This reverts commit cae4656d4a.
Reason for revert: Breaks downstream build (missing INSTANTIATE_TEST_SUITE_P in pc/data_channel_integrationtest.cc).
Original change's description:
> Split peer_connection_integrationtest.cc into pieces
>
> This creates two integration tests: One for datachannel, the other
> for every test that is not datachannel.
>
> It separates out the common framework to a new file in pc/test.
> Also applies some fixes to IWYU.
>
> Bug: None
> Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33244}
TBR=hbos@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: None
Change-Id: I7dbedd3256cb7ff47eb5f8cd46c7c044ed0aa1e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207283
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33255}
This creates two integration tests: One for datachannel, the other
for every test that is not datachannel.
It separates out the common framework to a new file in pc/test.
Also applies some fixes to IWYU.
Bug: None
Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33244}
and replace with specific compiler flags around the remaining failing
tests.
Bug: webrtc:3608, webrtc:11305, webrtc:11282
Change-Id: Iac45e52efcdfebc1bb85697a7606873255643e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206980
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33233}
These test that a datachannel will deliver messages that are sent
while the network is down, both with and without ICE restarts.
Bug: webrtc:11891
Change-Id: I6c6633a655b0dd8e2e265aaf98789ca10b36884e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206801
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33223}
This reverts commit 6b143c1c06.
Reason for revert:
Relanding with updated expectations for SctpTransport::Information
based on TransceiverStateSurfacer in Chromium.
Original change's description:
> Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController"
>
> This reverts commit 6cd4058504.
>
> Reason for revert: Breaks WebRTC Chromium FYI Bots
>
> First failure:
> https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
>
> Failed tests:
> WebRtcDataBrowserTest.CallWithSctpDataAndMedia
> WebRtcDataBrowserTest.CallWithSctpDataOnly
>
> Original change's description:
> > Fix unsynchronized access to mid_to_transport_ in JsepTransportController
> >
> > * Added several thread checks to JTC to help with programmer errors.
> > * Avoid a few Invokes() to the network thread here and there such
> > as for fetching sctp transport name for getStats(). The transport
> > name is now cached when it changes on the network thread.
> > * JsepTransportController instances now get deleted on the network
> > thread rather than on the signaling thread + issuing an Invoke()
> > in the dtor.
> > * Moved some thread hops from JTC over to PC which is where the problem
> > exists and also (imho) makes it easier to see where hops happen in
> > the PC code.
> > * The sctp transport is now started asynchronously when we push down the
> > media description.
> > * PeerConnection proxy calls GetSctpTransport directly on the network
> > thread instead of to the signaling thread + blocking on the network
> > thread.
> > * The above changes simplified things for webrtc::SctpTransport which
> > allowed for removing locking from that class and delete some code.
> >
> > Bug: webrtc:9987, webrtc:12445
> > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33191}
>
> TBR=tommi@webrtc.org,hta@webrtc.org
>
> Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9987
> Bug: webrtc:12445
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33204}
TBR=tommi@webrtc.org,hta@webrtc.org,guidou@webrtc.org
# Not skipping CQ checks because this is a reland.
Bug: webrtc:9987
Bug: webrtc:12445
Change-Id: Icb205cbac493ed3b881d71ea3af4fb9018701bf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206560
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33219}
This reverts commit 6cd4058504.
Reason for revert: Breaks WebRTC Chromium FYI Bots
First failure:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925
Failed tests:
WebRtcDataBrowserTest.CallWithSctpDataAndMedia
WebRtcDataBrowserTest.CallWithSctpDataOnly
Original change's description:
> Fix unsynchronized access to mid_to_transport_ in JsepTransportController
>
> * Added several thread checks to JTC to help with programmer errors.
> * Avoid a few Invokes() to the network thread here and there such
> as for fetching sctp transport name for getStats(). The transport
> name is now cached when it changes on the network thread.
