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
Remove rtc_base:rtc_base_approved It's now empty, let's remove it! Bug: webrtc:9838 Change-Id: I4b3310e882ea95fdf47903f9ad31e2efb35703f2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261242 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36774} 3 年前
Make AEC3 json parsing code testonly Reasons: - the code is no longer used in Chrome - it is conceptually weird for WebRTC to have JSON parsing in its API - there are concerns around the reliability of the underlying JSON library Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly. Bug: webrtc:1493351 Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062 Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41014} 2 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
Cleanup Call construction Return unique_ptr to clearly communicate ownership is transfered. Remove Call::Config alias Bug: None Change-Id: Ie3aa1da383ad65fae490d218fced443d44961eab Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323160 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40934} 2 年前
Reland "Enable SRTP GCM ciphers by default" This is a reland of commit d8633868b34dc1d841f0a9fd1afe2bc22aa8bde6 Original change's description: > Enable SRTP GCM ciphers by default > > Bug: webrtc:15178 > Change-Id: I0216ce8f194fffc820723d82b9c04a76573c2f4f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305381 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Reviewed-by: Victor Boivie <boivie@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40828} Bug: webrtc:15178 Change-Id: I5ea939ed6263547ebc177d9dd1763ba888936866 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321961 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Victor Boivie <boivie@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40862} 2 年前
Reland "Migrate WebRTC documentation to new renderer" This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. > > Reason for revert: > > Original change's description: > > Migrate WebRTC documentation to new renderer > > > > Bug: b/258408932 > > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39205} > > Bug: b/258408932 > Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39209} Bug: b/258408932 Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660 Owners-Override: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39231} 3 年前
Metronome: complete API migration. This CL finalizes the Metronome refactor undertaken in crbug.com/1381982 and enables it again in call.cc. Fixed: chromium:1381982 Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38605} 3 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
Add samples sum calculation Bug: b/261160916, webrtc:14852 Change-Id: I88e464fce4673dd9b9683219b8d2837206579ba5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293942 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39386} 3 年前
Add RtcEvent to store when MinimumSetDelay is set on NetEq To be able to simulate offline some scenario in which the javascript layer set the minimum base buffer size of neteq, it is required to record those API calls. This change introduces this. Bug: webrtc:14763 Change-Id: Ic817913eda60978d6fca3f8e12229aeec505ca25 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287122 Auto-Submit: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Lionel Koenig <lionelk@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39104} 3 年前
Add stats-related TODOs with crbugs. Someone wondered why framesRendered has not been implemented. I had a look, and discovered that a) we need to implement it, and b) our entire inter-frame, pause, and freeze metrics are measured at the wrong time because what WebRTC considered "OnRenderedFrame()" is not actually when the frame was rendered. So that we don't forget this again, I filed two crbugs and added TODOs in the code for future reference to anyone interested in these metrics. Bug: webrtc:15600, webrtc:15601 Change-Id: Id38df7874df715e9b9c0410efa4a9bc2af5d6232 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324306 Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40987} 2 年前
Allow absl::Nonnull and absl::Nullable. This CL includes follow-up changes from https://webrtc-review.googlesource.com/c/src/+/324280 Bug: none Change-Id: I6abad16e05cac7197c51ffa7b1d3fb991843df6e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325243 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41030} 2 年前
In PCLF remove ability to inject TaskQueueFactory and CallFactory Instead rely on TaskQueueFactory and Clock provided by the internal TimeController of the PCLF framework. Bug: webrtc:15574 Change-Id: I473e1f12ead97f866dbd45771ed5a59541c0c47c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325182 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41026} 2 年前
Export FieldTrialBasedConfig in order to use it in Chromium. Goal is to initialize peerconnections in Chromium using this based field trial config until a proper config that doesn't rely on the global field trial string can be used (https://crrev.com/c/4936314). Change-Id: I3d006e2445ccc4880b73b564c8ad4408242d3696 Bug: None Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323621 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Jeremy Leconte <jleconte@google.com> Cr-Commit-Position: refs/heads/main@{#40941} 2 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
