff81c0f2812707a985d23d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317802 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Tony Herre <herre@google.com> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#40662} 2 年前
encoded_image.cc
Update RtpSenderVideo::SendVideo/SendEncodedImage to take Timestamp/TimeDelta types Bug: webrtc:13757 Change-Id: I2f21b14ecf003c5cb0c4c92d0c6b9b6f11c35f71 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311945 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40450} 2 年前
encoded_image.h
Delete deprecated Timestamp accessor and setter in EncodedImage Bug: webrtc:9378 Change-Id: I5c67cca733f2fd646e73694524abf6b33438e8a4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321860 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40926} 2 年前
frame_buffer.cc
Rename EncodedImage property Timetamp to RtpTimestamp To avoid name collision with Timestamp type, To avoid confusion with capture time represented as Timestamp Bug: webrtc:9378 Change-Id: I8438a9cf4316e5f81d98c2af9dc9454c21c78e70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320601 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40796} 2 年前
frame_buffer.h
FrameBuffer::InsertFrame returns true on successful insertion This is cleaner than checking the size before and after, as is currently done in FrameBufferProxy Bug: webrtc:14168 Change-Id: Iac896ddf7b1b0b8513159451de7cd8a10668a49a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265663 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37222} 3 年前
frame_buffer_unittest.cc
FrameBuffer::InsertFrame returns true on successful insertion This is cleaner than checking the size before and after, as is currently done in FrameBufferProxy Bug: webrtc:14168 Change-Id: Iac896ddf7b1b0b8513159451de7cd8a10668a49a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265663 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37222} 3 年前
hdr_metadata.cc
Pass HdrMetadata between VideoFrame and EncodedImage for VP9 Bug: webrtc:8651 Change-Id: Ie4d7ee19bead84eda7788076662c4066edc3f024 Reviewed-on: https://webrtc-review.googlesource.com/c/109583 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25581}7 年前
hdr_metadata.h
Reorder methods and members of HdrMetadata Bug: webrtc:8651 Change-Id: I67941a5918d5cd31a7b04b11aa20c500d49e9a62 Reviewed-on: https://webrtc-review.googlesource.com/c/114283 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26002}7 年前
i010_buffer.cc
Use libyuv rotate methods Bug: webrtc:13826 Change-Id: I10a3b291a66eae1b867dd2fa1a1781c235feef33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290703 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39060} 3 年前
i010_buffer.h
Remove unused IXXXBuffer::PasteFrom Bug: webrtc:13262 Change-Id: Iac383ca5a30abd082eb93af8acdef40d6537ce7d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235202 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35264} 4 年前
i210_buffer.cc
Use libyuv rotate methods Bug: webrtc:13826 Change-Id: I10a3b291a66eae1b867dd2fa1a1781c235feef33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290703 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39060} 3 年前
i210_buffer.h
Add 420 and 422 10 bit h264 decoding. 422 10 bit format is called I210 in the code and implemented in I210Buffer, and 420 10-bit format format is using is using the already existing I010 format and implemented in I010Buffer. Bug: webrtc:13826 Change-Id: I6b6ed65b9fbb295386ea20f751bd0badc49ef21b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256964 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37252} 3 年前
i410_buffer.cc
Add 444 10 bits support for H264 and VP9 This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate. Bug: webrtc:14818 Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39123} 3 年前
i410_buffer.h
Add 444 10 bits support for H264 and VP9 This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate. Bug: webrtc:14818 Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39123} 3 年前
i420_buffer.cc
Add rtc::make_ref_counted to api/ This cl adds a forwarding header, a build target, and migrates headers in api/ to use it. Moving actual implementation, will follow, in https://webrtc-review.googlesource.com/c/src/+/265390. Bug: webrtc:12701 Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37200} 3 年前
i420_buffer.h
Update old TODO comments Bug: None Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37436} 3 年前
i422_buffer.cc
Use libyuv rotate methods Bug: webrtc:13826 Change-Id: I10a3b291a66eae1b867dd2fa1a1781c235feef33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290703 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39060} 3 年前
i422_buffer.h
Update old TODO comments Bug: None Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37436} 3 年前
i444_buffer.cc
Add rtc::make_ref_counted to api/ This cl adds a forwarding header, a build target, and migrates headers in api/ to use it. Moving actual implementation, will follow, in https://webrtc-review.googlesource.com/c/src/+/265390. Bug: webrtc:12701 Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37200} 3 年前
i444_buffer.h
Update old TODO comments Bug: None Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37436} 3 年前
nv12_buffer.cc
Add rtc::make_ref_counted to api/ This cl adds a forwarding header, a build target, and migrates headers in api/ to use it. Moving actual implementation, will follow, in https://webrtc-review.googlesource.com/c/src/+/265390. Bug: webrtc:12701 Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37200} 3 年前
nv12_buffer.h
Update old TODO comments Bug: None Change-Id: I96850df6cfa19303043108a59ef60d7b686ec747 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37436} 3 年前
recordable_encoded_frame.h