> * JsepTransportController instances now get deleted on the network
> thread rather than on the signaling thread + issuing an Invoke()
> in the dtor.
> * Moved some thread hops from JTC over to PC which is where the problem
> exists and also (imho) makes it easier to see where hops happen in
> the PC code.
> * The sctp transport is now started asynchronously when we push down the
> media description.
> * PeerConnection proxy calls GetSctpTransport directly on the network
> thread instead of to the signaling thread + blocking on the network
> thread.
> * The above changes simplified things for webrtc::SctpTransport which
> allowed for removing locking from that class and delete some code.
>
> Bug: webrtc:9987, webrtc:12445
> Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33191}
TBR=tommi@webrtc.org,hta@webrtc.org
Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9987
Bug: webrtc:12445
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33204}
This should account for the linux_x86_dbg bot flaking on the test.
Bug: none
Change-Id: I77f9134941c42eae078b2da57e9b05517bdda923
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206064
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33198}
* Added several thread checks to JTC to help with programmer errors.
* Avoid a few Invokes() to the network thread here and there such
as for fetching sctp transport name for getStats(). The transport
name is now cached when it changes on the network thread.
* JsepTransportController instances now get deleted on the network
thread rather than on the signaling thread + issuing an Invoke()
in the dtor.
* Moved some thread hops from JTC over to PC which is where the problem
exists and also (imho) makes it easier to see where hops happen in
the PC code.
* The sctp transport is now started asynchronously when we push down the
media description.
* PeerConnection proxy calls GetSctpTransport directly on the network
thread instead of to the signaling thread + blocking on the network
thread.
* The above changes simplified things for webrtc::SctpTransport which
allowed for removing locking from that class and delete some code.
Bug: webrtc:9987, webrtc:12445
Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33191}
(and a subclass of QueuedTask in one place, where needed for move
semantics).
Bug: webrtc:11339
Change-Id: I109de41a8753f177db1bbb8d21b6744eb3ad2de0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201734
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33021}
This version uses relative_packet_arrival_delay as the target metric.
Bug: none
Change-Id: Ie6eb575ce4d13fd005f026862892b14bd4fb1135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201620
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32962}
These tests create multiple transceivers, and attempt to renegotiate.
They serve to show where the limit is for adequate performance (arbitrarily
set as one second).
This version should pass on all platforms; it only tests up to 16 tracks.
Bug: webrtc:12176
Change-Id: I1561a56f6a392dbfa954319c538a9959c3a6f590
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193061
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32820}
This reverts commit f08db1be94.
Reason for revert: It looks like this breaks Chromium FYI Windows bots.
See https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/6988.
If this is not the culprit I will reland.
Original change's description:
> Enable FlexFEC as a receiver video codec by default
>
> - Add Flex FEC format as default supported receive codec
> - Disallow advertising FlexFEC as video sender codec by default until implementation is complete
> - Toggle field trial "WebRTC-FlexFEC-03-Advertised"s behavior for receiver to use as kill-switch to prevent codec advertising
>
> Bug: webrtc:8151
> Change-Id: Iff367119263496fb335500e96641669654b45834
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191947
> Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32639}
TBR=brandtr@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,crodbro@webrtc.org,crodbro@google.com,yinwa@webrtc.org,philipp.hancke@googlemail.com,hmaniar@nvidia.com
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8151
Change-Id: Ia1788a1cf34e0fc9500a081552f6ed03d0995d5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194334
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32657}
- Add Flex FEC format as default supported receive codec
- Disallow advertising FlexFEC as video sender codec by default until implementation is complete
- Toggle field trial "WebRTC-FlexFEC-03-Advertised"s behavior for receiver to use as kill-switch to prevent codec advertising
Bug: webrtc:8151
Change-Id: Iff367119263496fb335500e96641669654b45834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191947
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32639}
This fixes regressions caused by:
https://webrtc-review.googlesource.com/c/src/+/183120
... which disabled payload type demuxing when multiple video tracks are
present, to avoid one channel creating a default track intended for
another channel.
