Commit Graph

556 Commits

Author SHA1 Message Date
3bc0166a4e getStats: add kind alias for mediaType
see https://github.com/w3c/webrtc-stats/issues/301

IDL: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats
Change-Id: I7da443bd1cbf07c9a3118ac04329db28b3543b3f

BUG=webrtc:9674

Change-Id: I7da443bd1cbf07c9a3118ac04329db28b3543b3f
Reviewed-on: https://webrtc-review.googlesource.com/96420
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24532}
2018-09-03 11:49:30 +00:00
16e27a1dc5 Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
Original cl: https://webrtc-review.googlesource.com/95721

Bug: webrtc:9239
Change-Id: I7eac85839182bbcecd0d9bd71ae26f6a1c516df4
Reviewed-on: https://webrtc-review.googlesource.com/96401
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24529}
2018-09-03 09:00:01 +00:00
9a4fd9bf74 Use AsyncInvoker in DtmfSender instead of MessageHandler
Bug: webrtc:9702
Change-Id: Ib9a9a2cf5bbb7aff24e6690deca51a021961ead3
Reviewed-on: https://webrtc-review.googlesource.com/97182
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24518}
2018-08-31 22:49:26 +00:00
044a04d8b5 Use AsyncInvoker in DataChannel instead of MessageHandler
Bug: webrtc:9702
Change-Id: I76a6a97f792be632c1c2f4f5cbbd26a7ec243006
Reviewed-on: https://webrtc-review.googlesource.com/97183
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24517}
2018-08-31 22:48:20 +00:00
d25828a0bf Use AsyncInvoker in JsepTransportController instead of MessageHandler
Bug: webrtc:9702
Change-Id: I9171d6e7f16fe50be1c2b139bf7dd1d097000791
Reviewed-on: https://webrtc-review.googlesource.com/97181
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24516}
2018-08-31 22:12:45 +00:00
bb19276a32 Use AsyncInvoker in PeerConnection instead of MessageHandler
Bug: webrtc:9702
Change-Id: I89d66d1165a096601aed37b8febad60620073899
Reviewed-on: https://webrtc-review.googlesource.com/97180
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24515}
2018-08-31 20:10:30 +00:00
d81ac953aa Injects FrameEncryptorInterface into RtpSender.
This change injects the FrameEncryptorInterface and the FrameDecryptorInterface
into the RtpSenderInterface and RtpReceiverInterface respectively. This is the
second stage of the injection. In a follow up CL non owning pointers to these
values will be passed down into the media channel.

This change also updates the corresponding mock files.

Bug: webrtc:9681
Change-Id: I964084fc270e10af9d1127979e713493e6fbba7d
Reviewed-on: https://webrtc-review.googlesource.com/96625
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24489}
2018-08-30 00:33:54 +00:00
d8111e2169 Delete unused class MockPeerConnection.
It appears to be unused since cl
https://webrtc-review.googlesource.com/46940.

Also note that there's a recently added class
MockPeerConnectionInterface, under api/test/, which may be more
suitable for new tests.

Bug: None
Change-Id: I6cd9bd2ec8847605f478663b709cd80c54895707
Reviewed-on: https://webrtc-review.googlesource.com/95421
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24459}
2018-08-27 20:02:14 +00:00
ec4a060a55 Revert "Delete leftover includes and declarations for MediaConstraintsInterface"
This reverts commit a1e4ae23715867eca58488be307759ffa5901463.

Reason for revert: Breakage in downstream code still using constraints.

Original change's description:
> Delete leftover includes and declarations for MediaConstraintsInterface
> 
> Bug: webrtc:9239
> Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
> Reviewed-on: https://webrtc-review.googlesource.com/95721
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24442}

TBR=kwiberg@webrtc.org,nisse@webrtc.org,hta@webrtc.org

Change-Id: Idbef4c57a0d3b82e94a431c5407a86c9fcd4be41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9239
Reviewed-on: https://webrtc-review.googlesource.com/96160
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24444}
2018-08-27 11:26:42 +00:00
a1e4ae2371 Delete leftover includes and declarations for MediaConstraintsInterface
Bug: webrtc:9239
Change-Id: I1f49d6847572faecd6022a44222d6271302fe443
Reviewed-on: https://webrtc-review.googlesource.com/95721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24442}
2018-08-27 10:41:57 +00:00
fc1acd2364 Add support for enabling simulcast in "Plan B" using MediaConstraints.
BUG=webrtc:9655

