68f542bc创建于 3月16日历史提交
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Document how bitrate probing works from a RTP perspective BUG=webrtc:15182 No-Try: true Change-Id: I8e669650ae0ce2e7434420f7e8ff18aee714ed06 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306962 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40253} 2 年前
Update pc/ to not use implicit conversion from scoped_refptr<T> to T*. Bug: webrtc:13464 Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36588} 4 年前
Move Destroy/Create steps for DataChannelTransport to PeerConnection. This moves steps from the sdp code for pc state over to the PC class and slightly simplifies the contract between the two classes. Moving forward it's easier to consolidate those steps in the PC class with other grouped operations e.g. during teardown. Also removing GetDataMid() method in favor of the sctp_mid() property. Bug: none Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40981} 2 年前
Makes sure that RED is not added twice to the list of codecs when it is used with Opus. Bug: webrtc:15606 Change-Id: I3ab3ee287f5d2e3a0a46520608e5c0931e0bff90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325180 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#41028} 2 年前
Remove the SctpDataChannel::config_ member variable. Instead there are direct member variables for the various relevant states, some weren't needed, some can be const but the id member in particular needs special handling and can't be const. For dealing with the stream id, we now have SctpSid. A class that does range validation, checks thread safety, handles the special -1 case (for what's essentially an unsigned 16 bit int). Using a special type for this also has the effect that range checking happens more consistently (although I'm not modifying the structs in api/). With upcoming steps of avoiding thread hops, the ID may need to migrate to the network thread, which the thread checks will help with. Along the way, update SctpSidAllocator to use flat_set instead of std::set and moving some of the sctp data channel code to the cc file to help with more accurately tracking code coverage. Bug: webrtc:11547 Change-Id: Iea6e7647ab8f93052044c5afbcc449115206b4e9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296444 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39539} 3 年前
Enforce policy that SDP munging requires special approval This ensures that adding features by SDP munging gets a review by people who understand how this works in the community. Bug: none Change-Id: I36feb0e3c7896d4f7bec81078109d7914c349a0d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291339 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39229} 3 年前
Rename Set(Send|Recv)Parameters Set(Sender|Receiver)Parameters following the previous change to rename the classes derived from cricket::RtpParameters Also rename ChangedRecvParameters to ChangedReceiveParameters. BUG=webrtc:13931 Change-Id: Ia51dd39905a5cbb98162c3948930e43ccaf3786d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314500 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40677} 2 年前
Implement GetParameters/GetSources support for unsignaled SSRCs. Unsignaled SSRCs are only applicable for the receiver case (not sender). This CL updates the receievr's GetParameters() and GetSources() methods to lookup parameters/sources by the current SSRC (whether or not it was signaled) instead of only looking at the signaled SSRC. To clarify that the ssrc_ variable inside the [Audio/Video]RtpReceiver is the signaled ssrc (and not set if the current ssrc is unsignaled), we rename this variable to signaled_ssrc_. By the looks of it, other APIs like setting volume or packetizers also have a dependency on the assumptions that the SSRC is signaled. We will not address that in this CL, but this CL makes that more clear. Bug: webrtc:14811 Change-Id: I32c93d264ab441ade23a4078639744d25b791742 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290573 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39051} 3 年前
Remove all split channels related code Bug: webrtc:13931 Change-Id: I93b8ca0ba1ec15bf260236bbc914b41fbb30aa58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40376} 2 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 3 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
Use webrtc::TaskQueueBase type instead of rtc::Thread ...for signaling and worker thread members in BaseChannel classes. Bug: webrtc:15099 Change-Id: I83611ed2564e143aca19d0f12ce060b77eb9d2a7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325260 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41041} 2 年前
Use webrtc::TaskQueueBase type instead of rtc::Thread ...for signaling and worker thread members in BaseChannel classes. Bug: webrtc:15099 Change-Id: I83611ed2564e143aca19d0f12ce060b77eb9d2a7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325260 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41041} 2 年前
Remove all split channels related code Bug: webrtc:13931 Change-Id: I93b8ca0ba1ec15bf260236bbc914b41fbb30aa58 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310680 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Jeremy Leconte <jleconte@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40376} 2 年前