However, this isn't an issue when not bundling, as each track will be
delivered on separate transport.
And it's also not an issue when each track uses a distinct set of
payload types (e.g., VP8 is mapped to PT 96 in one m= section, and PT 97
in another).
This CL addresses both of those cases; PT demuxing is only disabled
when two bundled m= sections have overlapping payload types.
Bug: chromium:1139052, webrtc:12029
Change-Id: Ied844bffac2a5fac29147c11b56a5f83a95ecb36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187560
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32419}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
This is a reland of 11dc6571cb
One fix that makes Web Platform Tests pass in debug mode is applied.
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
Bug: chromium:980879
Change-Id: Ide31d929ac5ea118d83fdf6a35a592af23f7dfa7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181263
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31907}
This reverts commit 11dc6571cb.
Reason for revert: Breaks Chromium WPT tests
Original change's description:
> Implement transceiver.stop()
>
> This adds RtpTransceiver.StopStandard(), which behaves according to
> the specification at
> https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>
> It modifies RTCPeerConnection.getTransceivers() to return only
> transceivers that have not been stopped.
>
> Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
>
> Bug: chromium:980879
> Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31893}
TBR=sakal@webrtc.org,kthelgason@webrtc.org,hta@webrtc.org,guidou@webrtc.org,marinaciocea@webrtc.org
Change-Id: Ibdc24f7d41e481293ca74ba6d1572de64f7e4654
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:980879
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181262
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31897}
This adds RtpTransceiver.StopStandard(), which behaves according to
the specification at
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
It modifies RTCPeerConnection.getTransceivers() to return only
transceivers that have not been stopped.
Rebase of armax' https://webrtc-review.googlesource.com/c/src/+/172762
Bug: chromium:980879
Change-Id: I7d383ee874ccc0a006fdcf280496b5d4235425ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180580
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31893}
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}
This reverts commit 9a925c9ce3.
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 da6cda839d.
>
> 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 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}
This reverts commit 8e8b36a94a.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
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}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
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}
This reverts commit 133bf2bd28.
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 e57b266a20.
>
> Reason for revert: Fixed negotiation of send-only clients.
>
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762.
> >
> > 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}
This reverts commit e57b266a20.
Reason for revert: Fixed negotiation of send-only clients.
Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762.
>
> 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}
We make sure the fake clock is constructed first thing,
so that all subsequent calls to GetClockForTesting() are
consistent and non-racy.
This proper scoping also allows to remove some explicit
destructions which are no longer necessary.
Bug: webrtc:11282
Change-Id: Id9263617c2e2b025b17d9bcb9cd415d651405a8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166043
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30309}
The test previously assumed that doing an offer/answer exchange would
leave the ICE state completed which is unlikely in practice but probably
worked most of the time in test since the network components were faked.
Bug: webrtc:11280
Change-Id: I9bc0e1490b0b8401cc832b73da9dc7fe870bc9fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166400
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30289}
This CL propagates a "closed" signal from DTLS up to the
SCTP section of the data channel controller, where it causes
closing of all open datachannels.
Bug: chromium:1030631, webrtc:10360
Change-Id: I88bb9e1aff5c25f330edfd092ef609d4fcc3a9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162206
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30099}
This is in preparation for writing tests that verify that
they close, and that they close at the right times.
Bug: chromium:1030631, webrtc:10360
Change-Id: I8129a9fc9731c1bfe1a660e82e23c1aeff1e5087
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162181
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30091}
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
Move the GCM srtp cipher suites below the default SRTP_AES128_CM_SHA1_80 one.
This will not negotiate them by default since they have an impact on packet overhead for audio-only calls.
GCM can still be negotiated if the peer offers it as preferred cipher suite or answers with just that cipher suite.