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
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
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
Change PortInterface::Type to string_view and make type_ member const Bug: none Change-Id: Id1b0298eede5d2ae5010cc450d7bcb9eadd7b874 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318080 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40801} 2 年前
Rename api/stats_types.h to api/legacy_stats_types.h. As to not break downstream projects, the old name api/stats_types.h is kept around to help include api/legacy_stats_types.h. We can delete this in a follow-up. NOTRY=True Bug: webrtc:14180 Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38610} 3 年前
Rearrange api/OWNERS to show who's backup OWNERS tkchin and deadbeef are not working on webrtc on a daily basis at the moment, so non-urgent approvals should not go to them. Not mentioning this has led to misunderstandings. Bug: chromium:1371843 Change-Id: I91e99249d32e52d6083de9c2b1bfebfc4693acac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278201 Reviewed-by: Taylor Brandstetter <deadbeef@google.com> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38314} 3 年前
Discourage structs in api Structs make api harder to evolve: deprecated unused properties, change how data is represented. Classes with accessors allow more graduated and safer api evolution. Bug: None Change-Id: I8ebd5e072d51cf7f5800666cfdac523d0f9a937f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40714} 2 年前
Add reference, pointer, and co type aliases for rtc::ArrayView. Many STL containers define these type aliases, and they are easier to work with than add_const_t<add_lvalue_reference_t<value_type>>. In a followup, WTF::Vector in Blink's conversion constructor from other containers will be SFINAE-guarded using these type aliases. Bug: chromium:1408442 Change-Id: I7790e6f462a32e7e49bc6468afeda6b2e6d4b631 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300180 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Daniel Cheng <dcheng@chromium.org> Cr-Commit-Position: refs/heads/main@{#39771} 3 年前
rtc::ArrayView reverse iterators - rtc::ArrayView::rbegin() - rtc::ArrayView::rend() - rtc::ArrayView::crbegin() - rtc::ArrayView::crend() Bug: webrtc:7494 Change-Id: Id773d66cc9da8bd58def1dba35a706914440ef37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189880 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32458} 5 年前
Convert AsyncDnsResolver to use absl::AnyInvocable Bug: webrtc:12598 Change-Id: I0950231d6de7cf53116a573dcd97a3cf5514946c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318400 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40670} 2 年前
Reland "Deprecate all classes related to AsyncResolver" This reverts commit 08d431ec34a1c7ab52557702f2cebd9fdfacae9e. Reason for revert: Last (hopefully) Chrome blocker removed Original change's description: > Revert "Deprecate all classes related to AsyncResolver" > > This reverts commit 61a442809cc06de93a613186084d2dfa9c934d81. > > Reason for revert: Breaks roll into Chromium > > Original change's description: > > Deprecate all classes related to AsyncResolver > > > > and remove internal usage. > > > > Bug: webrtc:12598 > > Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860 > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#40919} > > Bug: webrtc:12598 > Change-Id: I8aef5e062e19a51baec75873eddfca2a10467d3c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322901 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40927} Bug: webrtc:12598 Change-Id: I3c7b07c831eb9ff808368433d9b9ae8ec4b2afb6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323720 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40944} 2 年前
Remove unused combined_audio_video_bwe. Bug: None Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com> Cr-Commit-Position: refs/heads/main@{#40160} 2 年前
Remove unused combined_audio_video_bwe. Bug: None Change-Id: Ie539351f98b7a0ebb5f08e0df5c5759a2bcb5588 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306520 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com> Cr-Commit-Position: refs/heads/main@{#40160} 2 年前
candidate: do not log full IP addresses for related address since this may contain sensitive data, just like the address. BUG=None Change-Id: I3faa1512a15467cd5cc4bcbcaebadb736f1bae07 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313040 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40473} 2 年前
Change PortInterface::Type to string_view and make type_ member const Bug: none Change-Id: Id1b0298eede5d2ae5010cc450d7bcb9eadd7b874 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318080 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40801} 2 年前
Updating AsyncAudioProcessing API, part 1. Add an API to pass AudioFrameProcessor as a unique_ptr. Bug: webrtc:15111 Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984 Reviewed-by: Olga Sharonova <olka@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40187} 2 年前
