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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
Add ScalabilityModeStringToEnum helper function. Bug: none Change-Id: Iea602c88afbfe1f8f8e94b353eda96d62b651bd8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319882 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40732} 2 年前
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
Adding some AV1 constants and helper functions I've added a basic AV1 impl to Chrome Remote Desktop and am looking into what is needed to test with I444 (Profile-1) in our platform. This CL adds a few helper functions, constants, and enums that can be used to configure the SDP with different AV1 profiles. More work is still needed but I wanted to get this in place first so I can build on it in the CRD host code. Change-Id: I1af9ebf31f833138e8c36e0c0a30e32289e7b58e Bug: chromium:1329660 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264000 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Joe Downing <joedow@google.com> Cr-Commit-Position: refs/heads/main@{#37182} 3 年前
Adding some AV1 constants and helper functions I've added a basic AV1 impl to Chrome Remote Desktop and am looking into what is needed to test with I444 (Profile-1) in our platform. This CL adds a few helper functions, constants, and enums that can be used to configure the SDP with different AV1 profiles. More work is still needed but I wanted to get this in place first so I can build on it in the CRD host code. Change-Id: I1af9ebf31f833138e8c36e0c0a30e32289e7b58e Bug: chromium:1329660 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264000 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Joe Downing <joedow@google.com> Cr-Commit-Position: refs/heads/main@{#37182} 3 年前
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327} 3 年前
Use std::make_unique instead of absl::make_unique. WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209} 6 年前
Export symbols needed by the Chromium component build (part 6). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: I67a4d016a11deca5ac5459826741dd2d3f7931d5 Reviewed-on: https://webrtc-review.googlesource.com/c/107400 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25298}7 年前
Delete EncoderSimulcastProxy in favor of SimulcastEncoderAdapter. Because the adapter has a passthrough mode, it can already handle both singlecast and simulcast cases, meaning the proxy is no longer providing value. Let's delete. Bug: webrtc:15001 Change-Id: I480eaba599448e9b82b8cf7f829dc35ad6ce0434 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297740 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39579} 3 年前
Export symbols needed by the Chromium component build (part 6). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: I67a4d016a11deca5ac5459826741dd2d3f7931d5 Reviewed-on: https://webrtc-review.googlesource.com/c/107400 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25298}7 年前
Fix incorrect use of scoped enumerations in format strings Scoped enums do not get automatically promoted to their underlying type, so these uses have undefined behavior and Clang recently started warning about it. Bug: chromium:1456289 Change-Id: I9cf4e5a68378930a3bf7d8ac7b0a21eaf0d12670 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309520 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Hans Wennborg <hans@chromium.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40321} 2 年前
Reland: Added support for H264 YUV444 (I444) decoding. PS#1 is a reland of "Added support for H264 YUV444 (I444) decoding." https://webrtc-review.googlesource.com/c/src/+/235340 Bug: chromium:1251096 Change-Id: Icd997c7f7732229954d5494343b4e7a70deb09d1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251303 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35964} 4 年前
Add H265 codec name and profile/tier/level utils. This adds H265 codec name and profile/tier/level handling needed for H265 SDP negotiation. Bug: webrtc:13485 Change-Id: I838b910042ce36f8ae3979c41a73ee46935c57d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315900 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> Cr-Commit-Position: refs/heads/main@{#40661} 2 年前
Add H265 codec name and profile/tier/level utils. This adds H265 codec name and profile/tier/level handling needed for H265 SDP negotiation. Bug: webrtc:13485 Change-Id: I838b910042ce36f8ae3979c41a73ee46935c57d8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315900 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com> Cr-Commit-Position: refs/heads/main@{#40661} 2 年前
Add support for more scalability modes (1.5:1 resolution ratio). Added modes: - S2T1h - S2T2h - S2T3h - S3T1h - S3T2h - S3T3h Bug: webrtc:13960 Change-Id: I618a30c68b0ce1609847ee33a2298fe8fa0720c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273664 Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37968} 3 年前
Add support for more scalability modes (1.5:1 resolution ratio). Added modes: - S2T1h - S2T2h - S2T3h - S3T1h - S3T2h - S3T3h Bug: webrtc:13960 Change-Id: I618a30c68b0ce1609847ee33a2298fe8fa0720c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273664 Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37968} 3 年前
Add ScalabilityModeStringToEnum helper function. Bug: none Change-Id: Iea602c88afbfe1f8f8e94b353eda96d62b651bd8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319882 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40732} 2 年前
Add ScalabilityModeStringToEnum helper function. Bug: none Change-Id: Iea602c88afbfe1f8f8e94b353eda96d62b651bd8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319882 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40732} 2 年前
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