BUG=chromium:713701
Change-Id: I79bd4ab827e5c7f55f5550d14db3f4217a7eff86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158404
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Justin Uberti <juberti@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29672}
This is a reland of 16d4c4d4fb after
downstream project was updated to be prepared for the new SdpType.
Original change's description:
> Implement rollback for setRemoteDescription
>
> Bug: chromium:980875
> Change-Id: I4575e9ad1902a20937f9812f49edee2a2441f76d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153525
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29422}
TBR=steveanton@webrtc.org
Bug: chromium:980875
Change-Id: Iba8d25bf2dc481b25a03eeae9818bd5f4c3eaa2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156569
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29460}
Bug: chromium:980875
Change-Id: I4575e9ad1902a20937f9812f49edee2a2441f76d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153525
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29422}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
Adds a field trial and configuration parameter to control whether
datagram transport may be used for data channels in a receive-only
manner. By default, if use_datagram_transport_for_data_channels is
enabled, PeerConnection will create a datagram transport and offer its
use for outgoing calls as well as accept incoming offers with compatible
datagram transport parameters.
With this change, a receive_only mode is added for datagram transport
data channels. When receive_only is set, the PeerConnection will not
create or offer datagram transports for outgoing calls, but will accept
incoming calls that offer compatible datagram transport parameters.
Bug: webrtc:9719
Change-Id: I35667bcc408ea4bbc61155898e6d2472dd262711
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154463
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29327}
This fixes a DCHECK during teardown in the case when the primary
DataChannelTranspot (eg. DatagramTransport) is successfully negotiated.
DatagramTransport expects the DataSink to be unset before it's deleted.
This was not caught by existing tests because the fallback transport
(SctpDataChannelTransport) does not have the same DCHECK.
Also adds a regression test for the issue, in which SCTP is available
as a fallback but DataChannelTransport is negotiated successfully.
Bug: webrtc:9719
Change-Id: I414d964d3c85d3d01cdb5e34d6b248659a613c39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154365
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29292}
This is a reland of 487f9a17e4
Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
>
> Also clears SctpTransport before deleting JsepTransport.
>
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport. This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
>
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
>
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
>
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP. Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left. For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports. Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
>
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}
Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels. There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.
PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.
Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks. This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state. This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.
For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.
Datagram transport use is negotiated. As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer. If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport. If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.
If DTLS is not enabled, there is no viable fallback. In this case, the data
channels are negotiated as media transport data channels. If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.
Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
Bug: webrtc:10419
Change-Id: I18528bf2526e933568bf052de76a434f012161da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148320
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28838}
In order to be able to detect and measure context around candidate pair changes.
Bug: webrtc:10419
Change-Id: Iab0d7e7c80d925d1aa44617fc35975fdc6bbc6b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147340
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28779}
We rather have an unmet expectation than let the test crash.
Bug: webrtc:10827
Change-Id: I9e3d2dfb7cb856976305cd50377a71a2ed2ab4b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146700
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28651}
instead of using factories for MediaEngine and RtcEventLog that rely on GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ie1135f70f4ae4d047c4d6bf2db61489a663385aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141875
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28328}
This is a reland of 9469c784db
Original change's description:
> Added OnIceCandidateError to API and implementation
>
> Bug: webrtc:3098
> Change-Id: I27ffd015ebf9e8130c1288f7331b0e2fdafb01ef
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135953
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28173}
TBR=steveanton@webrtc.org
Bug: webrtc:3098
Change-Id: I77af2065fc1479273f399e2b3d919f98fe8ac23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140641
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28179}
Bug: webrtc:3098
Change-Id: I27ffd015ebf9e8130c1288f7331b0e2fdafb01ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135953
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28173}
from a field trial to RTCConfiguration.
The test coverage is also expanded for the underlying feature.
Bug: None
Change-Id: Ic9c1362867e4a956c5453be7a9355083b6a442f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28143}
Since ScopedFieldTrials is not thread safe, if it goes out of
scope while other threads are still running and possibly querying
the field trials then it's possible to hit an MSAN failure.
