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2020-03-20Extend NetworkRoute with more info about local/remote endpointsJonas Oreland
This patch extends the NetworkRoute struct with more information about local/remote endpoints. It adds - adapter type - adapter id - relay (previously it was "only" network_id) The patch leaves the {local/remote}_network_id fields around and populated since downstream projects depend on them. They will be removed once they have migrated. OWNER: srte@ call/ test/ OWNER: asapersson@ video/ OWNER: hta@ p2p/ pc/ rtc_base/ BUG: webrtc:11434 Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30848}
2020-03-18Improve rollback for rtp data channelEldar Rello
Bug: chromium:1057333 Change-Id: I4df21bc183a8df398033ebf29a8407bacf873fac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170621 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Eldar Rello <elrello@microsoft.com> Cr-Commit-Position: refs/heads/master@{#30824}
2020-03-16Revert "remove mslabel and mslabel ssrc-specific attributes"Artem Titov
This reverts commit e3f257c4ee2079dee14ec8425eec691db3a9757c. Reason for revert: Breaks downstream projects Original change's description: > remove mslabel and mslabel ssrc-specific attributes > > Removes support for parsing and serializing > a=ssrc:1 mslabel:stream > a=ssrc:1 label:track > which have been superceeded by > a=ssrc:1 msid:stream track > a long time ago. > > Bug: webrtc:7110 > Change-Id: I3aca47728098b6e7e049b82ed34c59426d411c41 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168244 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30801} TBR=kthelgason@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com Change-Id: Ibd0ad11d2dee9f54bacab3dcca61dedccfc2c120 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7110 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170620 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30805}
2020-03-16remove mslabel and mslabel ssrc-specific attributesPhilipp Hancke
Removes support for parsing and serializing a=ssrc:1 mslabel:stream a=ssrc:1 label:track which have been superceeded by a=ssrc:1 msid:stream track a long time ago. Bug: webrtc:7110 Change-Id: I3aca47728098b6e7e049b82ed34c59426d411c41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168244 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30801}
2020-03-16RtpTransceiverInterface: add header_extensions_to_offer()Markus Handell
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}
2020-03-13Remove cricket::SessionDescription::Copy()Harald Alvestrand
To be submitted on or after March 13, 2020 (2 weeks after PSA). Bug: webrtc:10701 Change-Id: Ie4b6d31e1496b81714fe9f9418694fc4c2e69ecd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169443 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30784}
2020-03-11Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats APIArtem Titov
Bug: webrtc:11381 Change-Id: I7df3450e50da49d178e1e3a5d9f4970672d91aac Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169120 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30758}
2020-03-11RtpTransceiver: add kStopped enumeration value.Markus Handell
This change introduces a new kStopped enumeration value to RtpTransceiverDirection, preparing for later CLs which implement RTP header extension control, https://chromestatus.com/feature/5680189201711104. The new enumeration value is unused in the code. Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk Bug: chromium:980879 Change-Id: Id8cab9891236884542689fbf1b300e64a2cb636d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170050 Commit-Queue: Markus Handell <handellm@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30756}
2020-03-10Allow to negotiate absolute capture time rtp header extension.Minyue Li
Bug: webrtc:10739 Change-Id: I239d67a8c02bcc4175b142174b254e876bdd8d6d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169920 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30746}
2020-03-05Adds field trial to separate audio and video packets for delay-based overuse ↵Björn Terelius
detection. The decision to route audio packets to a separate overuse detector is off by default and requires the field trial WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/ The parameters control the threshold for switching over to the audio overuse detector if we stop receiving feedback for video. Bug: webrtc:10932 Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30694}
2020-03-05Implement new specification for degradation preferenceFlorent Castelli
The degradation preference is now based on the content hint of the track if it's unspecified. Bug: webrtc:11164 Change-Id: Iaa0dbf1c1bf68a46fc5131e534d423c30c5439c7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161233 Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30691}
2020-03-05Change network_priority from a double to an enum.Taylor Brandstetter
It can only be one of four possible values, so it never made sense for it to be a double. Other than the fact that its neighbor bitrate_priority is a double, and they're both defined as the same enum in the web spec. However, while bitrate_priority being a double offers more flexibility than the web spec, network_priority being a double is only confusing. Bug: webrtc:5658 Change-Id: I0784c116f3260c4b3a8b99a3cd85c8d66017e46f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168840 Reviewed-by: Anders Carlsson <andersc@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30685}
