Commit Graph

534 Commits

Author SHA1 Message Date
7008287219 Delete struct webrtc::PacketTime.
Replaced by a int64_t representing time in us.

Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
2018-08-07 10:07:15 +00:00
e20867ff6d Add AsyncResolverFactory interface and basic implementation.
The factory is plumbed down to P2PTransportChannel and will eventually
be used to resolve hostnames. Uses of PacketSocketFacotry::CreateAsyncResolver
will eventually be migrated to use this factory instead.

Bug: webrtc:4165
Change-Id: I1c48b2ffb8649609a831eba291f67ce544bb10eb
Reviewed-on: https://webrtc-review.googlesource.com/91300
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24176}
2018-08-02 21:20:15 +00:00
7a1c7f782a Modified peerconnection's "observer" slot to be nulled on close.
This prevents usage of the observer post-close; modified the "usage
report notification" handler to not report when called post-close.
This fits the description of the original bug, so likely fixes it.

Bug: chromium:868337
Change-Id: Ic6757d2fb335203a6a6aacb2c9b52854b40332f7
Reviewed-on: https://webrtc-review.googlesource.com/91121
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24164}
2018-08-01 09:44:57 +00:00
8a3ab0e7ea Revert "Add framesRendered to StatsReport"
This reverts commit dcfa938f9e768d463d3e336f4d014027504267dd.

Reason for revert: This CL blocks rolling WebRTC into chromium

Original change's description:
> Add framesRendered to StatsReport
> 
> Bug: webrtc:9568
> Change-Id: I6976f4c48b67f6a81f57260a91966debbef38eb4
> Reviewed-on: https://webrtc-review.googlesource.com/90840
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24121}

TBR=steveanton@webrtc.org,solenberg@webrtc.org,joachimr@fb.com

Change-Id: Ia58feefd0ab557bb39ff79840dc8fa5004fee753
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9568
Reviewed-on: https://webrtc-review.googlesource.com/90900
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24134}
2018-07-27 14:53:07 +00:00
dcfa938f9e Add framesRendered to StatsReport
Bug: webrtc:9568
Change-Id: I6976f4c48b67f6a81f57260a91966debbef38eb4
Reviewed-on: https://webrtc-review.googlesource.com/90840
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24121}
2018-07-26 22:50:02 +00:00
01560dead9 Use "wildcard" instead of "unknown" for ADAPTER_TYPE_ANY in stats.
TBR=hta@webrtc.org

Bug: None
Change-Id: I3ccff01f2ad51aebc1241fab2b41518b769adc8a
Reviewed-on: https://webrtc-review.googlesource.com/90800
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24120}
2018-07-26 22:02:02 +00:00
c0e9725916 Add API to report "interesting" usage patterns to PC client
Bug: chromium:866792
Change-Id: Ic8bec5494aaa617c833c90be2b912f7367b44929
Reviewed-on: https://webrtc-review.googlesource.com/90246
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24111}
2018-07-26 09:12:03 +00:00
d9e4a06374 Add CreateSessionDescription overload which takes a cricket::SessionDescription
This gives clients a way to create a SessionDescriptionInterface
from a parsed cricket::SessionDescription other than depending on
JsepSessionDescription.

Bug: webrtc:9544
Change-Id: I3eec87b24aa005e6cbc4a018ad452c0d6823435d
Reviewed-on: https://webrtc-review.googlesource.com/90382
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24105}
2018-07-25 18:03:05 +00:00
e41c433502 Move sigslot to proper third_party directory
Extract sigslot into separate target and move it to proper third_party
directory.

Bug: webrtc:8366
Change-Id: Id2e0712bd020bfad811947803c94553dce06d976
Reviewed-on: https://webrtc-review.googlesource.com/84141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 14:53:33 +00:00
27ab0e5ee5 Add CreateIceCandidate overload which takes a cricket::Candidate
This gives clients a clear way to create an IceCandidateInterface
instance for use with PeerConnection from a parsed
cricket::Candidate structure.

Previously, the only way was with the JsepIceCandidate constructor,
but this CL will allow us to move that class out of the API.