Bug: webrtc:10694
Change-Id: I93c94f1008e4478d98ec1545bbc0a7536739e479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139460
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28116}
The gathering of host candidates with mDNS names is asynchronous and its
completion can happen after a srflx candidate is gathered by the same
underlying socket. We have a broken check in UDPPort::CreateConnection()
that assumes the gathering of host and srflx candidates is sequential.
This CL also does minor refactoring and clean-up.
Bug: chromium:944577
Change-Id: Ic28136a9515081f40b232a22fcbf4209814ed33a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138043
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28030}
This reverts commit 46afbf9481.
Reason for revert: Tightened protocol name handling.
Original change's description:
> Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
>
> This reverts commit 37f2b43274.
>
> Reason for revert: fuzzer failures
>
> Original change's description:
> > Reland "Version 2 "Refactoring DataContentDescription class""
> >
> > This is a reland of 14b2758726
> >
> > Original change's description:
> > > Version 2 "Refactoring DataContentDescription class"
> > >
> > > (substantial changes since version 1)
> > >
> > > This CL splits the cricket::DataContentDescription class into
> > > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > > and cricket::SctpDataContentDescription (used for SCTP only).
> > >
> > > SctpDataContentDescription no longer inherits from
> > > MediaContentDescriptionImpl, and no longer contains "codecs".
> > >
> > > Due to usage of internal interfaces by consumers, shimming the old
> > > DataContentDescription API is needed.
> > >
> > > A new cricket::DataContentDescription class is defined, which is
> > > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > > It exposes as little functionality as possible, but supports the
> > > concerned consumer's usage
> > >
> > > Design document:
> > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> > >
> > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > >
Bug: webrtc:10358
Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27933}
This reverts commit 37f2b43274.
Reason for revert: fuzzer failures
Original change's description:
> Reland "Version 2 "Refactoring DataContentDescription class""
>
> This is a reland of 14b2758726
>
> Original change's description:
> > Version 2 "Refactoring DataContentDescription class"
> >
> > (substantial changes since version 1)
> >
> > This CL splits the cricket::DataContentDescription class into
> > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > and cricket::SctpDataContentDescription (used for SCTP only).
> >
> > SctpDataContentDescription no longer inherits from
> > MediaContentDescriptionImpl, and no longer contains "codecs".
> >
> > Due to usage of internal interfaces by consumers, shimming the old
> > DataContentDescription API is needed.
> >
> > A new cricket::DataContentDescription class is defined, which is
> > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > It exposes as little functionality as possible, but supports the
> > concerned consumer's usage
> >
> > Design document:
> > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> >
> > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> >
> > Bug: webrtc:10358
> > Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27853}
>
> Bug: webrtc:10358
> Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27896}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ied6d9fb96aafe9c957f2658b34b5331b1f359b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135986
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27917}
This is a reland of 14b2758726
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
Bug: webrtc:10358
Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27896}
Tested with a Web Platform Test; the test added here is useful, but
does not exercise the bug.
Bug: chromium:959128
Change-Id: Ia2e7f9e015b2345dd02d341b0fe27f58b64aa81e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135575
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27889}
This reverts commit 14b2758726.
Reason for revert: Internal import failed.
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
TBR=danilchap@webrtc.org,steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ibc16ba14c1cbf50345a9b79151b79df140482539
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27855}
(substantial changes since version 1)
This CL splits the cricket::DataContentDescription class into
two classes: cricket::RtpDataContentDescription (used for RTP data)
and cricket::SctpDataContentDescription (used for SCTP only).
SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".
Due to usage of internal interfaces by consumers, shimming the old
DataContentDescription API is needed.
A new cricket::DataContentDescription class is defined, which is
a shim over RtpDataContentDescription and SctpDataContentDescription.
