This reverts commit f79bfc65e52a35d27cf0db2d212e94043fb44da3.
Reason for revert: Potentially affects Chromium tests, see
failures on https://chromium-review.googlesource.com/c/chromium/src/+/2276338.
Original change's description:
> peerconnection: prefer spec names for signaling state
>
> Map the internal state names to the spec ones defined in
> https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum
> instead of exposing them. This only affects the (not specified)
> error strings.
>
> Bug: None
> Change-Id: Ib0b35bb3106b1688e8386f6fdd0b8c7fdebaf1dc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178390
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#31591}
TBR=hbos@webrtc.org,philipp.hancke@googlemail.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: None
Change-Id: I6df20c93f6944b819eb11f22ba30c6221de61d79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31610}
some codecs like RED and telephone-event have fmtp lines which
do not conform to the list-of-key=value convention. Add support
for parsing and serializing this by setting the name to the empty
string.
BUG=webrtc:11640
Change-Id: Ie3ef7c98f756940f97d27a39af0574aa37949f74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178120
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31609}
Policy will allow explicitly specify thread between which invokes are
allowed, or explicitly forbid any invokes.
Change-Id: I360e7cba3ce1c21abd5047c6f175d8c4e0e99c6f
Bug: webrtc:11728
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177526
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31604}
Added some checks to lock down setting the sctp id and in general
start clarifying logic that belongs to RTP data channels and
which ones belong to "SCTP-like" instances.
Change-Id: Ibfa63fde3e845d743f148e2d3c0927d0cc02913d
Bug: webrtc:11547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178221
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31593}
Map the internal state names to the spec ones defined in
https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum
instead of exposing them. This only affects the (not specified)
error strings.
Bug: None
Change-Id: Ib0b35bb3106b1688e8386f6fdd0b8c7fdebaf1dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178390
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31591}
Removes an old workaround for a firefox issue that was fixed in 2014.
BUG=webrtc:3212
Change-Id: I3ad71e29249908d840474cf3ee99a094c9201f6d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178381
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31584}
It was used only to provide parameters for MediaTransport.
Bug: webrtc:9719
Change-Id: I42e451ef84251ecf2b15010c7a3923b6fa2436be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177350
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31541}
This removes code from DataChannelController that exposes
an internal vector of data channels and puts the onus of
returning stats for a data channel, on the data channel
object itself. This will come in handy as we make threading
changes to the data channel object.
Change-Id: Ie164cc5823cd5f9782fc5c9a63aa4c76b8229639
Bug: webrtc:11547, webrtc:11687
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177244
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31533}
Add a few DCHECKs and comments about upcoming work.
Bug: webrtc:11547
Change-Id: I2d42f48cb93f31e70cf9fe4b3b62241c38bc9d8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177106
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31530}
This reverts commit 71db9acc4019b8c9c13b14e6a022cbb3b4255b09.
Reason for revert: breaks downstream project.
Reason for force push: win bot broken.
Original change's description:
> RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
>
> This change adds exposure of a new transceiver method for
> modifying the extensions offered in the next SDP negotiation,
> following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.
>
> Features:
> - The interface allows to control the negotiated direction as
> per https://tools.ietf.org/html/rfc5285#page-7.
> - The interface allows to remove an extension from SDP
> negotiation by modifying the direction to
> RtpTransceiverDirection::kStopped.
>
> Note: support for signalling directionality of header extensions
> in the SDP isn't implemented yet.
>
> https://chromestatus.com/feature/5680189201711104.
> Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
>
> Bug: chromium:1051821
> Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31487}
TBR=hta@webrtc.org,handellm@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: true
Bug: chromium:1051821
Change-Id: I70e1a07225d7eeec7480fa5577d8ff647eba6902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31516}
This unblocks injecting platform-specific resources, such as power
usage signals in Chrome.