Reland "RtpEncodingParameters::request_resolution patch 1" This reverts commit b625101da8d798c936cfd695505a5514644158b0. Reason for revert: Found problem that was specific how configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715. Thanks Rasmus and great that this was tested! Original change's description: > Revert "RtpEncodingParameters::request_resolution patch 1" > > This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2. > > Reason for revert: Breaks downstream test > > Original change's description: > > RtpEncodingParameters::request_resolution patch 1 > > > > This patch adds RtpEncodingParameters::request_resolution > > with documentation and plumming. No behaviour is changed yet. > > > > Bug: webrtc:14451 > > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Commit-Queue: Jonas Oreland <jonaso@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38172} > > Bug: webrtc:14451 > Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541 > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org> > Owners-Override: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38176} Bug: webrtc:14451 Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38178} 3 年前
render_resolution.h
Reland "RtpEncodingParameters::request_resolution patch 1" This reverts commit b625101da8d798c936cfd695505a5514644158b0. Reason for revert: Found problem that was specific how configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715. Thanks Rasmus and great that this was tested! Original change's description: > Revert "RtpEncodingParameters::request_resolution patch 1" > > This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2. > > Reason for revert: Breaks downstream test > > Original change's description: > > RtpEncodingParameters::request_resolution patch 1 > > > > This patch adds RtpEncodingParameters::request_resolution > > with documentation and plumming. No behaviour is changed yet. > > > > Bug: webrtc:14451 > > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262 > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Commit-Queue: Jonas Oreland <jonaso@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38172} > > Bug: webrtc:14451 > Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541 > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Reviewed-by: Jonas Oreland <jonaso@webrtc.org> > Owners-Override: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#38176} Bug: webrtc:14451 Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38178} 3 年前
resolution.h
RtpEncodingParameters::request_resolution patch 2 This cl/ implements configuring of encode resolution in the video_stream_encoder (webrtc_video_engine) in a way that is independent of frame resolution (i.e not using scale_resolution_down_by). The cl/ reuses the VideoAdapter as is, and hence the output resolution will be the same as it is today. Anticipated further patches 3) Hook up resource adaptation 4) Let VideoSource do adaption if possible Bug: webrtc:14451 Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38245} 3 年前
rtp_video_frame_assembler.cc
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
rtp_video_frame_assembler.h
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
rtp_video_frame_assembler_unittests.cc
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
video_adaptation_counters.cc
[Adaptation] Add more ResourceAdaptationProcessor logging. This should help debugging when adaptation is or is not happening unexpectedly. Log spam is prevented by not logging if the same result happened to the same resource already and we haven't adapted since then. Bug: webrtc:11616 Change-Id: Ia6c5cc35061d252f1c66f2f2bf3b94d2485498d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176221 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31378} 5 年前
video_adaptation_counters.h
[Adaptation] Add more ResourceAdaptationProcessor logging. This should help debugging when adaptation is or is not happening unexpectedly. Log spam is prevented by not logging if the same result happened to the same resource already and we haven't adapted since then. Bug: webrtc:11616 Change-Id: Ia6c5cc35061d252f1c66f2f2bf3b94d2485498d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176221 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31378} 5 年前
video_adaptation_reason.h
Unify AdaptationReason and AdaptReason enums. Moves the unified AdaptationReason to the api/ folder. Bug: webrtc:11392 Change-Id: I28782e82ef6cc3ca3b061f65b0bbdc3766df1f9c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172583 Commit-Queue: Evan Shrubsole <eshr@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31084} 6 年前
video_bitrate_allocation.cc
Count disabled due to low bw streams or layers as bw limited quality in GetStats Bug: webrtc:11015 Change-Id: I65cd890706f765366d89ded8c21fa7507797fc23 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155964 Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29421} 6 年前
video_bitrate_allocation.h
Use backticks not vertical bars to denote variables in comments for /api Bug: webrtc:12338 Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34554} 4 年前
video_bitrate_allocator.cc
Reland "Only enable conference mode simulcast allocations with flag enabled" This is a reland of 32ca95145c4636374266f5b5d4d1ac43658bc758 Fix includes not enabling the screenshare conference behavior on non screenshare sources even if the flag is enabled. Original change's description: > Only enable conference mode simulcast allocations with flag enabled > > Non-conference mode simulcast screenshares were mistakenly using the > conference mode semantics in the simulcast rate allocator, which broke > spec compliant usage in some situation. > > This behavior should only be used when explicitly using the SDP entry > "a=x-google-flag:conference" in both offer and answer. > > Bug: webrtc:11310, chromium:1093819 > Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31828} Bug: webrtc:11310 Bug: chromium:1093819 Change-Id: Ic933f93a5c4bad20583354fe821f8a1170e911cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180802 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31847} 5 年前