Reduce usage of audio/video codec specifics BUG=webrtc:15214 Change-Id: I8e68ac149af53529321ab44776c62afe4cc2f61e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324020 Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40960} 2 年前
Reland "ConnectionContext: remove media engine without blocking." This reverts commit 2d71807fe09aad67efcd660fe286044ff10982ba. Reason for revert: With the new AsyncAudioProcessing API, the issue that was introduced can now be worked around. Original change's description: > Revert "ConnectionContext: remove media engine without blocking." > > This reverts commit 2ba941e6bc1d20acb9cfda4b87ba53c80640bbcb. > > Reason for revert: Temporarily reverting due to b/269628432. > > Original change's description: > > ConnectionContext: remove media engine without blocking. > > > > Bug: webrtc:14449 > > Change-Id: I445114c14f4d440a5a8cac003266047fe4588dab > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288580 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Markus Handell <handellm@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38928} > > Bug: webrtc:14449 > Change-Id: If2f23662e486a1c1f85c318fc98c441aab9ace31 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295862 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Auto-Submit: Markus Handell <handellm@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39454} Bug: webrtc:14449 Change-Id: I43bb7a3b366eb60b3dc4b88dd9d47d570bb99bc2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311941 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40705} 2 年前
Reland "ConnectionContext: remove media engine without blocking." This reverts commit 2d71807fe09aad67efcd660fe286044ff10982ba. Reason for revert: With the new AsyncAudioProcessing API, the issue that was introduced can now be worked around. Original change's description: > Revert "ConnectionContext: remove media engine without blocking." > > This reverts commit 2ba941e6bc1d20acb9cfda4b87ba53c80640bbcb. > > Reason for revert: Temporarily reverting due to b/269628432. > > Original change's description: > > ConnectionContext: remove media engine without blocking. > > > > Bug: webrtc:14449 > > Change-Id: I445114c14f4d440a5a8cac003266047fe4588dab > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288580 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Markus Handell <handellm@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#38928} > > Bug: webrtc:14449 > Change-Id: If2f23662e486a1c1f85c318fc98c441aab9ace31 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295862 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Auto-Submit: Markus Handell <handellm@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39454} Bug: webrtc:14449 Change-Id: I43bb7a3b366eb60b3dc4b88dd9d47d570bb99bc2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311941 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40705} 2 年前
Avoid touching channel after OnSctpDataChannelClosed Bug: chromium:1454086 Change-Id: I39573b706c4031d091c45a182b13cb3b2dba6233 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309920 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40332} 2 年前
Refactor HasDataChannels Follow-up to: https://webrtc-review.googlesource.com/c/src/+/304241 This changes HasDataChannels() to not block on the network thread. Bug: chromium:1442604 Change-Id: I880e3ed554bc4265f675fb2aa48351a7f42ef9bb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304961 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40068} 2 年前
Attempt to recycle a stopped data m-line before creating a new one which avoids an infinitely growing SDP if the remote end rejects the datachannel section. This will reactivate the m-line even if all datachannels are closed. BUG=chromium:1442604 Change-Id: If60f93b406271163df692d96102baab701923602 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304241 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@{#40029} 3 年前
Reland "Don't create channel_manager++ when media_engine is not set" This reverts commit c6c02efb56b24df04ed9ab61252c14c7bddcca93. Reason for revert: Test now passes (and channel manager is gone) Original change's description: > Revert "Don't create channel_manager when media_engine is not set" > > This reverts commit c48ad732d6eb69f14dd6d44f801d62997cef2c2f. > > Reason for revert: breaks downstream project > > Original change's description: > > Don't create channel_manager when media_engine is not set > > > > Also remove a bunch of functions in ChannelManager that were just > > forwarding to MediaEngineInterface. > > > > Bug: webrtc:13931 > > Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304 > > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#36801} > > Bug: webrtc:13931 > Change-Id: I1e260a2489547bd9483b50e043c28d2805b0fa5a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261660 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Cr-Commit-Position: refs/heads/main@{#36811} Bug: webrtc:13931 Change-Id: I7b5b45b46095c18d489b6a9fe4c625971d6b3da6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261661 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36976} 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
