883 Commits

Author SHA1 Message Date
Henrik Boström
f71362f0cf Wire up RTCOutboundRtpStreamStats.totalEncodeTime.
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/130517 that calculated
this metric.

This CL is purely plumbing, exposing
VideoSendStream::total_encode_time_ms in standard getStats() as
RTCOutboundRtpStreamStats.totalEncodeTime (in seconds):
https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime

Bug: webrtc:10448
Change-Id: I715f1ef937e441169dee55b5e8d4fbf98811c5f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131940
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27501}
2019-04-09 07:34:38 +00:00
Amit Hilbuch
07f3279a73 Adding a restriction for legal RID values.
According to the spec, RID values should be constrained to only
alpha-numeric values. This was not enforced in our implementation to
allow for more flexibility.
It has been brought to our attention that some values that we currently
consider legal (such as the '~', '=' ';' characters) might cause confusion
with the simulcast syntax that uses these characters to indicate other
meanings.
What's worse, is that some characters, when used in RIDs (such as
\u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
users which might also be a health hazard.
This change resolves these issues by restricting RIDs to alpha-numeric.

Bug: webrtc:10491
Change-Id: I16e262c87525d0289764beacd098e1525a355463
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27499}
2019-04-08 22:41:24 +00:00
Sebastian Jansson
c01367db40 Deprecating ThreadChecker specific interface.
All changes outside thread_checker.h are by:
s/CalledOnValidThread/IsCurrent/
s/DetachFromThread/Detach/

Bug: webrtc:9883
Change-Id: Idbb1086bff0817db58e770116acf4c9d60fae8b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27494}
2019-04-08 16:58:07 +00:00
Ruslan Burakov
1e2d436cbf Change PlayoutLatency setLatency zero-threshold value.
This is needed to match behaviour described in this spec:
https://github.com/henbos/webrtc-timing/pull/2

Bug: webrtc:10287
Change-Id: Idce9af2ec63705dfbfb500b7dbbf755ed3eab571
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131336
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#27480}
2019-04-08 10:27:06 +00:00
Ruslan Burakov
4bac79ece2 Add SetJitterBufferMinimumDelay method to RtpReceiverInterface.
This change is required to allow modification of Jitter Buffer delay
in javascript via Origin Trial Experiment.
Link to experiment description:
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/Tgm4qiNepJc

Bug: webrtc:10287
Change-Id: I4f21380aad5982a4a60c55683b5173ce72ce0392
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131144
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27444}
2019-04-04 09:00:16 +00:00
Karl Wiberg
739506e45e Add thread safety annotations for some more PeerConnection members (part 12)
Plus all the annotations that were necessary to make things compile
again. I also had to send copies of some values owned by the signal
thread to the network thread, instead of letting the latter read them
itself.

Bug: webrtc:9987
Change-Id: Ic4b38696245584bab44956e60ac63753146e3ff4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131020
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27437}
2019-04-03 10:28:54 +00:00
Harald Alvestrand
78a5e96001 Reland "Add thread guards to JsepTransport"
This reverts commit caedb5db82b2bc8273910f4a0d1afb1d0e2994f3.

Reason for revert: Fixed issue (allow SetNeedsIceRestart from off-thread).

Original change's description:
> Revert "Add thread guards to JsepTransport"
>
> This reverts commit 7e1db52c93c57a180073906eda6a58919a9fd537.
>
> Reason for revert: Breaks downstream.
>
> Original change's description:
> > Add thread guards to JsepTransport
> >
> > This ensures that JsepTransport's methods are either only accessed on the thread
> > that creates it, or using methods that are marked for off-thread use
> > (using a lock to prevent simultaneous access).
> >
> > The intent is to document the existing contract, and to make it easy to find the
> > actions needed to convert the class to a pure single-threaded class.
> >
> > Bug: webrtc:10300
> > Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27427}
>
> TBR=steveanton@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org
>
> Change-Id: I30c65d2161de9376ccd1172e2b261f2280fb1d75
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10300
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130519
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27429}

Change-Id: Ic32bfc04d96e657fc67c3d3999f77969e55ed994
Bug: webrtc:10300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130962
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27434}
2019-04-03 09:41:38 +00:00
Karl Wiberg
d9bf720c20 Add thread safety annotations for some more PeerConnection members (part 11)
After reviewer feedback, this CL was reduced to just adding scary
comments on two variables.

