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101 Commits

Author SHA1 Message Date
Danil Chapovalov
4f281f142a Cleanup FakeRtcEventLog from thread awareness 
To avoid it relying on AsyncInvoker.

Bug: webrtc:12339
Change-Id: I086305a74cc05fc8ed88a651e71a8f707c2c1d5c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202252
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33044}
 2021-01-20 14:06:47 +00:00
Mirko Bonadei
5eb43b4777 Prefix HAVE_SCTP macro with WEBRTC_. 
Generated automatically with:

  git grep -l "\bHAVE_SCTP\b" | xargs \
    sed -i '' 's/HAVE_SCTP/WEBRTC_HAVE_SCTP/g'

Bug: webrtc:11142
Change-Id: I30e16a40ca7a7e388940191df22b705265b42cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202251
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33042}
 2021-01-20 10:51:07 +00:00
Niels Möller
4bab23f550 Update pc/ to use C++ lambdas instead of rtc::Bind 
(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}
 2021-01-18 09:55:33 +00:00
Harald Alvestrand
cc6ae44ae6 Reland "Improve structuring of test for audio glitches." 
This reverts commit 3ae09f541900f18a8b680e70372f7f1d5e06bd0a.

Reason for revert: Revert didn't actually stabilize test. Decreasing # of rounds instead.

Original change's description:
> Revert "Improve structuring of test for audio glitches."
>
> This reverts commit fdbaeda00362a385de85b4c08aa0b536062a8415.
>
> Reason for revert: Breaks downstream project, see https://bugs.chromium.org/p/webrtc/issues/detail?id=12371
>
> Original change's description:
> > Improve structuring of test for audio glitches.
> >
> > Bug: webrtc:12361
> > Change-Id: Ieddc3dafbb638b3bd73dd79bcafa499290fa4340
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201723
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32973}
>
> TBR=hbos@webrtc.org,hta@webrtc.org
>
> Change-Id: Ie337de79a80113958607a7508d136c05fe6d9167
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12361
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202024
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Alex Loiko <aleloi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32993}

TBR=aleloi@webrtc.org,hbos@webrtc.org,hta@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:12361
Change-Id: Ice79f2144d76bd7576cb415538afdd210625cc4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202247
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33019}
 2021-01-18 09:28:30 +00:00
Alex Loiko
3ae09f5419 Revert "Improve structuring of test for audio glitches." 
This reverts commit fdbaeda00362a385de85b4c08aa0b536062a8415.

Reason for revert: Breaks downstream project, see https://bugs.chromium.org/p/webrtc/issues/detail?id=12371

Original change's description:
> Improve structuring of test for audio glitches.
>
> Bug: webrtc:12361
> Change-Id: Ieddc3dafbb638b3bd73dd79bcafa499290fa4340
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201723
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32973}

TBR=hbos@webrtc.org,hta@webrtc.org

Change-Id: Ie337de79a80113958607a7508d136c05fe6d9167
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202024
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32993}
 2021-01-15 10:00:43 +00:00
Harald Alvestrand
fdbaeda003 Improve structuring of test for audio glitches. 
Bug: webrtc:12361
Change-Id: Ieddc3dafbb638b3bd73dd79bcafa499290fa4340
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201723
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32973}
 2021-01-14 10:29:58 +00:00
Harald Alvestrand
94324f2774 Add a test to detect excessive audio delay during renegotiation. 
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}
 2021-01-13 13:48:02 +00:00
Harald Alvestrand
1a9be30702 Add tests for adding many transceivers and renegotiating. 
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}
 2020-12-11 16:35:55 +00:00
Niels Möller
091617dda8 Change TestStunServer::Create to take a SocketServer rather than a thread as argument. 
Bug: None
Change-Id: I8b140c8cb40787473411ae55da3738166340127f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39512
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32753}
 2020-12-03 08:31:41 +00:00
Mirko Bonadei
ce4be1e640 Revert "Enable FlexFEC as a receiver video codec by default" 
This reverts commit f08db1be94e760c201acdc3a121e67453960c970.