Bug: webrtc:9544
Change-Id: Idfc1f1e0f5ee4c68d94599aae3fb824b23189a7c
Reviewed-on: https://webrtc-review.googlesource.com/90121
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24074}
2018-07-23 23:49:44 +00:00
ea1bb35e27 Cleanup networkroute.h
This change removes the constructors in favor of naming the fields
of the struct.

TBR=kwiberg@webrtc.org

Bug: None
Change-Id: I23ae1165c20994d2efef10184570065957b279af
Reviewed-on: https://webrtc-review.googlesource.com/90081
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24071}
2018-07-23 19:35:28 +00:00
24db573bea Step 1: Add RemoveTrackNew which returns an RTCError
Bug: webrtc:9534
Change-Id: I400bdcd0eb2e993b3f2252a2c7606cd105854e6b
Reviewed-on: https://webrtc-review.googlesource.com/89480
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24070}
2018-07-23 19:34:59 +00:00
a76af0ca2e Move base64.h to the proper location.
Move base64.h to the proper location and put redirect header into the
old place to be able to switch downstream users on new location.

Bug: webrtc:8366
Change-Id: I5191fe631d32178d2efd1315ca9abd4250102291
Reviewed-on: https://webrtc-review.googlesource.com/88223
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24069}
2018-07-23 15:40:36 +00:00
2ffed6d65c Enable clang::find_bad_constructs for sdk/android (part 1/2).
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.

Bug: webrtc:9251, webrtc:163, webrtc:9544
Change-Id: I7c211c4ac6b2e095e4c6594fce09fdb487bb1d9e
Reviewed-on: https://webrtc-review.googlesource.com/89600
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24056}
2018-07-20 21:35:40 +00:00
ee01a839d2 Remove MetricsObserverInterface.
The usage of MetricsObserverInterface to log metrics has been replaced
by RTC_HISTOGRAM_* macros in WebRTC.

Bug: webrtc:9409
Change-Id: I67df74a18942ac7ea4227e4affdf84f06258a287
Reviewed-on: https://webrtc-review.googlesource.com/86780
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24048}
2018-07-19 23:00:20 +00:00
76829d7c3d Add UMA metric for ICE candidate addition outcome
Bug: webrtc:9532
Change-Id: I58af94c03f5bbf25db2b558a8fe1ae53634fb99f
Reviewed-on: https://webrtc-review.googlesource.com/89063
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24032}
2018-07-19 02:23:59 +00:00
4f6d233dcc Added explicit EOR to sctp messages and coalesce messages on the receiving side.
TBR=pthatcher@webrtc.org

Bug: webrtc:7774
Change-Id: I41d1cd98d1e7b2ad479177eb2e328a5e2c704824
Reviewed-on: https://webrtc-review.googlesource.com/88900
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24031}
2018-07-19 01:26:59 +00:00
87b3c510b4 Implement changing degradation preference with setParameters()
The current default behavior is unchanged and points to MAINTAIN_FRAMERATE,
meaning there is no way to currently use BALANCED as we can't detect
when the value as been set or not.
Updating this is an API change that should be done in another CL and
properly communicated first.


Bug: webrtc:7607
Change-Id: Ic3877ad8dd7bc418296f21a04bc37f59ec55934a
Reviewed-on: https://webrtc-review.googlesource.com/88766
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24024}
2018-07-18 14:45:27 +00:00
0f5400acfa [Unified Plan] Implement FiredDirection for RtpTransceiver
Bug: webrtc:9236
Change-Id: Ib5a8215f3762f35b68d2a285c7d676f93f1212c5
Reviewed-on: https://webrtc-review.googlesource.com/88921
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24010}
2018-07-17 23:56:04 +00:00
056d811b6a Add counting of PCs with private IP addresses exposed
Bug: chromium:718508
Change-Id: I37f166808297c565cbb4b4393a23f7a18ab2862d
Reviewed-on: https://webrtc-review.googlesource.com/88640
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23990}
2018-07-16 18:04:09 +00:00
db90556972 Re-enable skipped test.
TBR=kwiberg@webrtc.org

Bug: webrtc:9442
Change-Id: I1cde15deac8202ce90c31578efd32f6cc4aabfca
Reviewed-on: https://webrtc-review.googlesource.com/88569
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23978}
2018-07-16 07:31:07 +00:00
7fc821d42d Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
This is a reland of 1a2cc0acba6a66f89249455d8e5775849b56f755

Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
>
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
>
> Original change's description:
> > Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
> >
> > We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> > to report the metrics in pc/ and p2p/ that are currently been reported
> > using MetricsObserverInterface.
> >
> > TBR=tommi@webrtc.org
> >
> > Bug: webrtc:9409
> > Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> > Reviewed-on: https://webrtc-review.googlesource.com/83782
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23914}
>
> TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}

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

Bug: webrtc:9409
Change-Id: Ib55f0b6c9bcb9d9585924a4dfac5cf643ff4d76b
Reviewed-on: https://webrtc-review.googlesource.com/88343
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23957}
2018-07-12 21:35:47 +00:00
4238628c30 Buffer ICE candidates that can't be added immediately.
This is required to make the test non-flaky.

Bug: webrtc:9494
Change-Id: Iae26028233fa4d990f082cbc1b023253e783ccc8
Reviewed-on: https://webrtc-review.googlesource.com/87438
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23951}
2018-07-12 12:46:30 +00:00
d78323faba Remove AddTrack override with MediaStreams
Bug: None
Change-Id: I992d356a7271fd89a174b0f458f9030092953b3e
Reviewed-on: https://webrtc-review.googlesource.com/88302
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23943}
2018-07-11 20:17:09 +00:00
065a52a655 Reland "Remove rtc::Optional alias and api:optional target"
This is an reland of 6f5b0f920af08d66e6b77ee4f91ade5797145368
Relanded after speculative revert without any changes.

TBR=ilnik@webrtc.org

Original change's description:
> Remove rtc::Optional alias and api:optional target
>
> Update left-overs where old target still was used.
>
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}

Bug: webrtc:9078
Change-Id: Ia33c6438253c6ec49f45d938e8a3607b51c418be
Reviewed-on: https://webrtc-review.googlesource.com/88160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23941}
2018-07-11 19:02:51 +00:00
70aa374d3e Remove not-updating-stats log message.
GetStats() is currently the only way to get smoothed audio levels. It's
expected to be called quite frequently. We use 100ms intervals, so a
call to GetStats() for any reason other than audio levels has a 50%
chance of triggering this log line. This makes it too noisy for LS_INFO.

The log line was added recently
(https://webrtc-review.googlesource.com/c/src/+/82260) and doesn't seem
very useful for diagnostic purposes, so remove it entirely.

Bug: webrtc:9519
Change-Id: I15700085c60b9929a4df2e2327012a4f16b505b6
Reviewed-on: https://webrtc-review.googlesource.com/88003
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23939}
2018-07-11 18:56:31 +00:00
78fef76e6a Revert "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
This reverts commit 1a2cc0acba6a66f89249455d8e5775849b56f755.

Reason for revert: It breaks internal Android debug build. Need further investigation.

Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
> 
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
> 
> Original change's description:
> > Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
> >
> > We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> > to report the metrics in pc/ and p2p/ that are currently been reported
> > using MetricsObserverInterface.
> >
> > TBR=tommi@webrtc.org
> >
> > Bug: webrtc:9409
> > Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> > Reviewed-on: https://webrtc-review.googlesource.com/83782
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23914}
> 
> TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
> 
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,tommi@webrtc.org,hta@webrtc.org,qingsi@google.com,qingsi@webrtc.org

Change-Id: I4a75fc7f52bfd0780526537a5a9a016fb9c20d6a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88320
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23938}
2018-07-11 18:37:36 +00:00
199e27bb63 RtpReceiverInterface::stream_ids() added.
This is the first step to removing streams from third_party/webrtc.
RtpReceiverInterface::streams() will have to be removed separately.
See https://crbug.com/webrtc/9480 for more information.

Bug: webrtc:9480
Change-Id: I6f9e6ddcda5e2245cc601d2cc6205b7b363f73ef
Reviewed-on: https://webrtc-review.googlesource.com/86840
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23929}
2018-07-11 10:14:56 +00:00
b661c658da Revert "Remove rtc::Optional alias and api:optional target"
This reverts commit 6f5b0f920af08d66e6b77ee4f91ade5797145368.