It exposes as little functionality as possible, but supports the
concerned consumer's usage
Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Bug: webrtc:10358
Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27853}
This is a reland of cd8d1cf68e
Original change's description:
> Surface ICE candidates that match an updated candidate filter.
>
> After this change an ICE agent can surface candidates that do not match
> the previous filter but are allowed by the updated one. The candidate
> filter, as part of the internal implementation in the ICE transport,
> manifests the RTCIceTransportPolicy field in RTCConfiguration.
>
> This new feature would allow an ICE agent to gather new candidates when
> the transport policy changes from e.g. 'relay' to 'all' without an ICE
> restart.
>
> A caveat in the current implementation remains, and a candidate can
> surface multiple times if the transport policy, or the candidate filter
> directly, performs multiple transitions from a value that disallows to
> one that allows the underlying candidate type. For example, if the
> transport policy is updated by 'all' -> 'relay' -> 'all', the same host
> candidate can surface after the second update.
>
>
> Bug: webrtc:8939
> Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27674}
Bug: webrtc:8939
Change-Id: I9c32b1ea05028ecd937ab4912779dd958faf734f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133582
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27694}
This reverts commit cd8d1cf68e.
Reason for revert: breaks an internal project
Original change's description:
> Surface ICE candidates that match an updated candidate filter.
>
> After this change an ICE agent can surface candidates that do not match
> the previous filter but are allowed by the updated one. The candidate
> filter, as part of the internal implementation in the ICE transport,
> manifests the RTCIceTransportPolicy field in RTCConfiguration.
>
> This new feature would allow an ICE agent to gather new candidates when
> the transport policy changes from e.g. 'relay' to 'all' without an ICE
> restart.
>
> A caveat in the current implementation remains, and a candidate can
> surface multiple times if the transport policy, or the candidate filter
> directly, performs multiple transitions from a value that disallows to
> one that allows the underlying candidate type. For example, if the
> transport policy is updated by 'all' -> 'relay' -> 'all', the same host
> candidate can surface after the second update.
>
>
> Bug: webrtc:8939
> Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27674}
TBR=shampson@webrtc.org,qingsi@webrtc.org,jeroendb@webrtc.org,sukhanov@webrtc.org
Change-Id: Idd51a640e55a612b42fe8b69e05dff57a22d021a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133581
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27677}
After this change an ICE agent can surface candidates that do not match
the previous filter but are allowed by the updated one. The candidate
filter, as part of the internal implementation in the ICE transport,
manifests the RTCIceTransportPolicy field in RTCConfiguration.
This new feature would allow an ICE agent to gather new candidates when
the transport policy changes from e.g. 'relay' to 'all' without an ICE
restart.
A caveat in the current implementation remains, and a candidate can
surface multiple times if the transport policy, or the candidate filter
directly, performs multiple transitions from a value that disallows to
one that allows the underlying candidate type. For example, if the
transport policy is updated by 'all' -> 'relay' -> 'all', the same host
candidate can surface after the second update.
Bug: webrtc:8939
Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27674}
This CL splits the cricket::DataContentDescription class into
two classes: cricket::DataContentDescription (used for RTP data) and
cricket::SctpDataContentDescription (used for SCTP only).
SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".
Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
Bug: webrtc:10358
Change-Id: Ie7160610506aeef56d1f821b5fdb5d9492201f43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27651}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.
Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.
Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
This updates behavior to be aligned with the WebRTC spec.
Bug: chromium:948055
Change-Id: Id3bbf05b3df084c9b7f7d12598c09187679d60fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130493
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27394}
The old iceConnectionState only becomes completed on the controlling side, and only if we're not configured to continually gather ICE candidates. This change makes the new ICE connection state do the same thing.
Bug: webrtc:10356
Change-Id: I82ca854a638a52674e4ca43364cf454dacb0cf1e
Reviewed-on: https://webrtc-review.googlesource.com/c/124360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26950}
