Commit Graph

572 Commits

Author SHA1 Message Date
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The usage of MetricsObserverInterface to log metrics has been replaced
by RTC_HISTOGRAM_* macros in WebRTC.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
1979384e40 Ensure that PC usage is recorded if a PC is alive for 60 seconds.
Bug: chromium:718508
Change-Id: Id2cbcb370b56cb8a6a6c821e0f89c51089cc8e6b
Reviewed-on: https://webrtc-review.googlesource.com/83140
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23723}
2018-06-25 10:25:38 +00:00
b6b29e0718 Convert video quality test from a TEST_F to a TEST fixture.
The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:

- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h

The following things are moved to API:

- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)

These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.

This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.

Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
2018-06-21 15:49:43 +00:00
c19ab07134 Add support for content-hint value "text"
This involves treating it just like "detailed", for now.
At a later stage we might want to modify codec parameters for it.

Bug: chromium:852701
Change-Id: I24678e1f7711bf03ca22273afaaf338e9e3ba1fe
Reviewed-on: https://webrtc-review.googlesource.com/83582
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Peter Boström <pbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23701}
2018-06-21 11:29:20 +00:00
b983bae923 Remove unused/deprecated DTMF methods
PeerConnectionInterface::CreateDtmfSender
DtmfSenderInterface::track

Bug: webrtc:9426
Change-Id: I7d151d8e0bdd60750ed60466083245631d540a91
Reviewed-on: https://webrtc-review.googlesource.com/84244
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23690}
2018-06-20 21:00:10 +00:00
beb2d9813c Removing usage of //build/config/compiler:no_size_t_to_int_warning.
Bug: webrtc:9251, webrtc:1348
Change-Id: I76e52abbfab5666cad73044b49172a9799539108
Reviewed-on: https://webrtc-review.googlesource.com/84144
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23686}
2018-06-20 13:44:26 +00:00
de212ca039 Removing some MSVC warning suppression flags.
Bug: webrtc:9251
Change-Id: Idf13b49648459a37fe0a3cac12ff993ce27439d9
Reviewed-on: https://webrtc-review.googlesource.com/84281
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23685}
2018-06-20 12:41:46 +00:00
5565981e17 Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters.
Target bitrate is set to 0.75 of the max bitrate.

Bug: webrtc:9341, webrtc:8655
Change-Id: I9a8c8bb95bb1532d45f05578832418464452340e
Reviewed-on: https://webrtc-review.googlesource.com/79821
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23676}
2018-06-20 07:26:09 +00:00
1d4a76da0a Fixing flakiness in PeerConnectionIntegrationTest.
EndToEndConnectionTimeWithTurnTurnPair was failing intermittently due to
a DCHECK being hit in ports.cc. This was caused by the ScopedFakeClock
being destroyed before the ports. The ports miscalculated a large
negative number for the rtt of a STUN request/response due to the global
clock changing. This fixes the problem by closing the PeerConnections
before the ScopedFakeClock goes out of scope.

Bug: webrtc:9422
Change-Id: Ia4aa3f638dff5da4317a35cf1514ec61472d0d74
Reviewed-on: https://webrtc-review.googlesource.com/84241
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23670}
2018-06-19 22:34:53 +00:00
66cadcc6b9 Replace rtc::Optional with absl::optional in pc
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'pc'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
2018-06-19 20:55:07 +00:00
a465344e39 Return SSRC stats with the old stats API when SSRCs are unsignaled.
This is the simplest possible fix, returning SSRC stats with a missing
track ID instead of returning no SSRC stats at all.

This means calling GetStats with the track selector argument will still not
work in this case.

Bug: webrtc:3342
Change-Id: I6b58fd5ac15b49274d3f1655e78ae36c4575e5fd
Reviewed-on: https://webrtc-review.googlesource.com/82260
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23667}
2018-06-19 17:28:25 +00:00
665174fdbb Reformat the WebRTC code base
Running clang-format with chromium's style guide.

The goal is n-fold:
 * providing consistency and readability (that's what code guidelines are for)
 * preventing noise with presubmit checks and git cl format
 * building on the previous point: making it easier to automatically fix format issues
 * you name it

Please consider using git-hyper-blame to ignore this commit.

Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
2018-06-19 14:00:39 +00:00
6bbeb080b8 Extract rtc_base/base64.h and rtc_base/base64.cc into separate target.
Extract rtc_base/base64.h and rtc_base/base64.cc into separate target
to prepare to move them into third_party

Bug: webrtc:8366
Change-Id: I477e6da2b9d09307439b3272261f31042f479d74
Reviewed-on: https://webrtc-review.googlesource.com/83980
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23645}
2018-06-18 16:44:47 +00:00