Reason for revert: Breaks internal project.

Original change's description:
> Remove rtc::Optional alias and api:optional target
> 
> Update left-overs where old target still was used.
> 
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}

TBR=danilchap@webrtc.org,kwiberg@webrtc.org

Change-Id: I95f5ec33520b823c3d0c9cb83d945d6a15355367
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9078
Reviewed-on: https://webrtc-review.googlesource.com/88140
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23921}
2018-07-11 07:41:41 +00:00
1a2cc0acba Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f

Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}

TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org

Bug: webrtc:9409
Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
Reviewed-on: https://webrtc-review.googlesource.com/88060
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23919}
2018-07-11 04:40:26 +00:00
13f4c896d5 Revert "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
This reverts commit 870bca1f418a1abf445169a638a61f9a649d557f.

Reason for revert: it breaks internal tests and builds

Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hta@webrtc.org,tommi@webrtc.org

Change-Id: I1afd92d44f3b8cf3ae9aa6e6daa9a3a272e8097f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23916}
2018-07-10 21:26:28 +00:00
870bca1f41 Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
to report the metrics in pc/ and p2p/ that are currently been reported
using MetricsObserverInterface.

TBR=tommi@webrtc.org

Bug: webrtc:9409
Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
Reviewed-on: https://webrtc-review.googlesource.com/83782
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23914}
2018-07-10 20:02:16 +00:00
6f5b0f920a Remove rtc::Optional alias and api:optional target
Update left-overs where old target still was used.

Bug: webrtc:9078
Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
Reviewed-on: https://webrtc-review.googlesource.com/84740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23913}
2018-07-10 18:02:23 +00:00
651b92e5d8 Regenerate mock peer connection to add missing mock methods.
Generated using gmock_gen.py with some editing.

This mock doesn't seem to be used by unittest in webrtc, but we need to use it in downstream unittests.

Bug: None
Change-Id: Ia7904ffdd22f3d16fe5fd515fa68833817b44481
Reviewed-on: https://webrtc-review.googlesource.com/85780
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23900}
2018-07-10 09:23:26 +00:00
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
e12c1fe8d9 Removing warning suppression flags from pc/.
Bug: webrtc:9251
Change-Id: Ic12126fc03309448fe71a17e6b65343949496f4f
Reviewed-on: https://webrtc-review.googlesource.com/86820
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23838}
2018-07-04 10:35:27 +00:00
43745937a8 Adding shampson (me) as an owner to pc/ & api/.
With deadbeef removed from these OWNERS files, Steve is the only OWNER
on our team. I'm adding myself, because I have worked in these
directories and it makes sense to be able to distribute the code
reviews.

NOTRY=True

Bug: None
Change-Id: I48e88a07ee42254d937391f500f273656853d98b
Reviewed-on: https://webrtc-review.googlesource.com/86980
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23826}
2018-07-03 20:39:17 +00:00
ec20710250 Adding ICE configurations to the PC perf test.
This adds multiple ICE configurations to the PeerConnection ramp up
performance test. The configurations added are:
-TLS TURN
-UDP TURN
-UDP peer to peer
-TCP peer to peer

Bug: webrtc:7668
Change-Id: If110d99e4d83b56ac093a1e43956292f1916a1bf
Reviewed-on: https://webrtc-review.googlesource.com/85140
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23824}
2018-07-03 19:45:27 +00:00
9f1de69008 Add ADAPTER_TYPE_ANY in AdapterType.
ADAPTER_TYPE_ANY can be used to set the network ignore mask if an
application does not want candidates from the any address ports, the
underlying network interface types of which are not determined in
gathering. The ADAPTER_TYPE_ANY is also given the maximum network cost
so that when there are candidates from explicit network interfaces,
these candidates from the any address ports as backups, if they ever
surface, are not preferred if the other candidates have at least the
same network condition.

Bug: webrtc:9468
Change-Id: I20c3a40e9a75b8fb34fad741ba5f835ecc3b0d92
Reviewed-on: https://webrtc-review.googlesource.com/85880
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23807}
2018-07-02 17:59:11 +00:00
dc99e244ca Removing deadbeef@ from OWNERS files.
Since I'm leaving Google.