2020-03-03Unbreak ICE renominationJonas Oreland
This patch fixes a problem in https://webrtc.googlesource.com/src/+/71ff07369837d6575c04ebff7002d07d6e0af25f that when adding standard compliance validation of ufrag/pwd accidentally broken ice renomination by introducing a new "constructor". Bug: chromium:1044521 Change-Id: If1b18b1d728e55db9da385b37162a9cb5e61ac48 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169549 Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30670}
2020-03-03Fix Chromium Roll failing because of -Wrange-loop-constructCourtney Edwards
Bug: webrtc:11398 Change-Id: I51f6f9968b3a94b5fec325e8b5d29fd2bb290ee1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169553 Commit-Queue: Courtney Edwards <courtneyfe@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30669}
2020-03-02Reland "Expose can_trickle_ice_candidates on PeerConnection"Harald Alvestrand
This reverts commit cb8c40138ca170f841bc45fa6771cdfc4b966e5f. Reason for revert: Added missing default. Original change's description: > Revert "Expose can_trickle_ice_candidates on PeerConnection" > > This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9. > > Reason for revert: Breaks downstream due to missing default > > Original change's description: > > Expose can_trickle_ice_candidates on PeerConnection > > > > Bug: chromium:708484 > > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450 > > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Taylor <deadbeef@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30653} > > TBR=deadbeef@webrtc.org,hta@webrtc.org > > Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:708484 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30655} TBR=deadbeef@webrtc.org,hta@webrtc.org Change-Id: I608da7781f158b4b02dd226d4dcd5615c4935fa8 Bug: chromium:708484 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169541 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30656}
2020-03-02Revert "Expose can_trickle_ice_candidates on PeerConnection"Harald Alvestrand
This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9. Reason for revert: Breaks downstream due to missing default Original change's description: > Expose can_trickle_ice_candidates on PeerConnection > > Bug: chromium:708484 > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450 > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Taylor <deadbeef@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30653} TBR=deadbeef@webrtc.org,hta@webrtc.org Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: chromium:708484 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30655}
2020-03-02Insert frame transformer between Depacketizer and Decoder.Marina Ciocea
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}
2020-03-02Expose can_trickle_ice_candidates on PeerConnectionHarald Alvestrand
Bug: chromium:708484 Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Taylor <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30653}
2020-02-28Deprecate use of cricket::MediaContentDescription::CopyHarald Alvestrand
One should use a std::unique_ptr to the object, as returned by Clone() instead, not a naked pointer. Bug: webrtc:10701 Change-Id: I10ab309207f2cb5aec83a6d09336699ed7b26f50 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169342 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30646}
2020-02-28Allow to negotiate dependency descriptor rtp header extensionDanil Chapovalov
Without exposing it in capabilities: this extension is not stable enough to expose it by default, but already in working state so with munge sdp can be experimented with. Bug: webrtc:10342 Change-Id: I6bac123325a90431e4769e86da79638869e36cfc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168961 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30645}
2020-02-28Insert frame transformer between Encoded and Packetizer.Marina Ciocea
Add a new API in RTPSenderInterface, to be called from the browser side to insert a frame transformer between the Encoded and the Packetizer. The frame transformer is passed from RTPSenderInterface through the library to be eventually set in RTPSenderVideo, where the frame transformation will occur in the follow-up CL https://webrtc-review.googlesource.com/c/src/+/169128. Insertable Streams Web API explainer: 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: I46cd0d8a798c2736c837e90cbf90d8901c7d27fb Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169127 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30642}
2020-02-27Remove deprecated "description" field of cricket::ContentInfoHarald Alvestrand
Bug: webrtc:10701 Change-Id: I631616fefb59b49603e0a98267b3e58d93edfb50 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169122 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30628}
2020-02-26Remove deprecated AddContent function in session_description.hHarald Alvestrand
Bug: webrtc:10701 Change-Id: Ia1b8e5585c777d8f4c308bb8e4baffe752477057 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168950 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30615}
2020-02-25Adding deadbeef back to OWNERS filesTaylor Brandstetter
Specifically api, pc and p2p. Bug: None Change-Id: I2ba19aaac5ca11a5282593f0db06bba326fe6891 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169041 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Tommi <tommi@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30609}