Add FuzzyMatchSdpVideoFormat convenience function for VideoEncoderFactoryTemplate. Bug: webrtc:13573 Change-Id: I6813f2a2524271be7862b700da4831575ec6e206 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279701 Commit-Queue: Xavier Lepaul‎ <xalep@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38463} 3 年前
Add temporary method SimulcastStream::SetNumberOfTemporalLayers Similar to https://webrtc-review.googlesource.com/c/src/+/267843, it turns out we need the setter method too to transition downstream code. Bug: webrtc:11607 Change-Id: I50df5e9c5d9301717d527089de61fcf783267ee1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268760 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37545} 3 年前
Add temporary method SimulcastStream::SetNumberOfTemporalLayers Similar to https://webrtc-review.googlesource.com/c/src/+/267843, it turns out we need the setter method too to transition downstream code. Bug: webrtc:11607 Change-Id: I50df5e9c5d9301717d527089de61fcf783267ee1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268760 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37545} 3 年前
Move definition of SpatialLayer to api/video_codecs/spatial_layer.h Bug: webrtc:7660 Change-Id: I54009ebc5f65b6875a8c079ab5264e0c5ce9f654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181500 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31942} 5 年前
Move definition of SpatialLayer to api/video_codecs/spatial_layer.h Bug: webrtc:7660 Change-Id: I54009ebc5f65b6875a8c079ab5264e0c5ce9f654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181500 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31942} 5 年前
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
Reland "Add option to disable quality scaling for AV1." This reverts commit 83102d39077f82f2d4539c160c659dcf789a5fdb. Reason for revert: reland with fix Original change's description: > Revert "Add option to disable quality scaling for AV1." > > This reverts commit 446dbc66fde7e9d5e684d3f71e357c2076a91740. > > Reason for revert: downstream break > > Original change's description: > > Add option to disable quality scaling for AV1. > > > > The main goal of this change is to disable the quality scaler when multiple spatial layers are used. > > > > Bug: b/295129711 > > Change-Id: I25e0b7440a8c2adee3e97720a1e0ee5e0a914334 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319181 > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#40709} > > Bug: b/295129711 > Change-Id: Iaeb13951d1b839bc0426120436035843bb3ee98f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320081 > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Owners-Override: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40742} Bug: b/295129711 Change-Id: Iab4846c2cd6074f50a3ebe9551432d449243b5d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320120 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40743} 2 年前
Delete deprecated VideoDecoder::InitDecode Bug: webrtc:13045 Change-Id: Id1ca822c3be5a4f496dd67b59eab31a79a74bf67 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228949 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34884} 4 年前
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
Remove deprecated signature of VideoDecoderFactory::QueryCodecSupport This function was deprecated in this CL https://webrtc-review.googlesource.com/c/src/+/229184 Bug: chromium:1187565 Change-Id: Ic0e18af69185b48accc441c4bbe1a2d8926db383 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230241 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34868} 4 年前
New VideoDecoderFactoryTemplate. The VideoDecoderFactoryTemplate takes decoder implementations as template arguments, making it possible to easily implement a VideoDecoderFactory only using the implementations required for the particular application. This will replace the BuiltinVideoDecoderFactory. Bug: webrtc:13573 Change-Id: I0213acd20b69dacf06fc6934851b73bd19b1afc8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268470 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37523} 3 年前
Update Dav1dDecoderTemplateAdapater to include AV1 profile1 support. Bug: webrtc:13573 Change-Id: Ibc81b6400eb8e22ce28ae434c83a22013ef96c79 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299946 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39757} 3 年前
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327} 3 年前
New VideoDecoderFactoryTemplate. The VideoDecoderFactoryTemplate takes decoder implementations as template arguments, making it possible to easily implement a VideoDecoderFactory only using the implementations required for the particular application. This will replace the BuiltinVideoDecoderFactory. Bug: webrtc:13573 Change-Id: I0213acd20b69dacf06fc6934851b73bd19b1afc8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268470 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37523} 3 年前
Use Video{Encoder, Decoder}FactoryTemplate instead of Internal{Encoder, Decoder}Factory. Bug: webrtc:13573 Change-Id: Id0e46a9b6053dedae3cbf0e5581768868900630b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269247 Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37596} 3 年前
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
Export symbols needed by the Chromium component build (part 8). This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h) to mark WebRTC symbols as visible from a shared library, this doesn't mean these symbols are part of the public API (please continue to refer to [1] for info about what is considered public WebRTC API). [1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md Bug: webrtc:9419 Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27410}7 年前
Format ^(api|call|common_audio|examples|media|net|p2p|pc)/ half of the remaining folders git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -E "^(api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I8b2cac13f4587d3ce9b2fccc7362967283f57ea2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302062 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39977} 3 年前