This CL adds AddAdaptationResource to PeerConnectionInterface and
integration tests verifying that if an injected resource is overusing,
resolution will soon be reduced.
To aid testing, some testing-only classes have been updated.
Bug: webrtc:11525
Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31505}
This allows const getters that query const state to be called without
marshalling calls between threads. This must not be used to
return pointers/references etc.
I'm starting by using this macro with the data channel which has a
few of these getters, as well as changing things a bit to make more
parts of the implementation, const.
Change-Id: I6ec7a3774cd8f7be2ef122fb7c7fc5919afee600
Bug: webrtc:11547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176846
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31489}
It has been reported that sometimes FPS is undefined, causing the test
to be flaky.
Bug: webrtc:11651
Change-Id: Ieea33833724defa46110aad5d103aa16bfbea861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176516
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31481}
The "data codecs" list is not used when SCTP is in use, so it's
unnecessary to remove the RTP codec from it.
Followup from https://webrtc-review.googlesource.com/c/src/+/176327
Bug: none
Change-Id: Ie2787ba7e3030214dd7130faa0bb9cf8b8e66a99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176369
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31404}
These were leftovers from a previous refactoring.
Bug: none
Change-Id: Iee12c2f7f9a7d80ae8e67aa9134ec84894f94960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176327
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31392}
This patch fixes a minor bug in the implementation of
surface_ice_candidates_on_ice_transport_type_changed. The existing
implementation correctly handles the surfacing, but accidentally also
set the SetNeedsIceRestartFlag, which made _next_ offer contain
a ice restart.
Modified existing testcase to verify this.
Bug: webrtc:8939
Change-Id: If566e3249296467668627e5941495f6036cbd903
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176127
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31363}
Logging of content name (mid) is valuable to debug issues
in scenarios with multiple m= line sections in SDP.
For example, video conferencing applications which
uses SFU and Unified Plan SDPs will likely to leverage
from more detailed logs when issues need to be debugged.
Bug: webrtc:10139
Change-Id: Id52ba3ad54af5caa0f8c03daaa51bdb0caf9fe67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175115
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31302}
Also re-enable the TestAnnotationsOnWrongQueueDebug test and rename
the test suite to SequenceCheckerDeathTest so that it gets executed
before other tests.
Bug: webrtc:11577
Change-Id: I3b8037644e4b9139755ccecb17e42b09327e4996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175346
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31290}
This adds priority to the API configuration of datachannels,
and passes the value in the OPEN message.
It does not yet influence SCTP prioritization of messages.
Bug: chromium:1083227
Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31287}
This reverts commit 9a925c9ce33a6ccdd11b545b11ba68e985c2a65d.
Reason for revert: The original CL is updated in PS #2 to
fix the googRtt issue which was that when the legacy sender
stats were put in "aggregated_senders" we forgot to update
rtt_ms the same way that we do it for "senders".
Original change's description:
> Revert "Improve outbound-rtp statistics for simulcast"
>
> This reverts commit da6cda839dac7d9d18eba8d365188fa94831e0b1.
>
> Reason for revert: Breaks googRtt in legacy getStats API
>
> Original change's description:
> > Improve outbound-rtp statistics for simulcast
> >
> > Bug: webrtc:9547
> > Change-Id: Iec4eb976aa11ee743805425bedb77dcea7c2c9be
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168120
> > Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > 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@{#31097}
>
> TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:9547
> Change-Id: I06673328c2a5293a7eef03b3aaf2ded9d13df1b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174443
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31165}
TBR=hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,hta@webrtc.org,elrello@microsoft.com
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9547
Change-Id: I723744c496c3c65f95ab6a8940862c8b9f544338
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31169}
This reverts commit 712ebbb5b73baf30f11711efdceb6f08248fac38.
There is apparently more usage in the wild than anticipated.
Bug: chromium:1068705
Change-Id: If2f3907e509570d305670206d8d3724413964208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31163}
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.
Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