Updating AsyncAudioProcessing API, part 1. Add an API to pass AudioFrameProcessor as a unique_ptr. Bug: webrtc:15111 Change-Id: I4cefa35399c05c6e81c496e0b0387b95809bd8f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301984 Reviewed-by: Olga Sharonova <olka@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40187} 2 年前
Rename cipher_suite to crypto_suite and replace "cs" in the appropriate places. This is the terminology used by https://www.rfc-editor.org/rfc/rfc4568#section-10.3.2.1 and https://www.iana.org/assignments/sdp-security-descriptions/sdp-security-descriptions.xhtml BUG=None Change-Id: I45f2c52eb266c0f94bdd710a9b941142b9411827 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314483 Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40502} 2 年前
[DataChannelInterface] Introduce DataChannelInterface::SendAsync() One problem with the existing Send() method is that it has a return value that is problematic for a fully async implementation. A second problem with Send() is that the return value is bool and not RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError. Also, start deprecating bool Send() in favor of void SendAsync() and adding network_safety_ flag for posting async operations to the network thread. This flag also takes over from the connected_to_transport_ which can now be removed. Bug: webrtc:11547, webrtc:13289 Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39817} 3 年前
[DataChannelInterface] Introduce DataChannelInterface::SendAsync() One problem with the existing Send() method is that it has a return value that is problematic for a fully async implementation. A second problem with Send() is that the return value is bool and not RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError. Also, start deprecating bool Send() in favor of void SendAsync() and adding network_safety_ flag for posting async operations to the network thread. This flag also takes over from the connected_to_transport_ which can now be removed. Bug: webrtc:11547, webrtc:13289 Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39817} 3 年前
Make DTLS role visible on DtlsTransport interface This is important for writing tests that affect the DTLS role. Bug: webrtc:13668 Change-Id: I5d9a93eca7996a8b74cdcfe412f59a85892e1ec1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251389 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35971} 4 年前
Make DTLS role visible on DtlsTransport interface This is important for writing tests that affect the DTLS role. Bug: webrtc:13668 Change-Id: I5d9a93eca7996a8b74cdcfe412f59a85892e1ec1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251389 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35971} 4 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
Add SetRetransmissionMode() to FecController, this will be used to control RTX settings in FecController. Currently FecController knows about network conditions, these information can be used to control RTX settings in-call. Change-Id: I8f84164aeac48ea13b7f1cf82fd7424431f98ada Bug: webrtc:15167 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304800 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40192} 2 年前
Define FecControllerOverride and plumb it down to VideoEncoder The purpose of this interface is to allow VideoEncoder to override the bandwidth allocation set by FecController in RtpVideoSender. This CL defines the interface and sends it down to VideoSender. Two upcoming CLs will: 1. Make LibvpxVp8Encoder pass it on to the (injectable) FrameBufferController, where it might be put to good use. 2. Modify RtpVideoSender to respond to the message sent to it via this API. TBR=kwiberg@webrtc.org Bug: webrtc:10769 Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962 Commit-Queue: Elad Alon <eladalon@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28416} 6 年前
Add FieldTrialsRegistry that verifies looked up field trials This new class implements the existing FieldTrialsView interface, extending it with the verification functionality. For now, the verification will only be performed if the rtc_strict_field_trials GN arg is set. Most classes extending FieldTrialsView today have been converted to extend from FieldTrialsRegistry instead to automatically perform verification. Bug: webrtc:14154 Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38453} 3 年前
Add FieldTrialsRegistry that verifies looked up field trials This new class implements the existing FieldTrialsView interface, extending it with the verification functionality. For now, the verification will only be performed if the rtc_strict_field_trials GN arg is set. Most classes extending FieldTrialsView today have been converted to extend from FieldTrialsRegistry instead to automatically perform verification. Bug: webrtc:14154 Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38453} 3 年前
Add option to log a warning for unregistered field trials Until now you only had the option to RTC_DCHECK for unregistered field trials. This makes it possible to log a warning instead. Bug: webrtc:14154 Change-Id: I8628054e3c9b5d690f241a93e61299126b732ed0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39417} 3 年前