Remove DataChannelType and deprecated option disable_sctp_data_channels Since there is only a single type of DataChannel now, the enum was only used when data channels were disabled at the PC API. That option has been deprecated 4 years ago, it's now time to remove it. Bug: webrtc:6625 Change-Id: I9e4ada1756da186e9639dd0fbf0249c55ea0b6c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215661 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33778} 5 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
Reduce logging verbosity of DTLS-SRTP RTCP transport since that transport is unset most of the time when rtcp-mux is used. BUG=None Change-Id: Ic1d732369c5544059112173af767488aed7ec8e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316926 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@{#40598} 2 年前
More unused sigslot includes This time, hit the BUILD files too (where possible). Bug: webrtc:11943 Change-Id: Ic8f2d77e1ba66f740efe0ef73b1ea6051356dedc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318100 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40654} 2 年前
Convert signals in rtp_transport_internal.h to CallbackList Bug: webrtc:11943 Change-Id: I8e0839363712d9d8b49c2f6cbdb5f3ac59d79219 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318882 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40700} 2 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 3 年前
Remove cricket::DtlsTransportState. Bug: webrtc:12762 Change-Id: I7a6535f7ce57b1d521364f3c62086377f5aebf57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218600 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34087} 4 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 3 年前
Replace Thread::Invoke with Thread::BlockingCall BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed Bug: webrtc:11318 Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38045} 3 年前
Replace Thread::Invoke with Thread::BlockingCall BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed Bug: webrtc:11318 Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38045} 3 年前
Remove sigslot usage from DtmfProviderInterface Bug: webrtc:11943 Change-Id: I452efbb099affc10e9197573fa0e40094a0d90ca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270420 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37681} 3 年前
Reland "Use backticks not vertical bars to denote variables in comments for /pc" Original change's description: > Revert "Use backticks not vertical bars to denote variables in comments for /pc" > > This reverts commit 37ee0f5e594dd772ec6d620b5e5ea8a751b684f0. > > Reason for revert: Revert in order to be able to revert https://webrtc-review.googlesource.com/c/src/+/225642 > > Original change's description: > > Use backticks not vertical bars to denote variables in comments for /pc > > > > Bug: webrtc:12338 > > Change-Id: I88cf10afa5fc810b95d2a585ab2e895dcc163b63 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226953 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#34575} > > TBR=hta@webrtc.org,titovartem@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I5eddd3a14e1f664bf831e5c294fbc4de5f6a88af > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12338 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227082 > Reviewed-by: Björn Terelius <terelius@webrtc.org> > Commit-Queue: Björn Terelius <terelius@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#34577} Bug: webrtc:12338 Change-Id: I96bd229b73613c162d11d75fa4f5934e1b4295c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227087 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34611} 4 年前
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174} 4 年前
Reject stun urls with query parameters for consistency with Chromium behavior, in particular stun:host?transport=udp gets rejected. BUG=chromium:1385735 Change-Id: I85a141ecf72480bfa09d8354d6dcaef8ca0cdcff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299943 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39916} 3 年前
Reland "ice server parsing: return RTCError with more details" This is a reland of commit c4b0bde7f2daabc4e0667fb73d096d1cf0819226 which changes the name of the new method and adds a deprecated backward compatible variant with the old name. Original change's description: > ice server parsing: return RTCError with more details > > surfacing those errors to the API (without specific details) instead of just the logging. > > BUG=webrtc:14539 > > Change-Id: Id907ebeb08b96b0e4225a016a37a12d58889091b > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278340 > 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@{#38356} Bug: webrtc:14539 Change-Id: I0a5482e123f25867582d62101b31ed207b95ea1a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278962 Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38364} 3 年前
Reject stun urls with query parameters for consistency with Chromium behavior, in particular stun:host?transport=udp gets rejected. BUG=chromium:1385735 Change-Id: I85a141ecf72480bfa09d8354d6dcaef8ca0cdcff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299943 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39916} 3 年前
Use SequenceChecker from public API Bug: webrtc:12419 Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33220} 5 年前