2020-02-20Rollback transport created by data channelEldar Rello
No-Try: True Bug: chromium:1032987 Change-Id: I2c0dbd6a19e71a391dc2e0d30676d4efa26a9525 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168306 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30561}
2020-02-19Remove wildcard ownership for build files.Mirko Bonadei
No-Try: True Bug: webrtc:10381 Change-Id: I852d9a2da7e0c5c12f508a1c788b0b5753503aba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168769 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30558}
2020-02-19Remove inactive OWNERS.Mirko Bonadei
No-Try: True Bug: webrtc:10381 Change-Id: I3b56c74d913a47e4297518005b0cb19de8fafbff Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168421 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30556}
2020-02-18Make CNAME optional.Taylor Brandstetter
Before this change, lack of a CNAME results in losing all SSRC information. This isn't necessary; we don't even use the CNAME for anything on the receiving side. Note that lack of a CNAME is technically a violation of https://tools.ietf.org/html/rfc5576#section-6.1. Bug: webrtc:10385 Change-Id: If9836b6c518367b29ffa1fb00752e52d51915d37 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168581 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30552}
2020-02-18Replace DataSize and DataRate factories with newer versionsDanil Chapovalov
This is search and replace change: find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g" git cl format Bug: webrtc:9709 Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30545}
2020-02-18Ship GenericDescriptor00 by default.Markus Handell
The change ships GenericDescriptor00 and authentication by default, but doesn't expose it by default, and makes WebRTC respond to offers carrying it. The change adds a unit test for the new semantics. Tests well in munge-sdp. Frame marking replaced by http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00 in the offer results in an answer containing the extension as first entry. Bug: webrtc:11367 Change-Id: I0ef91b7d4096d949c3d547ece7d6c4d39aa241da Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168661 Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30542}
2020-02-13Populate sdp_fmtp_line and channels of RTCCodecStatsJohannes Kron
Change RtpCodecCapability::parameters and RtpCodecParameters::parameters to map from unordered_map to get welldefined FMTP lines. Bug: webrtc:7061 Change-Id: Ie61f76bbab915d72369e36e3f40ea11838827940 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168190 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30512}
2020-02-12Sort threading for sctp_mid_ variableHarald Alvestrand
Split the sctp_mid_ variable into two variables, sctp_mid_n_ and sctp_mid_s_, each of which is only accessed by one thread. Bug: webrtc:9987 Change-Id: I4dce944b920f4698e2606a7b85776791cbf55c28 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168243 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30503}
2020-02-10Removing myself from OWNERS in webrtc.Seth Hampson
No-Try: True Bug: None Change-Id: I632d5384321c88202a23cc3fa6938afac0f796ba Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168460 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30494}
2020-02-10Use newer version of TimeDelta and TimeStamp factories in webrtcDanil Chapovalov
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-07Revert "Reland "Reland "Reland "Distinguish between send and receive codecs""""Johannes Kron
This reverts commit 184ea66aed43161f05d80fbb74183a2efccca352. Reason for revert: Breaks downstream projects. TBR=steveanton@webrtc.org Original change's description: > Reland "Reland "Reland "Distinguish between send and receive codecs""" > > This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5. > > Reason for revert: Keep logic as is. > > Original change's description: > > Revert "Reland "Reland "Distinguish between send and receive codecs""" > > > > This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe. > > > > Reason for revert: Breaks perf test on iOS. > > > > Original change's description: > > > Reland "Reland "Distinguish between send and receive codecs"" > > > > > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2. > > > > > > Reason for revert: Flaky test in Chromium fixed. > > > > > > Original change's description: > > > > Revert "Reland "Distinguish between send and receive codecs"" > > > > > > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f. > > > > > > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium. > > > > > > > > Original change's description: > > > > > Reland "Distinguish between send and receive codecs" > > > > > > > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. > > > > > > > > > > Reason for revert: Fixed negotiation of send-only clients. > > > > > > > > > > Original change's description: > > > > > > Revert "Distinguish between send and receive codecs" > > > > > > > > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > > > > > > > > > > > Reason for revert: breaks negotiation with send-only clients > > > > > > > > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > > > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > > > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > > > > > > > > > > > Original change's description: > > > > > > > Distinguish between send and receive codecs > > > > > > > > > > > > > > Even though send and receive codecs