Add FieldTrialsRegistry that verifies looked up field trials This new class implements the existing FieldTrialsView interface, extending it with the verification functionality. For now, the verification will only be performed if the rtc_strict_field_trials GN arg is set. Most classes extending FieldTrialsView today have been converted to extend from FieldTrialsRegistry instead to automatically perform verification. Bug: webrtc:14154 Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38453} 3 年前
Use ScopedFieldTrials in FieldTrialsTest Resetting the global state between runs was previously handled by a RAII type, but the semantics of that type changed to remove this behavior in [1]. [1] https://webrtc-review.googlesource.com/c/src/+/276269 Bug: webrtc:14731, webrtc:14705 Change-Id: I8425cb71f49ea000434d500e0b3978324e4c3195 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285782 Commit-Queue: Emil Lundmark <lndmrk@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38800} 3 年前
Cleanup FieldTrialView Delete alias WebRtcKeyValueConfig as unused Replace .find() == 0 with absl::StartsWith per clang-tidy recommendation https://clang.llvm.org/extra/clang-tidy/checks/abseil/string-find-startswith.html Bug: webrtc:10335 Change-Id: I1f09c262844c0678a8d8c0d0d3274df3d947737c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299181 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39690} 3 年前
Make all encodedaudioframes inherit from TransformableAudioFrameI'face Make outgoing encoded audio frames inherit from the same Audio interface that incoming frames inherit from, to align them and make it possible to eg clone frames regardless of their direction. Also begin removing GetHeader() from the Audio interface, replacing it with getters for the specific values we actually need to propagate in the API: sequence number and CSRCs. This makes it much easier to treat incoming and outgoing frames the same, even if they don't have full RtpHeaders prepared at the point of the transform. Bug: chromium:1453226 Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Palak Agarwal <agpalak@google.com> Cr-Commit-Position: refs/heads/main@{#40309} 2 年前
Make all encodedaudioframes inherit from TransformableAudioFrameI'face Make outgoing encoded audio frames inherit from the same Audio interface that incoming frames inherit from, to align them and make it possible to eg clone frames regardless of their direction. Also begin removing GetHeader() from the Audio interface, replacing it with getters for the specific values we actually need to propagate in the API: sequence number and CSRCs. This makes it much easier to treat incoming and outgoing frames the same, even if they don't have full RtpHeaders prepared at the point of the transform. Bug: chromium:1453226 Change-Id: Ib5b39b30dea8a378b3b26efb1589dfd64741d201 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308141 Commit-Queue: Tony Herre <herre@google.com> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Palak Agarwal <agpalak@google.com> Cr-Commit-Position: refs/heads/main@{#40309} 2 年前
Expose video mimeType for insertable streams which allows determining what codec (data format) is used. Chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4941907 Split from https://webrtc-review.googlesource.com/c/src/+/318283 to reduce CL size and avoid audio woes. BUG=webrtc:15579 Change-Id: I404107af526df3009c16d2a6148784fe87dfa807 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323721 Reviewed-by: Tony Herre <herre@google.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#41007} 2 年前
Move rtc::FunctionView to the public API Bug: webrtc:10138 Change-Id: Icc25a2a277a9608701aaddd546882366739991ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27227}7 年前
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
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
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
Reland "Deprecate all classes related to AsyncResolver" This reverts commit 08d431ec34a1c7ab52557702f2cebd9fdfacae9e. Reason for revert: Last (hopefully) Chrome blocker removed Original change's description: > Revert "Deprecate all classes related to AsyncResolver" > > This reverts commit 61a442809cc06de93a613186084d2dfa9c934d81. > > Reason for revert: Breaks roll into Chromium > > Original change's description: > > Deprecate all classes related to AsyncResolver > > > > and remove internal usage. > > > > Bug: webrtc:12598 > > Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860 > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#40919} > > Bug: webrtc:12598 > Change-Id: I8aef5e062e19a51baec75873eddfca2a10467d3c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322901 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40927} Bug: webrtc:12598 Change-Id: I3c7b07c831eb9ff808368433d9b9ae8ec4b2afb6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323720 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40944} 2 年前