Break out a few more targets from rtc_pc_base Also use apply-iwyu -r to clean out some not-needed includes. Bug: webrtc:13805 Change-Id: Id12b6a0e340f686fdfbb9df6fedac324bdcc4b2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254680 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36174} 4 年前
Remove FakePortAllocator's dependency on ScopedKeyValueConfig. Breaking this dependency is required for using FakePortAllocator in chromium tests to make the windows component build work. Bug: chromium:1408420 Change-Id: I4215b92c1d1430156107605e5b054926b30f83f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291114 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Sameer Vijaykar <samvi@google.com> Cr-Commit-Position: refs/heads/main@{#39180} 3 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
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
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064} 4 年前
Update IWYU tool with a mapping file Also apply IWYU to all .cc files in pc/, and correct BUILD file to match. Note: Some files came out wrong when iwyu was applied. These are not included. Bug: none Change-Id: Ib5ea46b8fcc505414d0447cca7218ad3afc2e321 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252280 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36064} 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
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
Add histogram for DTLS peer signature algorithm in order to estimate the impact of deprecating SHA1. Chromium UMA CL: https://chromium-review.googlesource.com/c/chromium/src/+/4894345 BUG=webrtc:15517 Change-Id: I5216ba2a8cbba2f276af20d31aa5e111e7c3a141 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321620 Reviewed-by: David Benjamin <davidben@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40882} 2 年前
Remove sigslot usage from SctpTransportInternal Bug: webrtc:11943 Change-Id: I42edf8e2e15e580bcda090447a7aae4a56366b33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270661 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37867} 3 年前
Move channel creation functions into RtpTransceiver This breaks the link from sdp_offer_answer.cc to channel.h. Bug: webrtc:13931 Change-Id: I75608f75713bf4e69013ac5f5b17c19e53d07519 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36757} 3 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
Simplify PeerConnection::SetConfiguration * Consolidate ice candidate pool size checks (was in 3 places) * Consolidate ICE server configuration parsing (was in 2 locations) * Remove separate blocking call in PC for SetActiveResetSrtpParams(). * Remove unnecessary blocking call inside SetActiveResetSrtpParams implementation. Bug: none Change-Id: I38c8964f82f91c77c1fd18c407aefaab1d0c7c0d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324303 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40984} 2 年前
Ensure Call is notified of un demuxable packets With this cl, packets that are discarded in RtpTransport now notifies Call, so that they can be part of BWE even if they are dropped. These packets have been recevied on the transport, and has bin decrypted and parsed and thus can be accounted for. The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded. Bug: webrtc:14928 Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39727} 3 年前
Allow single-mline offers without BUNDLE group when using max-bundle since BUNDLE is not meaningful for those cases. This matches Firefox behavior. BUG=chromium:1444615 Change-Id: Id841b7e30a1c920efd977caebc71ab25d084577a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/305640 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@{#40151} 2 年前
Reland "dtls: allow dtls role to change during DTLS restart" This is a reland of commit 02b5f3c9c12cddf3fc6e9125238b77ddb44f3b53 without making SetRemoteFingerprint private (but adding a deprecation warning) Original change's description: > dtls: allow dtls role to change during DTLS restart > > which is characterized by a change in remote fingerprint and > causes a new DTLS handshake. This allows renegotiating the > client/server role as well. > Spec guidance is provided by > https://www.rfc-editor.org/rfc/rfc5763#section-6.6 > > BUG=webrtc:5768 > > Change-Id: I0e8630c0c5907cc92720762a4320ad21a6190d28 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271680 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Cr-Commit-Position: refs/heads/main@{#37821} Bug: webrtc:5768 Change-Id: I8dd674db8b683160013e1b4aa7776775d130978f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272221 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#37838} 3 年前
Remove default "unknown" encoderImplementation/decoderImplementation which means this will not show up in getStats inbound-rtp/outbound-rtp until the encoder/decoder is known. This has implications in particular for inbound-rtp where the value is currently "unknown" until video frames have been received. This is safe to change as the previous change to gate decoderImplementation behind getUserMedia access already broke the assumption that the field is always string. BUG=webrtc:14906 Change-Id: Ie6040ada3656e80f792c0c32c1b86ad1d6609d3c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293600 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40334} 2 年前