may be the same, they might have > > > > > > > different support in HW. Distinguish between send and receive codecs > > > > > > > to be able to keep track of which codecs have HW support. > > > > > > > > > > > > > > Bug: chromium:1029737 > > > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > > > Cr-Commit-Position: refs/heads/master@{#30284} > > > > > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > > > > > > No-Presubmit: true > > > > > > No-Tree-Checks: true > > > > > > No-Try: true > > > > > > Bug: chromium:1029737 > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > > > > > Cr-Commit-Position: refs/heads/master@{#30292} > > > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > > > > > > > Bug: chromium:1029737 > > > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#30348} > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f > > > > No-Presubmit: true > > > > No-Tree-Checks: true > > > > No-Try: true > > > > Bug: chromium:1029737 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205 > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#30360} > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3 > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: chromium:1029737 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206 > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30367} > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: chromium:1029737 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30373} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > Bug: chromium:1029737 > Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30415} TBR=steveanton@webrtc.org,kron@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: chromium:1029737 Change-Id: Ice25339e7dfb9fc75049bd207d097b0910bd4446 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168341 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30484}
2020-02-05Don't crash when renegotiating after the peer rejects data channelsSteve Anton
Bug: webrtc:11320 Change-Id: I5a58d550574a4e0702fc6f05b7fb663fbc23d0b4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168200 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30463}
2020-02-05Cleanup: remove unused sctp_content_nameHarald Alvestrand
This accessor seems to be unused, and has a name that we don't want to support ("content_name"). Bug: none Change-Id: I2f332176429dd8e1895f821d30e4beaaa4650ec2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168195 Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30460}
2020-02-05disallow pairing ICE-TCP with a local ip addressPhilipp Hancke
BUG=chromium:1038754 Change-Id: Iab7186efd39a94bffde19e0c39a49f6bc61802ec Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167060 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30457}
2020-01-30Do not transition ICE gathering state to 'complete' when closingSteve Anton
Bug: webrtc:4728 Change-Id: I6bcb3dd0eb47dc945d96555f9481146f22ceb4fa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167440 Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30433}
2020-01-29Reland "Reland "Reland "Distinguish between send and receive codecs"""Johannes Kron
This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5. Reason for revert: Keep logic as is. Original change's description: > Revert "Reland "Reland "Distinguish between send and receive codecs""" > > This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe. > > Reason for revert: Breaks perf test on iOS. > > Original change's description: > > Reland "Reland "Distinguish between send and receive codecs"" > > > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2. > > > > Reason for revert: Flaky test in Chromium fixed. > > > > Original change's description: > > > Revert "Reland "Distinguish between send and receive codecs"" > > > > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f. > > > > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium. > > > > > > Original change's description: > > > > Reland "Distinguish between send and receive codecs" > > > > > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. > > > > > > > > Reason for revert: Fixed negotiation of send-only clients. > > > > > > > > Original change's description: > > > > > Revert "Distinguish between send and receive codecs" > > > > > > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > > > > > > > > > Reason for revert: breaks negotiation with send-only clients > > > > > > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > > > > > > > > > Original change's description: > > > > > > Distinguish between send and receive codecs > > > > > > > > > > > > Even though send and receive codecs may be the same, they might have > > > > > > different support in HW. Distinguish between send and receive codecs > > > > > > to be able to keep track of which codecs have HW support. > > > > > > > > > > > > Bug: chromium:1029737 > > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > > Cr-Commit-Position: refs/heads/master@{#30284} > > > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > > > > > No-Presubmit: true > > > > > No-Tree-Checks: true > > > > > No-Try: true > > > > > Bug: chromium:1029737 > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#30292} > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > > > > Bug: chromium:1029737 > > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 > > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#30348} > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: chromium:1029737 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205 > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30360} > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: chromium:1029737 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30367} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1029737 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364 > Commit-Queue: Johannes Kron <kron@webrtc.org> > Reviewed-by: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30373} TBR=steveanton@webrtc.org,kron@webrtc.org Bug: chromium:1029737 Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30415}