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}
 2020-11-20 20:31:39 +00:00
Harsh Maniar
f08db1be94 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}
 2020-11-19 13:47:28 +00:00
Harald Alvestrand
4efa9d0a5f Remove obsolete GetRemoteAudioSSL* functions. 
Bug: webrtc:12054
Change-Id: I56d198cfa2c336155c5173ccd524107d12e6a382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190921
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32528}
 2020-10-30 08:27:31 +00:00
Taylor Brandstetter
d3ef499418 Enable payload type based demuxing with multiple tracks when applicable. 
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}
 2020-10-16 03:09:22 +00:00
Eldar Rello
bd9c33ad59 Add a test to show that H264FmtpSpsPpsIdrInKeyframe parameter is not present in generated SDP 
Bug: webrtc:11769
Change-Id: I0e69d18101678a0de12af060396735d3dc0a08e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185964
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32300}
 2020-10-02 20:49:47 +00:00
Harald Alvestrand
45be0a9810 Add a test for transceivers being removed when stopped. 
Bug: chromium:980879
Change-Id: Icd6b83b4c0ddf5bd3a6121238ec3b34864b64b6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185961
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32260}
 2020-09-30 18:48:23 +00:00
Harald Alvestrand
1ee3325051 When stopping a transceiver, end the receiver's track. 
Bug: webrtc:11840
Change-Id: Ib8171c58fcb13c33ab03398eb3021c07e55ff008
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185181
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32188}
 2020-09-24 21:37:32 +00:00
Erik Språng
ceb44959ca Reland: Wires up WebrtcKeyValueBasedConfig in media engines. 
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}
 2020-09-22 16:08:22 +00:00
Artem Titov
5956a17ed6 Revert "Wires up WebrtcKeyValueBasedConfig in media engines." 
This reverts commit 591b2ab82ead157b5f5a85d5082bd15fe8c51809.

Reason for revert: Breaks downstream project

Original change's description:
> Wires up WebrtcKeyValueBasedConfig in media engines.
> 
> 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}

TBR=mbonadei@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org,perkj@webrtc.org

Change-Id: I3e169149a8b787aa6366bb357abb71794534c63a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11926
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184507
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32132}
 2020-09-17 20:17:38 +00:00
Erik Språng
591b2ab82e Wires up WebrtcKeyValueBasedConfig in media engines. 
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}
 2020-09-17 16:24:10 +00:00
Harald Alvestrand
bedb605c82 Transition ICE gathering state to "new" once all transports go away 
Bug: chromium:1115080
Change-Id: I524ed48ffc2520ce21ad4bdc25fa3b86d9e41af5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182081
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31976}
 2020-08-20 18:55:52 +00:00
Harald Alvestrand
6060df5948 Reland "Implement transceiver.stop()" 
This is a reland of 11dc6571cb4ff3e71dee1557dfff8d9076e108d3

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}
 2020-08-11 10:46:23 +00:00
Harald Alvestrand
a88c9776de Revert "Implement transceiver.stop()" 
This reverts commit 11dc6571cb4ff3e71dee1557dfff8d9076e108d3.

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}
 2020-08-10 18:06:30 +00:00
Harald Alvestrand
11dc6571cb 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}
 2020-08-10 13:29:15 +00:00
Niels Möller
2a70703eb8 Delete MediaTransportInterface and DatagramTransportInterface 
Bug: webrtc:9719
Change-Id: Ic9936a78ab42f4a9bb4cc3265f0a2cf36946558f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31536}
 2020-06-17 08:41:14 +00:00
Jonas Oreland
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
Danil Chapovalov
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
Eldar Rello
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
Henrik Boström
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
Henrik Boström
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
Per Åhgren
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
Eldar Rello
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
Johannes Kron
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
Eldar Rello
d9ebe01540 Improve rollback for rtp data channel 
Bug: chromium:1057333
Change-Id: I4df21bc183a8df398033ebf29a8407bacf873fac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170621
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#30824}
 2020-03-18 21:03:20 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc 
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
 2020-02-10 12:21:17 +00:00
Johannes Kron
8e8b36a94a Revert "Reland "Reland "Reland "Distinguish between send and receive codecs"""" 
This reverts commit 184ea66aed43161f05d80fbb74183a2efccca352.

Reason for revert: Breaks downstream projects.