Reland "[DataChannel] Send and receive packets on the network thread." This reverts commit 7f16fcda0fd5bb625584b71311dd37b54c096136. Reason for reland: Re-landing after addressing issues in downstream code and hardening the ObserverAdapter from situations where attempted usage of data channel proxies could occur after shutting down the peer connection and terminating the network thread. Original change's description: > Revert "[DataChannel] Send and receive packets on the network thread." > > This reverts commit fe53fec24e02d2d644220f913c3f9ae596bbb2d9. > > Reason for revert: Speculative revert, may be breaking downstream project > > Original change's description: > > [DataChannel] Send and receive packets on the network thread. > > > > This updates sctp channels, including work that happens between the > > data channel controller and the transport, to run on the network > > thread. Previously all network traffic related to data channels was > > routed through the signaling thread before going to either the network > > thread or the caller's thread (e.g. js thread in chrome). Now the > > calls can go straight from the network thread to the JS thread with > > enabling a special flag on the observer (see below) and similarly > > calls to send data, involve 2 threads instead of 3. > > > > * Custom data channel observer adapter implementation that > > maintains compatibility with existing observer implementations in > > that notifications are delivered on the signaling thread. > > The adapter can be explicitly disabled for implementations that > > want to optimize the callback path and promise to not block the > > network thread. > > * Remove the signaling thread copy of data channels in the controller. > > * Remove several PostTask operations that were needed to keep things > > in sync (but the need has gone away). > > * Update tests for the controller to consistently call > > TeardownDataChannelTransport_n to match with production. > > * Update stats collectors (current and legacy) to fetch the data > > channel stats on the network thread where they're maintained. > > * Remove the AsyncChannelCloseTeardown test since the async teardown > > step has gone away. > > * Remove sid_s in the channel code since we only need the network > > state now. > > * For the custom observer support (with and without data adapter) and > > maintain compatibility with existing implementations, added a new > > proxy macro that allows an implementation to selectively provide > > its own implementation without being proxied. This is used for > > registering/unregistering a data channel observer. > > * Update the data channel proxy to map most methods to the network > > thread, avoiding the interim jump to the signaling thread. > > * Update a plethora of thread checkers from signaling to network. > > > > Bug: webrtc:11547 > > Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142 > > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > > Reviewed-by: Henrik Boström <hbos@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39760} > > Bug: webrtc:11547 > Change-Id: Id0d65594bf727ccea5c49093c942b09714d101ad > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300341 > Auto-Submit: Andrey Logvin <landrey@webrtc.org> > Owners-Override: Andrey Logvin <landrey@webrtc.org> > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39764} Bug: webrtc:11547 Change-Id: I47dfa7e7168be0cd2faab4f8f3ebf110c3728af5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300360 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39786} 3 年前
Delete api/stats_types.h in favor of api/legacy_stats_types.h The file was renamed, see https://groups.google.com/u/1/g/discuss-webrtc/c/ZQiP4f_bpw4 Bug: webrtc:14180 Change-Id: Ia76c85ba7d9da6b3a93d0a67a4b6a5187e07e230 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283084 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38616} 3 年前
Replace "rcvd" with "received" for readability following guidance in https://google.github.io/styleguide/cppguide.html#General_Naming_Rules BUG=None Change-Id: I105591a7f709d65a3d3320f7f44137d432cf7ce0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302282 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#39937} 3 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 3 年前
Remove rtc_base/scoped_ref_ptr.h. The type rtc::scoped_refptr<T> is now part of api/. Please include it from api/scoped_refptr.h. More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o. Bug: webrtc:9887, webrtc:8205 No-Try: True Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf Reviewed-on: https://webrtc-review.googlesource.com/c/119041 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26414}7 年前
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}7 年前
Never pass a signed char to ctype macros like isdigit() Bug: None Change-Id: I451bb2c1f175a77aefbc8363009bf35a769fe941 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264442 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37037} 3 年前
Cleanup: Move some more protocol names into media_protocol_names Bug: None Change-Id: I29ccee993ece01ffbafa85f09abb7cf64dba82d7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237020 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35295} 4 年前