2020-01-29Allow non-identical datagram transport parameters.Bjorn A Mellem
Currently, datagram transports must report identical transport parameters in order to negotiate use of the datagram transport. This is not strictly necessary, they just need parameters that fit some notion of "compatability" (eg. both ends share some mutually-supported version of the datagram protocol). This change allows datagram transports to implement their own notion of compatible transport parameters, by adding a SetRemoteTransportParameters method to DatagramTransportInterface which checks if the remote parameters are compatible with the local endpoint and returns an error if they are not. Bug: webrtc:9719 Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30412}
2020-01-29Adding absolute capture timestamp to AudioTrackSinkInterface.Minyue Li
Bug: webrtc:10739 Change-Id: I8c134cbe82452ac71625cd0c810c783a73f17822 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167532 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30408}
2020-01-28Remove iceRegatherIntervalRangeSteve Anton
This was an ICE configuration experiment added a couple years ago that did not end up being used. Bug: webrtc:11316 Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30395}
2020-01-28[Stats] Include fecPackets[Reeceived/Discarded] in Members()Henrik Boström
This refers to modern getStats() only. The metrics has been implemented for a while in C++ but was accidentally not included in the Members() list, meaning they were not exposed in lists (including exposure in Chrome/JavaScript). The Chromium whitelist already include them. TBR=hta@webrtc.org Bug: webrtc:11317 Change-Id: I0c3ee9c552975fc37db2d87196c66e662c994aed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167530 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30391}
2020-01-25Validate ICE ufrag/pwd according to the specSteve Anton
https://tools.ietf.org/html/draft-ietf-mmusic-ice-sip-sdp-39#section-5.4 Bug: chromium:1044521 Change-Id: Ia95718437dfc270b52cdf822e861a3da7cbbab76 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167281 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30375}
2020-01-24Revert "Reland "Reland "Distinguish between send and receive codecs"""Johannes Kron
This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe. Reason for revert: Breaks perf test on iOS. Original change's description: > Reland "Reland "Distinguish between send and receive codecs"" > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2. > > Reason for revert: Flaky test in Chromium fixed. > > Original change's description: > > Revert "Reland "Distinguish between send and receive codecs"" > > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f. > > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium. > > > > Original change's description: > > > Reland "Distinguish between send and receive codecs" > > > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. > > > > > > Reason for revert: Fixed negotiation of send-only clients. > > > > > > Original change's description: > > > > Revert "Distinguish between send and receive codecs" > > > > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > > > > > > > Reason for revert: breaks negotiation with send-only clients > > > > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > > > > > > > Original change's description: > > > > > Distinguish between send and receive codecs > > > > > > > > > > Even though send and receive codecs may be the same, they might have > > > > > different support in HW. Distinguish between send and receive codecs > > > > > to be able to keep track of which codecs have HW support. > > > > > > > > > > Bug: chromium:1029737 > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > > Cr-Commit-Position: refs/heads/master@{#30284} > > > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > > > > No-Presubmit: true > > > > No-Tree-Checks: true > > > > No-Try: true > > > > Bug: chromium:1029737 > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#30292} > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > > > > Bug: chromium:1029737 > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 > > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30348} > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: chromium:1029737 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30360} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1029737 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30367} TBR=steveanton@webrtc.org,kron@webrtc.org Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: chromium:1029737 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364 Commit-Queue: Johannes Kron <kron@webrtc.org> Reviewed-by: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30373}
