Commit Graph

1278 Commits

Author SHA1 Message Date
2e94de596e Add GetSctpStats to PeerConnectionInternal, remove sctp_data_channels()
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}
2020-06-16 16:36:42 +00:00
7d3cfbf90d Inject signaling and network threads to DataChannel.
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}
2020-06-16 10:22:19 +00:00
67f80cea66 test: remove obsolete comment
BUG=None

Change-Id: Ib343c2aa7b16ce8a75959a3412f5cf2b305a207c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177249
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31528}
2020-06-15 21:00:56 +00:00
6f727da62b Revert "RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions."
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}
2020-06-12 16:26:49 +00:00
4c1e7cc19b [Adaptation] Add ability to inject resources on the PeerConnection.
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}
2020-06-11 14:17:01 +00:00
9276e2c39b Remove enable_simulcast_stats config flag as not needed anymore
Bug: webrtc:9547
Change-Id: Ie50453aa3496d16bfadfc9fdd3e7e6982278cfba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176841
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31492}
2020-06-10 15:59:32 +00:00
0ca13d97d2 Introduce BYPASS_PROXY_CONSTMETHOD0.
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}
2020-06-10 13:52:36 +00:00
71db9acc40 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}
2020-06-10 13:02:44 +00:00
8498c259b9 test: use kCnCodecName instead of hardcoding "CN"
BUG=None

Change-Id: I8d10cf18418c74fea58f4e31f81e5fe4f00d65bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176519
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31482}
2020-06-10 10:12:32 +00:00
7804c54b97 [Stats flake] Mark outbound-rtp.framesPerSecond as optional.
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}
2020-06-10 09:29:00 +00:00
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release.
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
2020-06-08 12:59:40 +00:00
10ef847289 Correct name of DC.dataChannelIdentifier stats member
Bug: webrtc:8787
Change-Id: Ie32b38f0671e89e94017f439de7614142328642f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176509
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31457}
2020-06-07 21:57:50 +00:00
bc1f910456 Reland "Added mid to error messages reported during SDP apply."
This reverts commit 341434e4da2c193b8842917d73afed6eea3a4332.

Reason for revert: another attempt to land with Chromium
test updated to accept both error messages by CL:
https://chromium-review.googlesource.com/c/chromium/src/+/2228545

Original change's description:
> Revert "Added mid to error messages reported during SDP apply."
>
> This reverts commit d2890e8833796f13c4a1243769be966bebdfcaa7.
>
> Reason for revert: speculative: WebRtcBrowserTest.NegotiateUnsupportedVideoCodec broken on all FYI bots, example: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/6659
>
> Original change's description:
> > Added mid to error messages reported during SDP apply.
> >
> > Bug: webrtc:10139
> > Change-Id: I7462b632e00a2da7b189b63022d30f594700b68a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176400
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> > Cr-Commit-Position: refs/heads/master@{#31421}
>
> TBR=tommi@webrtc.org,yura.yaroshevich@gmail.com
>
> Change-Id: I18972815df10e2bd7b914ad82df9596009c2fecc
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10139
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176418
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Olga Sharonova <olka@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31425}

TBR=tommi@webrtc.org,olka@webrtc.org,yura.yaroshevich@gmail.com

# Not skipping CQ checks because this is a reland.

Bug: webrtc:10139
Change-Id: I603d3891c43ac396bf0ba98c6de189663235c8af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176448
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/master@{#31445}
2020-06-04 11:16:46 +00:00
341434e4da Revert "Added mid to error messages reported during SDP apply."
This reverts commit d2890e8833796f13c4a1243769be966bebdfcaa7.

Reason for revert: speculative: WebRtcBrowserTest.NegotiateUnsupportedVideoCodec broken on all FYI bots, example: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Linux%20Tester/6659

Original change's description:
> Added mid to error messages reported during SDP apply.
> 
> Bug: webrtc:10139
> Change-Id: I7462b632e00a2da7b189b63022d30f594700b68a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176400
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Cr-Commit-Position: refs/heads/master@{#31421}

TBR=tommi@webrtc.org,yura.yaroshevich@gmail.com

Change-Id: I18972815df10e2bd7b914ad82df9596009c2fecc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10139
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176418
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31425}
2020-06-03 13:17:07 +00:00
d2890e8833 Added mid to error messages reported during SDP apply.
Bug: webrtc:10139
Change-Id: I7462b632e00a2da7b189b63022d30f594700b68a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176400
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/master@{#31421}
2020-06-03 11:54:52 +00:00
3bd1c1858f Cleanup: Remove non-required helper function to remove codecs
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}
2020-06-02 10:35:35 +00:00
b59f337fbd Remove leftover SCTP "codec name" constants
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}
2020-05-30 15:09:48 +00:00
1a4975642b fix typos in comments
BUG=none