Reland "Remove old-style OnFailure callbacks" This is a reland of 1a290e4495c8132e7ff2c44d78de5e1d7eefdb9e after fixing the downstream projects. Original change's description: > Remove old-style OnFailure callbacks > > Also delete default implementation of new-style OnFailure, > since it can't call the deprecated function. > > Deprecating the old-style OnFailure callback turned out to be impossible, > since one can't have the new-style callback call the old-style one. > > Bug: chromium:589455 > Change-Id: Icf529ddb02d99ad9e205095d5a1fbeb0da91dd0e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146219 > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30570} Bug: chromium:589455 Change-Id: I7227e3c6886c504043b019b621e45658cbd6fd53 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168941 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30583} 6 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
Use unique_ptr in JsepCandidateCollection Bug: None Change-Id: I80ffacf3a355879b56a03b5cb59bffa32114dac1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147601 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28712} 6 年前
Remove RTC_DISALLOW_COPY_AND_ASSIGN more. Bug: webrtc:13555, webrtc:13082 Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35749} 4 年前
Remove JsepSessionDescription::kDefaultVideoCodecName which is only used in tests. BUG=None Change-Id: If215ad84e6756af2ee90777a27376400f8f4d8e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294721 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39450} 3 年前
Add missing method definition for StatsReport::Value::id_val() Also add a preprocessor definition to avoid redefinition in downstream projects. Bug: webrtc:15241 Change-Id: Ic55d98c3d3a69b9b19195ee78f03af6e38fdd0e2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308601 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Commit-Queue: Sameer Vijaykar <samvi@google.com> Cr-Commit-Position: refs/heads/main@{#40289} 2 年前
Remove preprocessor definition for StatsReport::Value::id_val() This is no longer needed after downstream redefinitions are deleted. Bug: webrtc:15241 Change-Id: Iea6839bff781fe7d0c56b4739f3d43398c70f2b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308681 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sameer Vijaykar <samvi@google.com> Cr-Commit-Position: refs/heads/main@{#40306} 2 年前
Implement support for Chrome task origin tracing. #2/4 This prepares TaskQueueBase sub classes to be able to migrate to the location and traits-based API. It re-introduces a Location class into the webrtc namespace, which is meant to be overridden by Chromium. Bug: chromium:1416199 Change-Id: I712c7806a71b3b99b2a2bf95e555b357c21c15ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294381 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39400} 3 年前
Fix missing libc++ includes in webrtc Several files refer to symbols declared in headers not explicitly included. This compiles now because libc++ tranitively includes these headers via other libc++ headers; however, these transitive includes are not guaranteed to exist and in Chrome, will no longer exist once libc++ is compiled with modules. Bug: chromium:543704 Change-Id: I638bb02df3d050a48345248e80aebd2dd60956c1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295762 Auto-Submit: Alan Zhao <ayzhao@google.com> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39448} 3 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
Change stream.AddTrack/RemoveTrack to take a scoped_refptr argument This better reflects the ownership passing of AddTrack, and is more consistent for RemoveTrack. Bug: webrtc:13980 Change-Id: Ide5baccf15fc687a4e092f8831ce8c0fea46604e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259740 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36603} 4 年前
Use string_view to pass track ids to constructors Bug: webrtc:13579 Change-Id: Icbd08d5fba9d150295675d730b7261d23992488c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264441 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37035} 3 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
Revert "Define cricket::MediaType in terms of webrtc::MediaType" This reverts commit 3ce6391b38397d107478041299fa998500e4fc42. Reason for revert: Breaks downstream test Original change's description: > Define cricket::MediaType in terms of webrtc::MediaType > > This is one step in getting rid of cricket::MediaType. > > Bug: webrtc:12754 > Fixes: webrtc:12764 > Change-Id: Idee832572bdc4c0e3bfdec6fb31ec0ba9db3e995 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218346 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33994} TBR=mbonadei@webrtc.org,hta@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: I64772018dea55e4f0946464364a60a39cec7e9ec No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:12754 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218603 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34000} 4 年前
Add base class NetworkPredictor and NetworkPredictorFactory and wire up. Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe. Bug: webrtc:10492 Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201 Commit-Queue: Ying Wang <yinwa@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27543}7 年前