video_bitrate_allocator.h
Reland "Only enable conference mode simulcast allocations with flag enabled" This is a reland of 32ca95145c4636374266f5b5d4d1ac43658bc758 Fix includes not enabling the screenshare conference behavior on non screenshare sources even if the flag is enabled. Original change's description: > Only enable conference mode simulcast allocations with flag enabled > > Non-conference mode simulcast screenshares were mistakenly using the > conference mode semantics in the simulcast rate allocator, which broke > spec compliant usage in some situation. > > This behavior should only be used when explicitly using the SDP entry > "a=x-google-flag:conference" in both offer and answer. > > Bug: webrtc:11310, chromium:1093819 > Change-Id: Ibcba75c88a8405d60467546b33977a782e04e469 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179081 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Commit-Queue: Florent Castelli <orphis@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31828} Bug: webrtc:11310 Bug: chromium:1093819 Change-Id: Ic933f93a5c4bad20583354fe821f8a1170e911cd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180802 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31847} 5 年前
video_bitrate_allocator_factory.h
Format almost everything. This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505} 6 年前
video_codec_constants.h
Allow encoders to receive preferred pixel formats from native buffers Adds a field to EncoderInfo called preferred_pixel_formats which a software encoder populates with the pixel formats it supports. When a kNative frame is received for encoding, the VideoStreamEncoder will first try to get a frame that is accessible by the software encoder in that pixel format from the kNative frame. If this fails it will fallback to converting the frame using ToI420. This minimizes the number of conversions made in the case that the encoder supports the pixel format of the native buffer or where conversion can be accelerated. For example, in Chromium, the capturer can emit an NV12 frame, which can be consumed by libvpx which supports NV12. Testing: Tested in Chrome with media::VideoFrame adapters. Bug: webrtc:11977 Change-Id: I9becc4100136b0c0128f4fa06dedf9ee4dc62f37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187121 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#32353} 5 年前
video_codec_type.h
Add codec name H265 to support H265 in WebRTC Bug: webrtc:13485 Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40773} 2 年前
video_content_type.cc
Remove extra usage of video-content-type header extension This extension is documented to carry one bit: Screenshare. It's been used for carrying simulcast layers and experiment IDs. This CL removes that usage. Bug: webrtc:15383 Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40457} 2 年前
video_content_type.h
Remove extra usage of video-content-type header extension This extension is documented to carry one bit: Screenshare. It's been used for carrying simulcast layers and experiment IDs. This CL removes that usage. Bug: webrtc:15383 Change-Id: I048b283cde59bf1f607d8abdd53ced07a7add6f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312420 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40457} 2 年前
video_frame.cc
Adds reference time to webrt::VideoFrame The new reference time contains a monotonically increasing clock time and represents the time when the frame was captured. Not all platforms provide the "true" sample capture time in |reference_time| but might instead use a somewhat delayed (by the time it took to capture the frame) version of it. Bug: webrtc:15539 Change-Id: I95eff8b0f7bff8d3ae65798bf82046e1ac2b0cf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325261 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Markus Handell <handellm@google.com> Cr-Commit-Position: refs/heads/main@{#41036} 2 年前
video_frame.h
Adds reference time to webrt::VideoFrame The new reference time contains a monotonically increasing clock time and represents the time when the frame was captured. Not all platforms provide the "true" sample capture time in |reference_time| but might instead use a somewhat delayed (by the time it took to capture the frame) version of it. Bug: webrtc:15539 Change-Id: I95eff8b0f7bff8d3ae65798bf82046e1ac2b0cf2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325261 Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Markus Handell <handellm@google.com> Cr-Commit-Position: refs/heads/main@{#41036} 2 年前
video_frame_buffer.cc
Add 444 10 bits support for H264 and VP9 This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate. Bug: webrtc:14818 Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39123} 3 年前
video_frame_buffer.h
Add 444 10 bits support for H264 and VP9 This CL adds support for I410 buffers (444 10 bits) and modify vp9 and h264 for being able to convert input buffer to it when appropiate. Bug: webrtc:14818 Change-Id: I2fb3dc9d80c5338944c6df74dd6217a0454180d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290721 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39123} 3 年前
video_frame_metadata.cc
Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test" This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e. Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9()) Original change's description: > Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test" > > This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084. > > Reason for revert: Breaks upstream project > > Original change's description: > > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test > > > > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame. > > > > Also default-initialized VideoFrameMetadata::ssrc_ to 0. > > > > Bug: webrtc:14708 > > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560 > > Commit-Queue: Tove Petersson <tovep@google.com> > > Reviewed-by: Tony Herre <herre@google.com> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39411} > > Bug: webrtc:14708 > Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Auto-Submit: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39413} Bug: webrtc:14708 Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tove Petersson <tovep@google.com> Cr-Commit-Position: refs/heads/main@{#39418} 3 年前