Makes sure that RED is not added twice to the list of codecs when it is used with Opus. Bug: webrtc:15606 Change-Id: I3ab3ee287f5d2e3a0a46520608e5c0931e0bff90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325180 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#41028} 2 年前
MediaSession: ensure transport description factory exists BUG=None Change-Id: Ic29526c0c182257331d81ff3e66c5ae91ddf4ce1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321186 Reviewed-by: Florent Castelli <orphis@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#40826} 2 年前
Makes sure that RED is not added twice to the list of codecs when it is used with Opus. Bug: webrtc:15606 Change-Id: I3ab3ee287f5d2e3a0a46520608e5c0931e0bff90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325180 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#41028} 2 年前
Move rtc::make_ref_counted to api/ Bug: webrtc:12701 Change-Id: If49095b101c1a1763c2a44a0284c0d670cce953f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265390 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37219} 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
Update pc/ to not use implicit conversion from scoped_refptr<T> to T*. Bug: webrtc:13464 Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36588} 4 年前
Update pc/ to not use implicit conversion from scoped_refptr<T> to T*. Bug: webrtc:13464 Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36588} 4 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
Move VideoTrack's content_hint property to the signaling thread. Bug: webrtc:13673, webrtc:13681 Change-Id: I06810338bf5e44665e4d005d35636e9a98b1bd0b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251684 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36010} 4 年前
Delete TestListener and top-level thread wrapping. Instead use rtc::AutoThread in tests that need that. Bug: webrtc:9714 Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681 Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36950} 3 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
Move Destroy/Create steps for DataChannelTransport to PeerConnection. This moves steps from the sdp code for pc state over to the PC class and slightly simplifies the contract between the two classes. Moving forward it's easier to consolidate those steps in the PC class with other grouped operations e.g. during teardown. Also removing GetDataMid() method in favor of the sctp_mid() property. Bug: none Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40981} 2 年前
New PeerConnectionFactory::CreateVideoTrack with refcounted source Bug: webrtc:15017 Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39635} 3 年前
Generalize ssrc-group check to apply to groups other than SIM BUG=chromium:1477075 Change-Id: I20f094dee11ea26a180471ce52d78d916f922f29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322440 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40888} 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
Update pc/ to not use implicit conversion from scoped_refptr<T> to T*. Bug: webrtc:13464 Change-Id: I768646af8ded6338ef51486b8d69db1ad71e9a2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259500 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36588} 4 年前
Makes sure that RED is not added twice to the list of codecs when it is used with Opus. Bug: webrtc:15606 Change-Id: I3ab3ee287f5d2e3a0a46520608e5c0931e0bff90 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325180 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tomas Lundqvist <tomasl@google.com> Cr-Commit-Position: refs/heads/main@{#41028} 2 年前
Create SctpDataChannel objects on the network thread. * Change data channel creation code to return RTCError for more detailed/accurate errors. * Move DataChannelController::sid_allocator_ to the network thread. * Add a temporary duplicate vector of channels on the network thread. This will eventually be the main vector. * Delete one test that turns out to be racy (as long as we're using both the signaling and network threads). Bug: webrtc:11547, webrtc:12796 Change-Id: I93ab721a09872d075046a907df60e8aee4263371 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298624 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39719} 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
New PeerConnectionFactory::CreateVideoTrack with refcounted source Bug: webrtc:15017 Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39635} 3 年前
New PeerConnectionFactory::CreateVideoTrack with refcounted source Bug: webrtc:15017 Change-Id: I04c794d8959583bb4cc5c3898f4175783ec49f16 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249363 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39635} 3 年前
Check use_rtx() in PeerConnectionFactory::GetRtpSenderCapabilities Following https://webrtc-review.googlesource.com/c/src/+/262666, use_rtx() was checked in PeerConnectionFactory::GetRtpReceiverCapabilities but was missed in GetRtpSenderCapabilities. Therefore clients that hardcode use_rtx = false end up in an inconsistent state where RTX is not fully disabled. Bug: None Change-Id: Ice5f29a77c59e9081f9dd72c13c819024a34a7dd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316243 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40625} 2 年前