TBR=steveanton@webrtc.org

Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive codecs"""
>
> This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5.
>
> Reason for revert: Keep logic as is.
>
> Original change's description:
> > Revert "Reland "Reland "Distinguish between send and receive codecs"""
> >
> > This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
> >
> > Reason for revert: Breaks perf test on iOS.
> >
> > Original change's description:
> > > Reland "Reland "Distinguish between send and receive codecs""
> > >
> > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> > >
> > > Reason for revert: Flaky test in Chromium fixed.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive codecs""
> > > >
> > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > > >
> > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive codecs"
> > > > >
> > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > > >
> > > > > Reason for revert: Fixed negotiation of send-only clients.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive codecs"
> > > > > >
> > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > > > >
> > > > > > Reason for revert: breaks negotiation with send-only clients
> > > > > >
> > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive codecs
> > > > > > >
> > > > > > > Even though send and receive codecs may be the same, they might have
> > > > > > > different support in HW. Distinguish between send and receive codecs
> > > > > > > to be able to keep track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > > >
> > > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > > >
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30348}
> > > >
> > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30360}
> > >
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30367}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30373}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30415}

TBR=steveanton@webrtc.org,kron@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: chromium:1029737
Change-Id: Ice25339e7dfb9fc75049bd207d097b0910bd4446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168341
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30484}
 2020-02-07 15:11:08 +00:00
Johannes Kron
184ea66aed Reland "Reland "Reland "Distinguish between send and receive codecs""" 
This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5.

Reason for revert: Keep logic as is.

Original change's description:
> Revert "Reland "Reland "Distinguish between send and receive codecs"""
>
> This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
>
> Reason for revert: Breaks perf test on iOS.
>
> Original change's description:
> > Reland "Reland "Distinguish between send and receive codecs""
> >
> > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> >
> > Reason for revert: Flaky test in Chromium fixed.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive codecs""
> > >
> > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > >
> > > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive codecs"
> > > >
> > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > >
> > > > Reason for revert: Fixed negotiation of send-only clients.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive codecs"
> > > > >
> > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > > >
> > > > > Reason for revert: breaks negotiation with send-only clients
> > > > >
> > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive codecs
> > > > > >
> > > > > > Even though send and receive codecs may be the same, they might have
> > > > > > different support in HW. Distinguish between send and receive codecs
> > > > > > to be able to keep track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > > >
> > > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > > >
> > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > >
> > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > >
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30348}
> > >
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30360}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30367}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30373}

TBR=steveanton@webrtc.org,kron@webrtc.org


Bug: chromium:1029737
Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30415}
 2020-01-29 18:53:54 +00:00
Bjorn A Mellem
0cda7b832a Allow non-identical datagram transport parameters. 
Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport.  This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).

This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.

Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
 2020-01-29 18:14:24 +00:00
Johannes Kron
a104ceb0ce Revert "Reland "Reland "Distinguish between send and receive codecs""" 
This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.

Reason for revert: Breaks perf test on iOS.

Original change's description:
> Reland "Reland "Distinguish between send and receive codecs""
> 
> This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> 
> Reason for revert: Flaky test in Chromium fixed.
> 
> Original change's description:
> > Revert "Reland "Distinguish between send and receive codecs""
> > 
> > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > 
> > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > 
> > Original change's description:
> > > Reland "Distinguish between send and receive codecs"
> > > 
> > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > 
> > > Reason for revert: Fixed negotiation of send-only clients.
> > > 
> > > Original change's description:
> > > > Revert "Distinguish between send and receive codecs"
> > > >
> > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > >
> > > > Reason for revert: breaks negotiation with send-only clients
> > > >
> > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive codecs
> > > > >
> > > > > Even though send and receive codecs may be the same, they might have
> > > > > different support in HW. Distinguish between send and receive codecs
> > > > > to be able to keep track of which codecs have HW support.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > >
> > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > 
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > 
> > > 
> > > Bug: chromium:1029737
> > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30348}
> > 
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> > 
> > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30360}
> 
> TBR=steveanton@webrtc.org,kron@webrtc.org
> 
> Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30367}

TBR=steveanton@webrtc.org,kron@webrtc.org

Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30373}
 2020-01-24 16:44:17 +00:00
Johannes Kron
9bac68c0cc Reland "Reland "Distinguish between send and receive codecs"" 
This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.