Use SequenceChecker(SequenceChecker::kDetached) in a few places. This CL is partly a test to see if there's an impact on binary size: - Not a big difference for binaries (decrease): -776b to -4Kb - For libraries (libwebrtc.a) it actually increases the size: +40Kb Secondarily this CL is basically to introduce this pattern to the code base. In terms of LOC, this makes things slightly more compact. From: class Foo { public: Foo() { checker_.Detach(); } private: SequenceChecker checker_; }; To: class Foo { public: Foo() = default; private: SequenceChecker checker_{SequenceChecker::kDetached}; }; Bug: none Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39664} 3 年前
Reland "Deprecate all classes related to AsyncResolver" This reverts commit 08d431ec34a1c7ab52557702f2cebd9fdfacae9e. Reason for revert: Last (hopefully) Chrome blocker removed Original change's description: > Revert "Deprecate all classes related to AsyncResolver" > > This reverts commit 61a442809cc06de93a613186084d2dfa9c934d81. > > Reason for revert: Breaks roll into Chromium > > Original change's description: > > Deprecate all classes related to AsyncResolver > > > > and remove internal usage. > > > > Bug: webrtc:12598 > > Change-Id: Ie208682bfa0163f6c7a8e805151cfbda76324496 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322860 > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#40919} > > Bug: webrtc:12598 > Change-Id: I8aef5e062e19a51baec75873eddfca2a10467d3c > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322901 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Auto-Submit: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40927} Bug: webrtc:12598 Change-Id: I3c7b07c831eb9ff808368433d9b9ae8ec4b2afb6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323720 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40944} 2 年前
Deprecate AsyncResolver config fields and remove internal usage. Bug: webrtc:12598 Change-Id: Ic43cbcd13e4de44b02351c89da12844606368623 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317604 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40627} 2 年前
Reland: Remove unsupported configuration value, allow_codec_switching This reverts commit 6b0c5babe0700f12493cf659e1b35c58d2327995. Reason for revert: Relanding once downstream issues have been addressed Original change's description: > Revert "Remove unsupported configuration value, allow_codec_switching" > > This reverts commit 8f7a17f80f43a47ce3801a3cfd2afda3575c8023. > > Reason for revert: breaks downstream > > Original change's description: > > Remove unsupported configuration value, allow_codec_switching > > > > Bug: webrtc:11341 > > Change-Id: I8ff598848996bd63ccc572e11f8f69c892a4a459 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324284 > > Reviewed-by: Philip Eliasson <philipel@webrtc.org> > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#40995} > > Bug: webrtc:11341 > Change-Id: I784fd95062fc71f8dcc139b05121985f60709004 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324780 > Owners-Override: Philip Eliasson <philipel@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#40998} Bug: webrtc:11341 Change-Id: I3cb3e699fd76942c51f0f42a99bcb19ac607632e No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324782 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41032} 2 年前
Pass datachannel priority in DC open messages This adds priority to the API configuration of datachannels, and passes the value in the OPEN message. It does not yet influence SCTP prioritization of messages. Bug: chromium:1083227 Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31287} 5 年前
Remove RTC_DISALLOW_COPY_AND_ASSIGN more. Bug: webrtc:13555, webrtc:13082 Change-Id: I9c07708108da0a26f5e228384fd56cef4d1540b3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35749} 4 年前
Use absl::string_view type as parameter for RTCError message Bug: webrtc:13579 Change-Id: Ia9f90e6c3b008fc614d378cae4c407becfc597c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298447 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39610} 3 年前
Ensure that RTCErrorOr<T, E> doesn't require T to be default constructible Bug: webrtc:15214 Change-Id: Ic2d61c64d770943472374f61ad71249e88c3f6f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/307520 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40223} 2 年前
Use absl::string_view type as parameter for RTCError message Bug: webrtc:13579 Change-Id: Ia9f90e6c3b008fc614d378cae4c407becfc597c9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298447 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39610} 3 年前
Remove unnecessary overload in RtcEventLogOutput Bug: webrtc:13579 Change-Id: I3ea4b8ce8d111ae6b9ce7e92f75bd4196bc9656b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268420 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37508} 3 年前
Remove unnecessary std::string overloads Makes std::string version of rtc::RtcEventLogOutput::Write() no longer pure virtual while making the absl::string_view version pure virtual. Also removes unnecessary overloads in subclasses. BUG=webrtc:13579 Change-Id: I8fb449560b795a1ef76fab27533d9042d0c34cd1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268062 Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37484} 3 年前