2020-01-23Reland "Reland "Distinguish between send and receive codecs""Johannes Kron
This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2. Reason for revert: Flaky test in Chromium fixed. Original change's description: > Revert "Reland "Distinguish between send and receive codecs"" > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f. > > Reason for revert: Breaks Chromium import due to flaky test in Chromium. > > Original change's description: > > Reland "Distinguish between send and receive codecs" > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. > > > > Reason for revert: Fixed negotiation of send-only clients. > > > > Original change's description: > > > Revert "Distinguish between send and receive codecs" > > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > > > > > Reason for revert: breaks negotiation with send-only clients > > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > > > > > Original change's description: > > > > Distinguish between send and receive codecs > > > > > > > > Even though send and receive codecs may be the same, they might have > > > > different support in HW. Distinguish between send and receive codecs > > > > to be able to keep track of which codecs have HW support. > > > > > > > > Bug: chromium:1029737 > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#30284} > > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > > > No-Presubmit: true > > > No-Tree-Checks: true > > > No-Try: true > > > Bug: chromium:1029737 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30292} > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > > > Bug: chromium:1029737 > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30348} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1029737 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30360} TBR=steveanton@webrtc.org,kron@webrtc.org Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: chromium:1029737 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30367}
2020-01-23Revert "Reland "Distinguish between send and receive codecs""Johannes Kron
This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f. Reason for revert: Breaks Chromium import due to flaky test in Chromium. Original change's description: > Reland "Distinguish between send and receive codecs" > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. > > Reason for revert: Fixed negotiation of send-only clients. > > Original change's description: > > Revert "Distinguish between send and receive codecs" > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > > > Reason for revert: breaks negotiation with send-only clients > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > > > Original change's description: > > > Distinguish between send and receive codecs > > > > > > Even though send and receive codecs may be the same, they might have > > > different support in HW. Distinguish between send and receive codecs > > > to be able to keep track of which codecs have HW support. > > > > > > Bug: chromium:1029737 > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#30284} > > > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: chromium:1029737 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Commit-Queue: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30292} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > > Bug: chromium:1029737 > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30348} TBR=steveanton@webrtc.org,kron@webrtc.org Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: chromium:1029737 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30360}
2020-01-22Allow DTMF delay configurabilityAaron Alaniz
This commit enables developers to configure the "," delay value from the WebRTC spec value of 2 seconds. This flexibility allows developers to comply with existing WebRTC clients. Bug: webrtc:11273 Change-Id: Ia94b99e041df882e2396d0926a8f4188afe55885 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165700 Commit-Queue: Steve Anton <steveanton@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30354}
2020-01-22Reland "Distinguish between send and receive codecs"Johannes Kron
This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8. Reason for revert: Fixed negotiation of send-only clients. Original change's description: > Revert "Distinguish between send and receive codecs" > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d. > > Reason for revert: breaks negotiation with send-only clients > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264] > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER) > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters. > > Original change's description: > > Distinguish between send and receive codecs > > > > Even though send and receive codecs may be the same, they might have > > different support in HW. Distinguish between send and receive codecs > > to be able to keep track of which codecs have HW support. > > > > Bug: chromium:1029737 > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Steve Anton <steveanton@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#30284} > > TBR=steveanton@webrtc.org,kron@webrtc.org > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1029737 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30292} TBR=steveanton@webrtc.org,kron@webrtc.org Bug: chromium:1029737 Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30348}
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