Reason for revert: Flaky test in Chromium fixed.

Original change's description:
> Revert "Reland "Distinguish between send and receive codecs""
> 
> This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> 
> Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> 
> Original change's description:
> > Reland "Distinguish between send and receive codecs"
> > 
> > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > 
> > Reason for revert: Fixed negotiation of send-only clients.
> > 
> > Original change's description:
> > > Revert "Distinguish between send and receive codecs"
> > >
> > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > >
> > > Reason for revert: breaks negotiation with send-only clients
> > >
> > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > >
> > > Original change's description:
> > > > Distinguish between send and receive codecs
> > > >
> > > > Even though send and receive codecs may be the same, they might have
> > > > different support in HW. Distinguish between send and receive codecs
> > > > to be able to keep track of which codecs have HW support.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > >
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30292}
> > 
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> > 
> > 
> > Bug: chromium:1029737
> > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30348}
> 
> TBR=steveanton@webrtc.org,kron@webrtc.org
> 
> Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30360}

TBR=steveanton@webrtc.org,kron@webrtc.org

Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30367}
 2020-01-23 23:02:59 +00:00
Johannes Kron
00a30873c4 Revert "Reland "Distinguish between send and receive codecs"" 
This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.

Reason for revert: Breaks Chromium import due to flaky test in Chromium.

Original change's description:
> Reland "Distinguish between send and receive codecs"
> 
> This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> 
> Reason for revert: Fixed negotiation of send-only clients.
> 
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> >
> > Reason for revert: breaks negotiation with send-only clients
> >
> > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> >
> > Original change's description:
> > > Distinguish between send and receive codecs
> > >
> > > Even though send and receive codecs may be the same, they might have
> > > different support in HW. Distinguish between send and receive codecs
> > > to be able to keep track of which codecs have HW support.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30284}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30292}
> 
> TBR=steveanton@webrtc.org,kron@webrtc.org
> 
> 
> Bug: chromium:1029737
> Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30348}

TBR=steveanton@webrtc.org,kron@webrtc.org

Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30360}
 2020-01-23 13:10:53 +00:00
Johannes Kron
133bf2bd28 Reland "Distinguish between send and receive codecs" 
This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.

Reason for revert: Fixed negotiation of send-only clients.

Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30292}

TBR=steveanton@webrtc.org,kron@webrtc.org


Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30348}
 2020-01-22 13:55:41 +00:00
Yves Gerey
100fe639b2 Fix race condition around rtc::ScopedFakeClock. 
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}
 2020-01-17 18:55:29 +00:00
Steve Anton
692f3c70e4 Explicitly wait for ICE to complete in VerifyBestConnection 
Bug: webrtc:11281
Change-Id: I94eeac3e08c1a2abc9057c5dad648e987f049c97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166402
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30290}
 2020-01-16 23:48:49 +00:00
Steve Anton
a9b67ceab6 Explicitly wait for ICE state to transition to 'completed' in VerifyIceStates 
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}
 2020-01-16 23:41:09 +00:00
Harald Alvestrand
2697ac1a1b Stop an SCTP connection when the DTLS transport closes. 
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}
 2019-12-16 14:54:56 +00:00
Harald Alvestrand
17ea068e8c Integration test that verifies that data channels open. 
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}
 2019-12-13 23:03:34 +00:00
Ying Wang
ef3998ffd1 Add directive to make webrtc metrics optional. 
Bug: webrtc:11144
Change-Id: I4e75e6aec033784685de3670e880bb9f2b6ee8d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30040}
 2019-12-09 13:55:50 +00:00
Eldar Rello
0095d37137 Replace hostCandidate with address and port in RTCPeerConnectionIceErrorEvent 
Bug: chromium:1013564
Change-Id: Ie1bb86ed6a2a7d73fe6ee666f973d809ed05a7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#30004}
 2019-12-04 13:18:22 +00:00
Bjorn A Mellem
7a9a092708 Delete media transport integration. 
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}
 2019-11-26 19:19:36 +00:00
Qingsi Wang
25ec8882f7 Make ICE transports injectable. 
Bug: chromium:1024965
Change-Id: I4961f50aee34c82701299f59a95cb90d231db6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158820
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29807}
 2019-11-15 21:31:19 +00:00