Change-Id: I3e500213a7a272b6422db35575389b368c0e3ef2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176131
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#31380}
2020-05-28 17:56:45 +00:00
e309651f33 Don't SetNeedsIceRestartFlag if widening candidate filter when surface_ice_candidates_on_ice_transport_type_changed
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}
2020-05-27 08:42:10 +00:00
b41316cd4c test: fix typo
BUG=none

Change-Id: Ie50a5666c2db08579580c3b75475171ab884ede5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176124
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31357}
2020-05-26 16:27:57 +00:00
2af35ab984 FakeAudioCaptureModule: remove lock recursions.
This change removes lock recursions and adds thread annotations.

The module had incorrect locking WRT the callback critical section:

ProcessFrameP: locks crit_
ReceiveFrameP: locks crit_callback_
-------------
SendFrameP: locks crit_callback_
MicrophoneVolume: locks crit_

Lock crit_callback_ was rolled in under crit_ instead.

Bug: webrtc:11567
Change-Id: I974fe91d44de0ddf1a1287fe91db9dfe63a61af9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175662
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31313}
2020-05-18 18:01:58 +00:00
3a35312b64 In pc/ replace mock macros with unified MOCK_METHOD macro
Bug: webrtc:11564
Change-Id: I09b28654b7b71a77224e7cf72fdf6a1e4823e67a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175137
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31310}
2020-05-18 17:06:25 +00:00
c325246753 Log content name (aka mid) in messages from channel.cc
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}
2020-05-18 13:09:19 +00:00
c18b7bfeb6 JsepTransport: remove lock recursions.
This change removes lock recursions and adds thread annotations.

Bug: webrtc:11567
Change-Id: Iefe1875182b7f8f8df3e9bd02e964530389b0b3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175123
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31296}
2020-05-18 09:58:09 +00:00
37e42bed01 Give correct error code when SCTP is abruptly terminated.
Bug: chromium:1030631
Change-Id: I1890d6c7b30c06de1f4fdc6fe0cf1ff62ea4a63d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174830
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31293}
2020-05-18 05:23:27 +00:00
ec3ba734e9 Don't wrap the main thread when running death tests.
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}
2020-05-17 17:15:10 +00:00
fd5ae7f959 Pass datachannel priority in DC open messages
This adds priority to the API configuration of datachannels,
and passes the value in the OPEN message.

It does not yet influence SCTP prioritization of messages.

Bug: chromium:1083227
Change-Id: I46ddd1eefa0e3d07c959383788b9e80fcbfa38d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175107
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31287}
2020-05-17 10:57:27 +00:00
fa8019c3c3 Clear address:port in icecandidateerror for tcp servers with private IP
Bug: chromium:1072401
Change-Id: I6af81a2b2b22b5f8d7edb8fb7f66f69b866db1c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174753
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31275}
2020-05-15 11:30:20 +00:00
1aec2bf115 reorder sdes suites to not prefer gcm
BUG=chromium:713701

Change-Id: I1ef00df7a7b86a83ae97d4c7c5f41d85eb60b391
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174803
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31225}
2020-05-12 14:43:43 +00:00
6efc14b33d VideoTrackSourceInterface: make some newly introduced methods pure virtual.
Bug: webrtc:11114
Change-Id: Ic4d3835ae84b6a652c49f30a9c275870bbf3dacf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174440
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31211}
2020-05-11 12:28:32 +00:00
a0ff50c031 Reland "Improve outbound-rtp statistics for simulcast"
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}
2020-05-05 20:22:19 +00:00
9a925c9ce3 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}
2020-05-05 13:38:51 +00:00
c1aaf4cb38 Revert "disallow pairing ICE-TCP with a local ip address"
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}
2020-05-05 06:59:45 +00:00
cc73ed3e70 APM: Add build flag to allow building WebRTC without APM
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}
2020-04-26 23:06:44 +00:00
1f0677d01e Remove some TODOs from pc/.
Bug: webrtc:10198
Change-Id: I1782a8ef1248578fcc3ffc8c03b5419225a51350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31113}
2020-04-20 23:51:16 +00:00
ee0864364d Remove DetermineIceRole workaround.
Bug: webrtc:10198, chromium:628676
Change-Id: I65a57a2d23b714f9cdddc9122f4b50d523d04dfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173337
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31101}
2020-04-17 13:48:35 +00:00
da6cda839d 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}
2020-04-17 11:28:00 +00:00
b33a0ca1ee Remove deprecated ssl_identity methods
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/170637