Remove unnecessary std::string overloads Makes std::string version of rtc::RtcEventLogOutput::Write() no longer pure virtual while making the absl::string_view version pure virtual. Also removes unnecessary overloads in subclasses. BUG=webrtc:13579 Change-Id: I8fb449560b795a1ef76fab27533d9042d0c34cd1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268062 Commit-Queue: Ali Tofigh <alito@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37484} 3 年前
Remove unnecessary overload in RtcEventLogOutput Bug: webrtc:13579 Change-Id: I3ea4b8ce8d111ae6b9ce7e92f75bd4196bc9656b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268420 Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37508} 3 年前
Delete RTPHeader::payload_type_frequency as unused payload type frequency is not communicated inside an RTP packet and thus do not belong to the RTPHeader Bug: None Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39515} 3 年前
Delete RTPHeader::payload_type_frequency as unused payload type frequency is not communicated inside an RTP packet and thus do not belong to the RTPHeader Bug: None Change-Id: Ic3e48f1b0507a96ddc697503145f7c8785962926 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296763 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39515} 3 年前
RtpPacketInfo: deprecated ctors and getter removed Bug: webrtc:10739, b/246753278 Change-Id: I04d8a7886a7a1be7e155300a0c0ff3266fe6f28b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275944 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38152} 3 年前
RtpPacketInfo: deprecated ctors and getter removed Bug: webrtc:10739, b/246753278 Change-Id: I04d8a7886a7a1be7e155300a0c0ff3266fe6f28b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275944 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38152} 3 年前
Surface local_capture_clock_offset from RtpSource - Propagating RtpPacketInfo::local_capture_clock_offset, an existing field that is related to the abs-capture-timestamp header extension field estimated_capture_clock_offset - Propagated through SourceTracker::SourceEntry Bug: webrtc:10739, b/246753278 Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38129} 3 年前
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
RtpPacketInfo: new ctor + deprecated ctors clean-up New ctor added without optional and media specific fields. Bug: webrtc:10739, b/246753278 Change-Id: I7e15849aced6ed0a7ada725ea171a15ea1e9bc5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275941 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38124} 3 年前
Extract common codec fields into RtpCodec This creates the RtpCodec structure for the common fields used in RtpCodecParameters and RtpCodecCapability. Remove the unused fields from both that were defined from ORTC and never implemented as well. Bug: webrtc:15064 Change-Id: I37b4c83e2051a888fc99cc0d9f7aeb8d74f0421d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301182 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39862} 3 年前
Request keyframe via setParameters after the W3C changes in approach documented here: https://github.com/w3c/webrtc-extensions/pull/167 chromium CL: https://chromium-review.googlesource.com/c/chromium/src/+/4643591 Note that this does not follow the route taken by the W3C API but still considers this flag a part of the encodingParameters. BUG=chromium:1354101 Change-Id: If0f0ec09bebddea1f01dd8afbe4747c21afe6793 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286741 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40656} 2 年前
Change SetLocalContent in channel classes to avoid Invoke. With these changes, we now often have 0 invokes and at most 1 when calling SetLocalContent on a channel. Before we had at least 1 and typically 2. Summary of changes. * Updating RtpExtension::DeduplicateHeaderExtensions to return a sorted vector (+test) for easy detection of changes. * Before updating the transport on the network thread, detect if actual changes to the demuxer criteria or changes to the rtp header extensions have been made. * Consolidate both transport updates to a single call instead of two. * Added DCHECK guards to catch regressions in number of invokes. A possible upcoming improvement is to update the transport asynchronously. Bug: webrtc:13536 Change-Id: I71ef7b181635a796ffa1e3a02a0f661d28a4870c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244700 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35638} 4 年前
Insert frame transformer between Depacketizer and Decoder. Add a new API in RTReceiverInterface, to be called from the browser side to insert a frame transformer between the Depacketizer and the Decoder. The frame transformer is passed from RTReceiverInterface through the library to be eventually set in RtpVideoStreamReceiver, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169130. This change is part of the implementation of the Insertable Streams Web API: https://github.com/alvestrand/webrtc-media-streams/blob/master/explainer.md Design doc for WebRTC library changes: http://doc/1eiLkjNUkRy2FssCPLUp6eH08BZuXXoHfbbBP1ZN7EVk Bug: webrtc:11380 Change-Id: I6b73cd16e3907e8b7709b852d6a2540ee11b4fed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169129 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30654} 6 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