video_frame_metadata.h
Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test" This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e. Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9()) Original change's description: > Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test" > > This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084. > > Reason for revert: Breaks upstream project > > Original change's description: > > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test > > > > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame. > > > > Also default-initialized VideoFrameMetadata::ssrc_ to 0. > > > > Bug: webrtc:14708 > > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560 > > Commit-Queue: Tove Petersson <tovep@google.com> > > Reviewed-by: Tony Herre <herre@google.com> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39411} > > Bug: webrtc:14708 > Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Auto-Submit: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39413} Bug: webrtc:14708 Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tove Petersson <tovep@google.com> Cr-Commit-Position: refs/heads/main@{#39418} 3 年前
video_frame_metadata_unittest.cc
Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test" This reverts commit 8bf321062973939ef35f529640f5e69852e89a7e. Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9()) Original change's description: > Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test" > > This reverts commit 437bf78ed9518b21fc39b94f6ee42d5b157e6084. > > Reason for revert: Breaks upstream project > > Original change's description: > > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test > > > > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame. > > > > Also default-initialized VideoFrameMetadata::ssrc_ to 0. > > > > Bug: webrtc:14708 > > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560 > > Commit-Queue: Tove Petersson <tovep@google.com> > > Reviewed-by: Tony Herre <herre@google.com> > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39411} > > Bug: webrtc:14708 > Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Auto-Submit: Andrey Logvin <landrey@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39413} Bug: webrtc:14708 Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Tove Petersson <tovep@google.com> Cr-Commit-Position: refs/heads/main@{#39418} 3 年前
video_frame_type.h
Use common VideoFrameTypeToString helper This CL cleans up all local conversions, in favor of the common helper function. Bug: webrtc:15210 Change-Id: Id77e1c6b1151a2542d92e220e91d5b11285479b8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311060 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianhui J Dai <jianhui.j.dai@intel.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40420} 2 年前
video_layers_allocation.h
Add experimental extension RtpVideoLayersAllocation The extension is suggested to be used for signaling per target bitrate, resolution and frame rate to a SFU to allow a SFU to know what video layers a client is currently targeting. It is hoped to replace the current Target bitrate RTCP XR message currently used only for screen share. Bug: webrtc:12000 Change-Id: Id7b55e7ddaf6304e31839fd0482b096e1dbe8925 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185980 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32313} 5 年前
video_rotation.h
Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}8 年前
video_sink_interface.h
Route min/max FPS constraints to VideoStreamEncoder. This change - adds new type VideoTrackSourceConstraints expressing min/max FPS constraints. - adds new method VideoTrackSourceInterface::ProcessConstraints. - adds new method VideoSinkInterface<>::OnConstraintsChanged. - updates AdaptedVideoTrackSource and VideoBroadcaster to forward the constraints to sinks. - adds several unit tests for the added functionality. - and finally, implements OnConstraintsChanged in VideoStreamEncoder. Chromium will be updated in coming CLs to supply constraints set through the MediaStream module. go/rtc-0hz-present Bug: chromium:1255737 No-Try: true Change-Id: Iffef239217269c332a1aaa902ddeae2440929e22 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235040 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35197} 4 年前
video_source_interface.cc
Move VideoStreamEncoderInterface to api/. Bug: webrtc:8830 Change-Id: I17908b4ef6a043acf22e2110b9672012d5fa7fc0 Reviewed-on: https://webrtc-review.googlesource.com/74481 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23334}7 年前
video_source_interface.h
2 年前
video_stream_encoder_settings.h
Switch encoder on init failure Currently if encoder initialization fails WebRTC doesn't send any video. This CL adds functionality that changes encoder type in such case and restores the video. If encoder selector is available we switch to encoder it recommends. Otherwise, VP8 is used as the default fallback encoder. Bug: webrtc:13572 Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35761} 4 年前
video_timing.cc
In VideoPlayoutDelay delete access to integer representation of min/max values Bug: webrtc:13756 Change-Id: I1a81c25e5e3fab68a44e94a5ab93e8184c824683 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316864 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40612} 2 年前
video_timing.h
In VideoPlayoutDelay delete access to integer representation of min/max values Bug: webrtc:13756 Change-Id: I1a81c25e5e3fab68a44e94a5ab93e8184c824683 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316864 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40612} 2 年前