Enable multithreaded OpenH264 encoding behind field trial This uses the field trial introduced is crbug.com/1406331 and extends the usage to OpenH264. This simplifies experimentation whether this change improves performance without requiring multi-slice encoding. BUG=webrtc:14368 Change-Id: I0031e59059f7113dd5453234869c957d46f311bb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294340 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39371} 3 年前
RtpSenderInterface::SetEncoderSelector This cl/ adds a way of setting an EncoderSelector on a specific RtpSenderInterface. This makes it possible to easily use different EncoderSelector on different streams within the same or different PeerConnections. The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor. Iff a EncoderSelector is set on the RtpSender, it will take precedence over the VideoEncoderFactory::GetEncoderSelector. Bug: webrtc:14122 Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37150} 3 年前
Fuzzy match the SdpVideoFormat in VideoEncoderFactoryTemplate::CreateVideoEncoder. Bug: webrtc:13573 Change-Id: I1223f5f989d5298d3a6f7f6d06fd752e650adfd9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300342 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39766} 3 年前
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327} 3 年前
Make header files self contained. This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327} 3 年前
Remove references to AudioCodec and VideoCodec constructors The preferred method to create codecs is to use the function cricket::CreateAudioCodec or cricketCreateVideoCodec. Empty codec objects are deprecated and should be replaced with alternatives such as methods returning an absl::optional object instead. Bug: webrtc:15214 Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40226} 2 年前
Remove references to AudioCodec and VideoCodec constructors The preferred method to create codecs is to use the function cricket::CreateAudioCodec or cricketCreateVideoCodec. Empty codec objects are deprecated and should be replaced with alternatives such as methods returning an absl::optional object instead. Bug: webrtc:15214 Change-Id: I7fe40f64673cd407830dbbb0e541b85a3aee93aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307521 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40226} 2 年前
Encoder type agnostic resolution based fallback WebRTC-Video-EncoderFallbackSettings/resolution_threshold_px:X sets resolution threshold to switch from primary to fallback encoder. When the field trial is present, VP8-specific resolution based fallback settings, provided by WebRTC-VP8-Forced-Fallback-Encoder-v2, are ignored. Bug: none Change-Id: I8f2e28438547f3896c7fc288ed6634720328f3a0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40526} 2 年前
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
Use backticks not vertical bars to denote variables in comments Bug: webrtc:12338 Change-Id: I89c8b3a328d04203177522cbdfd9e606fd4bce4c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228246 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34696} 4 年前
Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED. The new macros will replace the old one when old one's usage will be removed. The idea of the renaming to provide a clear signal that this is debug build only macros and will be stripped in the production build. Bug: webrtc:9065 Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35348} 4 年前
Make DefaultTemporalLayers explicitly request a key frame Bug: webrtc:10758 Change-Id: I426bfee7c3cdc2ac058f7e7f44368530a28b02a5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143169 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28345} 6 年前
Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create Previously, FecControllerOverride was passed to Vp8FrameBufferController::SetFecControllerOverride. Passing to the factory is a more elegant way, since it's only used when the controller is constructed. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28443} 6 年前
Pass FecControllerOverride to Vp8FrameBufferControllerFactory::Create Previously, FecControllerOverride was passed to Vp8FrameBufferController::SetFecControllerOverride. Passing to the factory is a more elegant way, since it's only used when the controller is constructed. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: Iae599889e7ca9003e3200c2911239cbb763ee65a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144380 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Elad Alon <eladalon@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28443} 6 年前
Enforce VideoEncoderConfig.num_temporal_layers >= 1. This change clarifies the semantics of this field: unset: Depends on context. == 0: Invalid. == 1: No temporal layering. >= 2: Temporal layering. We should try to remove the wrapping optional later. Bug: webrtc:11297 Change-Id: Id765f2dc1d31a4ba3cd424978ac6054cd60152ea Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166528 Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30336} 6 年前
Remove deprecated version of Vp8FrameBufferControllerFactory::Create Bug: webrtc:10769 Change-Id: I31de2d21d5325eb4b175815739c2477352842425 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144381 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28448} 6 年前
add 422 8 and 10 bit decoding support Bug: webrtc:14195 Change-Id: I2048d567850ae669d76d9e593752683f3c76499f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266180 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37306} 3 年前
add 422 8 and 10 bit decoding support Bug: webrtc:14195 Change-Id: I2048d567850ae669d76d9e593752683f3c76499f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266180 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37306} 3 年前