Reland: FrameGeneratorCapturer: don't generate video before Start is called It is partial reland, which adds call to Start() to all relevant places, but doesn't actually switches frame generator to not produce frames from the moment it was created. Bug: b/272350185 Change-Id: I6e3bd7af6f5cd8d9baff79c2aada7b2ddfae1c8d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310782 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40379} 2 年前
Only answer with non-stopped RTP header extensions This extends the RTP header extension API usage to generating answers. Also re-adds unit tests removed by the revert. BUG=chromium:1051821 Change-Id: Ib754284e9a77cb49e22bea7072c475d240f2563b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298740 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@{#39800} 3 年前
Remove another ctor from BasicPortAllocator This constructor isn't used in production. Removing it further made the construction state of the class simpler, allowed for removal of the separate Init() method and making more members const. Bug: none Change-Id: Ibc8516a01ce7e385207251d841d21bb7b72c9d9a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318281 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40678} 2 年前
Remove another ctor from BasicPortAllocator This constructor isn't used in production. Removing it further made the construction state of the class simpler, allowed for removal of the separate Init() method and making more members const. Bug: none Change-Id: Ibc8516a01ce7e385207251d841d21bb7b72c9d9a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318281 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40678} 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
Generalize ssrc-group check to apply to groups other than SIM BUG=chromium:1477075 Change-Id: I20f094dee11ea26a180471ce52d78d916f922f29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322440 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40888} 2 年前
Move Destroy/Create steps for DataChannelTransport to PeerConnection. This moves steps from the sdp code for pc state over to the PC class and slightly simplifies the contract between the two classes. Moving forward it's easier to consolidate those steps in the PC class with other grouped operations e.g. during teardown. Also removing GetDataMid() method in favor of the sctp_mid() property. Bug: none Change-Id: I938f953099d327377abd94e6b2c9ece803d88e40 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324300 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40981} 2 年前
sdp: accept bundle-only media section without rtcp-mux following the example C1 in https://www.rfc-editor.org/rfc/rfc8829.html#section-7.3 and the rules from https://www.rfc-editor.org/rfc/rfc8843.html#section-9.3.1.1 BUG=chromium:1444615 Change-Id: I6aedc5a669a9c53b9d65fb564804913203a453f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304980 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40058} 2 年前
Implement codec selection api The implementation covers the latest specification, but does not support mixed-codec simulcast at the moment. Changing codec for audio and video is supported. Bug: webrtc:15064 Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40616} 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
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
Add optional init_send_encodings to AddTrack This patch adds variant of PeerConnectionInterface::AddTrack that takes an initial_send_encodings. This allows for setting/modifying encoding parameters before sdp negotiation is performed/complete (e.g requested_resolution). This is already available if using RtpTransciverInit and AddTransceiver, but was not added to AddTrack because of concerns that it complicated matching with existing transceivers. This CL sidesteps that by never matching to a preexisting transceiver if initial_send_encodings are specified. Note: 1) The patch adds a new method rather than an extra (e.g optional) argument to existing AddTrack. This is to avoid problems with downstream mocks. 2) chromium "problems" was fixed in https://chromium-review.googlesource.com/c/chromium/src/+/3952684 and https://chromium-review.googlesource.com/c/chromium/src/+/3956060 Bug: webrtc:14451 Change-Id: I19b5a03872730280fbf868ca5d3a2f46443359f3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278783 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38437} 3 年前
Remove another ctor from BasicPortAllocator This constructor isn't used in production. Removing it further made the construction state of the class simpler, allowed for removal of the separate Init() method and making more members const. Bug: none Change-Id: Ibc8516a01ce7e385207251d841d21bb7b72c9d9a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318281 Reviewed-by: Per Kjellander <perkj@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40678} 2 年前
Generalize ssrc-group check to apply to groups other than SIM BUG=chromium:1477075 Change-Id: I20f094dee11ea26a180471ce52d78d916f922f29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322440 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40888} 2 年前