pc: Add asynchronous RtpSender::SetParameters() call As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627} 3 年前
Remove public GenerateKeyFrame(list-of-rids) API from RtpSender since the spec and implementation took a different route BUG=chromium:1354101 Change-Id: I6beda0db89b9e771ad2a7b51ba739bc46e18a331 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318200 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40665} 2 年前
RtpTransceiverInterface: add header_extensions_to_offer() This change adds exposure of a new transceiver method for getting the total set of supported extensions stored as an attribute, and their direction. If the direction is kStopped, the extension is not signalled in Unified Plan SDP negotiation. Note: SDP negotiation is not modified by this change. Changes: - RtpHeaderExtensionCapability gets a new RtpTransceiverDirection, indicating either kStopped (extension available but not signalled), or other (extension signalled). - RtpTransceiver gets the new method as described above. The default value of the attribute comes from the voice and video engines as before. https://chromestatus.com/feature/5680189201711104. go/rtp-header-extension-ip Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk Bug: chromium:1051821 Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30800} 6 年前
Reland "Remove some default implementations in api/rtp_transcever_interface" This reverts commit 226a2e32d03df3a2cf4bf3c616ac00dcd91ed3d2. Reason for revert: Downstream fixed (will submit when true) Original change's description: > Revert "Remove some default implementations in api/rtp_transcever_interface" > > This reverts commit 40941ee72d30676296f3545004e7a6e30b959612. > > Reason for revert: breaks downstream project > > Original change's description: > > Remove some default implementations in api/rtp_transcever_interface > > > > Bug: webrtc:11839 > > Change-Id: I6ddc0468e75bc346e12fc3dc64236ca2ab52e708 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244504 > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#35701} > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:11839 > Change-Id: I8a3eb0a279b28ed8b55745af97596c4a853669be > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247186 > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35713} # Not skipping CQ checks because this is a reland. Bug: webrtc:11839 Change-Id: Ie25f1a5fdb4ef8ebf200780755a69dc09dd28ccb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247189 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35737} 4 年前
sdp: reject spec simulcast answers without the rid extension which is mandatory to implement per https://datatracker.ietf.org/doc/html/rfc8853#section-5.5 BUG=chromium:1422258 Change-Id: I3639b15453aaa074fbe9f26b722f5997b439224a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306661 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40208} 2 年前
Delete implicit conversion from rtc::scoped_refptr<T> to T* Bug: webrtc:13464 Change-Id: I24c742c11a4ea5c4e307e170ee4fbd4e81cf1814 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260325 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36808} 3 年前
README.md

How to write code in the api/ directory

Mostly, just follow the regular style guide, but:

  • Note that api/ code is not exempt from the “.h and .cc files come in pairs” rule, so if you declare something in api/path/to/foo.h, it should be defined in api/path/to/foo.cc.
  • Headers in api/ should, if possible, not #include headers outside api/. It’s not always possible to avoid this, but be aware that it adds to a small mountain of technical debt that we’re trying to shrink.
  • .cc files in api/, on the other hand, are free to #include headers outside api/.
  • Avoid structs in api, prefer classes.

The preferred way for api/ code to access non-api/ code is to call it from a .cc file, so that users of our API headers won’t transitively #include non-public headers.

For headers in api/ that need to refer to non-public types, forward declarations are often a lesser evil than including non-public header files. The usual rules still apply, though.

.cc files in api/ should preferably be kept reasonably small. If a substantial implementation is needed, consider putting it with our non-public code, and just call it from the api/ .cc file.

Avoid defining api with structs as it makes harder for the api to evolve. Your struct may gain invariant, or change how it represents data. Evolving struct from the api is particular challenging as it is designed to be used in other code bases and thus needs to be updated independetly from its usage. Class with accessors and setters makes such migration safer. See Google C++ style guide for more.

If you need to evolve existent struct in api, prefer first to convert it into a class.