Bug: None
Notry: True
Change-Id: Ibb5c3e09fce007d149200dcb6cac74be53084764
Reviewed-on: https://webrtc-review.googlesource.com/86461
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23795}
2018-07-02 00:40:38 +00:00
776199a55a Enable PeerConnectionEndToEndTest.CallWithLegacySdp on ASan.
Bug: None
Change-Id: I9f695bd0a13b0130f4d773803e010b69020c2ac1
Reviewed-on: https://webrtc-review.googlesource.com/86131
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23777}
2018-06-29 05:00:46 +00:00
82d171c824 Skip PeerConnectionEndToEndTest.CallWithCustomCodec on Win ASan builds.
Bug: None
Change-Id: Iaee0bdee03e23aae916a641c6230e14ae229c6df
Reviewed-on: https://webrtc-review.googlesource.com/86130
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23776}
2018-06-29 04:57:36 +00:00
b2a7478221 Fix usage logging of TURN and STUN servers
Also adds tests, and adds a bit of logging in ParseIceServers.

Bug: chromium:718508
Change-Id: Id41ccb7cccbdab5af76e380b32b4d8ba0c4a0a72
Reviewed-on: https://webrtc-review.googlesource.com/86121
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23769}
2018-06-28 12:52:07 +00:00
72b751af0b Add PeerConnection GetRtpSender/ReceiverCapabilities
Those are static functions in the spec, so implemented as member functions
of the PeerConnectionFactory instead.

Bug: webrtc:7577, webrtc:9441
Change-Id: Iccb24180e096e713d24e7e25ecfd5d7bbd7638f9
Reviewed-on: https://webrtc-review.googlesource.com/85341
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23768}
2018-06-28 12:40:07 +00:00
183e09d23c Correct data histogram entry for incoming DC
Both incoming and outgoing datachannels should cause
the DATA_ADDED flag to be set.

This CL also moves all tests into their own file, and
improves scaffolding.

Bug: chromium:718508
Change-Id: I5c4c257ccb6f26799f7593bce8b27ebf59015b1e
Reviewed-on: https://webrtc-review.googlesource.com/85348
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23766}
2018-06-28 10:33:23 +00:00
d1003d74b2 A new PeerConnection level perf test.
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.

Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-27 23:19:05 +00:00
07563732f6 [Unified Plan] Avoid offering two senders with the same ID
This can happen with the following sequence of API calls:
1) AddTrack(track) + offer/answer
2) RemoveTrack(track's sender) + offer/answer
3) AddTrack(same track)

Since the first transceiver had already been used to send, it will
not get re-used by the second call to AddTrack. Another RtpSender
will be created with its ID = the track ID. But the code hits a
DCHECK when CreateOffer is later called since both m= sections will
offer the same track ID component of the MSID.

The fix implemented here is to randomly generate a sender ID if
there is already an RtpSender with the track's ID.

Bug: webrtc:8734
Change-Id: Ic2dda23d66e364e77ff7505e1c37e53105a17dae
Reviewed-on: https://webrtc-review.googlesource.com/84249
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23748}
2018-06-26 19:06:17 +00:00
1bc9716078 [Unified Plan] Do not initialize recvonly transceivers with any send streams
Bug: None
Change-Id: Ie519a9ea3740f0b4fac97a4ffd486e7b4fa47cd9
Reviewed-on: https://webrtc-review.googlesource.com/84560
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23747}
2018-06-26 17:41:56 +00:00
111fdfd732 Refactor RtpSender to take the sender ID as a constructor argument
This change also standardizes the RtpSender to a single constructor
and moves the |track| and |stream_ids| arguments to setter methods.

Bug: webrtc:8734
Change-Id: I227a84868a80797f6cc2a1af6eec6d76da8ea159
Reviewed-on: https://webrtc-review.googlesource.com/84248
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23730}
2018-06-25 21:01:02 +00:00
d5b8ee1e17 Re-enable PeerConnectionEndToEndTest.Call on TSan.
Bug: webrtc:4719
Change-Id: Ic24c0921892a45bd28cd91f8ce6bdd9593ef1d59
Reviewed-on: https://webrtc-review.googlesource.com/85281
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23729}
2018-06-25 15:36:02 +00:00