Bug: webrtc:11450
Change-Id: I69928ed7236c6a8a569c7dc0383f7debb4408179
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171224
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31086}
2020-04-16 14:21:41 +00:00
3ebb6e93f4 Remove WebRTC-ExcludeTransportSequenceNumberFromFec.
Bug: webrtc:11503
Change-Id: I5e0b7038286d9501a617e002b70638f34ac556ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173580
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31072}
2020-04-15 08:11:30 +00:00
adc4da30f4 [InsertableStreams] Fix video receiver simulcast.
Save the frame transformer set on unsignaled receivers, and set the
transformer when the ssrc becomes known.

Pass the receiver's ssrc on registering the transformed frame callback,
to associate separate frame transformer sinks for each receiver.

Bug: chromium:1065838

Bug: chromium:1065838
Change-Id: I2a214bdb6cb9a8012928a03f046f311c344370f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173201
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31051}
2020-04-11 12:04:24 +00:00
8206bc0f00 Handle missing tcptype on TCP ICE candidates.
Our implementation accepts TCP candidates with a missing tcptype
field, treating this as a passive candidate.

However, if you try to convert such a candidate to SDP and back,
which chromium started to do at some point, this was resulting in an
error. This CL fixes that.

Bug: webrtc:11423
Change-Id: Iec48d340f421f63f2b7a16c9496ea92ccd165981
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172020
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31026}
2020-04-07 21:17:29 +00:00
16d0d371d5 Apply performance-for-range-copy fixes.
This CL has been generated running https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html.

Bug: None
Change-Id: Ia9f6c91776fc8b3ab28fba87ba8ce112f87d5cf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172805
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30996}
2020-04-03 11:36:52 +00:00
55c991cc81 [InsertableStreams] Save the transformer to be set on Reconfigure.
Bug: chromium:1052765
Change-Id: Ie1e91d4e9033b8c542cd576f9f04bacb1904c027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172781
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30983}
2020-04-02 14:04:50 +00:00
08d1806e54 Extend rtc::AdapterType with 2g, 3G, 4G & 5G enum values.
This patch adds new enum values for different types of cellular
connections.

The new costs are currently blocked when sending to remote,
(so that arbitrary network switches does not starts occurring).

The end-game for this series to be able to distinguish between
different type of cellular connections in the ice-layer (e.g when
selecting/switching connections).

BUG: webrtc:11473
Change-Id: I587ac8fdff4f6cdd0f8905f327232f58818db4f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172582
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30970}
2020-04-02 07:48:36 +00:00
6f402f991e Remove unnecessary breaks after return.
Patch author: thakis@chromium.org.

TBR=kwiberg@webrtc.org

No-Try: True
Bug: chromium:1066980
Change-Id: Ifcc7e831337bb2a9bf06b0af0bbd9d1c586db78a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172627
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30968}
2020-04-01 22:20:37 +00:00
57cabed0b0 Replace std::string::find() == 0 with absl::StartsWith.
Bug: None
Change-Id: I070c4a5d19455f3a5c5d3ccc05f418545c351987
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30960}
2020-04-01 11:15:00 +00:00
3e9af7fe05 Insert audio frame transformer between depacketizer and decoder.
The frame transformer is passed from RTPReceiverInterface through the
library to be eventually set in ChannelReceive, where the frame
transformation will occur in the follow-up CL.

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: I5af06d1431047ef50d00e304cf95e92a832b4220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171872
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30956}
2020-04-01 08:15:53 +00:00
00b46f7f2a PeerConnection owns the PacketSocketFactory dependency.
The PacketSocketFactory dependency (if present on the object passed to
CreatePeerConnection(...)) is given as a raw pointer to the
PortAllocator, but the unique_ptr remains in the dependencies object
which is destroyed at the end of the Initialize call.

Bug: webrtc:11467
Change-Id: I2ccb22b6313fc6b2887bb581704f73a703092af3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172043
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Jorge Moreira Broche <jemoreira@google.com>
Cr-Commit-Position: refs/heads/master@{#30953}
2020-03-31 22:11:37 +00:00
3e98368ec5 Reland "Distinguish between send and receive codecs"
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.

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}
2020-03-29 21:03:27 +00:00
8515d5a4ab Refactor ssl_stream_adapter API to show object ownership
Backwards compatible overloads are provided.

Bug: none
Change-Id: I065ad6b269fe074745f9debf68862ff70fd09628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170637
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30851}
2020-03-21 18:53:46 +00:00