Change-Id: Ieb5fe5d97b6d4381608a51593bca5423979d1b9f
Reviewed-on: https://webrtc-review.googlesource.com/95481
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24424}
2018-08-24 09:55:59 +00:00
6fcdc2f708 Support domain name ICE candidates
Bug: webrtc:4165
Change-Id: Icc06bb13120080635cb722b8a8720e7d25426e2d
Reviewed-on: https://webrtc-review.googlesource.com/85540
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24415}
2018-08-24 04:54:43 +00:00
e4749c2bf0 The default logic for creating video bitrate allocator.
It is a mirror of `VideoCodecInitializer::CreateBitrateAllocator`

Bug: webrtc:9513
Change-Id: Ib2e83e9f757387a2f6f6101d5d21512f1d507a95
Reviewed-on: https://webrtc-review.googlesource.com/92320
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24413}
2018-08-23 20:50:32 +00:00
f06f923ef0 Delete almost all use of MediaConstraintsInterface in the PeerConnection API
Bug: webrtc:9239
Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b
Reviewed-on: https://webrtc-review.googlesource.com/74420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24396}
2018-08-23 07:14:37 +00:00
6b1985de95 Reimplement rtc::ToString and rtc::FromString without streams.
Bug: webrtc:8982
Change-Id: I3977435b035fdebef449732301d6e77fc899e7ba
Reviewed-on: https://webrtc-review.googlesource.com/86941
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24319}
2018-08-16 16:14:01 +00:00
cdb87f1a65 Add a missing #include from jsepicecandidate.cc to absl/memory/memory.h
jsepicecandidate.cc uses absl::make_unique without including
absl/memory/memory.h. This happens to work in C++14 due to a transitive
#include, but doesn't work in C++17.

Bug: chromium:752720
Change-Id: I995496f452b9eaa2e70b82cd3b7926b936d7dac0
Reviewed-on: https://webrtc-review.googlesource.com/94340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24294}
2018-08-15 15:19:07 +00:00
6f80f098ec Adds the custom TLS certificate verifier pointer to the reconfigure option as
well. This allows the verifier to be attached at a later point after ice
candidates.

Bug: webrtc:9623
Change-Id: I06f31256c494f6a790c6047e8602b8665dfe2f7e
Reviewed-on: https://webrtc-review.googlesource.com/93943
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24280}
2018-08-14 17:58:24 +00:00
948b7e3755 Revert "Add initial support for RtpEncodingParameters max_framerate."
This reverts commit ced5cfdb35a20c684df927eab37e16d35979555f.

Reason for revert: Breaks downstream project.

Original change's description:
> Add initial support for RtpEncodingParameters max_framerate.
> 
> Add support to set the framerate to the maximum of |max_framerate|.
> Different framerates are currently not supported per stream for video.
> 
> Bug: webrtc:9597
> Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
> Reviewed-on: https://webrtc-review.googlesource.com/92392
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24270}

TBR=steveanton@webrtc.org,magjed@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: I508fe48e0c53996654f657357913ac307dc256bd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9597
Reviewed-on: https://webrtc-review.googlesource.com/94060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24277}
2018-08-14 07:25:23 +00:00
b336c2784f Implement serialization for ICE candidates with hostname addresses.
Bug: webrtc:4165
Change-Id: I5ba0f25e458013ac3982648fc33d92d2a00e8fdd
Reviewed-on: https://webrtc-review.googlesource.com/93250
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24275}
2018-08-13 17:39:58 +00:00
ced5cfdb35 Add initial support for RtpEncodingParameters max_framerate.
Add support to set the framerate to the maximum of |max_framerate|.
Different framerates are currently not supported per stream for video.

Bug: webrtc:9597
Change-Id: Ie326617b66bd97be387f809a7f82b97b8f3ff5fe
Reviewed-on: https://webrtc-review.googlesource.com/92392
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24270}
2018-08-13 09:59:04 +00:00
656d609a95 Add UTC time to init event in AEC debug dump.
Bug: webrtc:9616
Change-Id: I1350212f0b8835fb64427483269da96d51670c01
Reviewed-on: https://webrtc-review.googlesource.com/92620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24267}
2018-08-11 20:29:07 +00:00
f820a5ea1c Fix use after move in SafeSetError
There is not actually any noticeable bug with the code as it was
since the RTCError move operators don't reset the type of the moved
object. But the clang static analyzer complains about this and it's
bad practice.

Bug: webrtc:9593
Change-Id: I8c04f193d10733371e0125c5349f9798f916eecf
Reviewed-on: https://webrtc-review.googlesource.com/93500
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24266}
2018-08-10 20:13:26 +00:00
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