Bug: webrtc:9987
Change-Id: Id1e251ffd02e4ca8050235bd9f3971b5363f0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27431}
2019-04-03 09:05:41 +00:00
Gustaf Ullberg
caedb5db82 Revert "Add thread guards to JsepTransport"
This reverts commit 7e1db52c93c57a180073906eda6a58919a9fd537.

Reason for revert: Breaks downstream.

Original change's description:
> Add thread guards to JsepTransport
> 
> This ensures that JsepTransport's methods are either only accessed on the thread
> that creates it, or using methods that are marked for off-thread use
> (using a lock to prevent simultaneous access).
> 
> The intent is to document the existing contract, and to make it easy to find the
> actions needed to convert the class to a pure single-threaded class.
> 
> Bug: webrtc:10300
> Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27427}

TBR=steveanton@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org

Change-Id: I30c65d2161de9376ccd1172e2b261f2280fb1d75
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10300
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130519
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27429}
2019-04-03 07:15:38 +00:00
Harald Alvestrand
7e1db52c93 Add thread guards to JsepTransport
This ensures that JsepTransport's methods are either only accessed on the thread
that creates it, or using methods that are marked for off-thread use
(using a lock to prevent simultaneous access).

The intent is to document the existing contract, and to make it easy to find the
actions needed to convert the class to a pure single-threaded class.

Bug: webrtc:10300
Change-Id: Ib5cdc027632c36baec55179937d6eb664bbaf6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/121946
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27427}
2019-04-02 21:49:36 +00:00
Danil Chapovalov
9435c61021 Expose TaskQueueFactory for webrtc::Call in peer connection api
making a step for GlobalTaskQueueFactory to be optional way
to provide TaskQueueFactory

Bug: webrtc:10284
Change-Id: Ife838b3691c256820973118bc5b3cb372dea09cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130488
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27423}
2019-04-02 18:46:54 +00:00
Karl Wiberg
7a651c6e58 Add thread safety annotations for some more PeerConnection members (part 10)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I2b08c7db10dda7b18ad4ba036125f2a56ebf80a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130478
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27419}
2019-04-02 17:40:37 +00:00
Karl Wiberg
2cc368fd7a Add thread safety annotations for some more PeerConnection members (part 9)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: Ie958f4d86319e86527567ca1273a0595ccceee17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130490
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27411}
2019-04-02 10:48:16 +00:00
Mirko Bonadei
66e7679fb8 Export symbols needed by the Chromium component build (part 8).
This CL uses RTC_EXPORT (defined in rtc_base/system/rtc_export.h)
to mark WebRTC symbols as visible from a shared library, this doesn't
mean these symbols are part of the public API (please continue to refer
to [1] for info about what is considered public WebRTC API).

[1] - https://webrtc.googlesource.com/src/+/HEAD/native-api.md

Bug: webrtc:9419
Change-Id: Ib2c29054b2ae008f5291bd3b762a504b18534326
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27410}
2019-04-02 10:13:36 +00:00
Harald Alvestrand
5c4d2ee615 RTCDataChannel: Ignore "id" when "negotiated" is false
This updates behavior to be aligned with the WebRTC spec.

Bug: chromium:948055
Change-Id: Id3bbf05b3df084c9b7f7d12598c09187679d60fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130493
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27394}
2019-04-01 14:06:29 +00:00
Niels Möller
9d8eaac4ee Delete unneeded direct includes of common_types.h
And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:

api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/

There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.

Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
2019-04-01 07:18:13 +00:00
Harald Alvestrand
1f928d3316 Close data channels when ID assignment fails.
This prevents crashes due to unassigned IDs.

Bug: chromium:945256
Change-Id: I63f3a17cc7dff07dab58a6bc59fe3606b23e8e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129902
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27349}
2019-03-28 17:34:07 +00:00
Mirko Bonadei
e46f5db8bf Add missing using declarations for names in testing namespace.
This code was unnecessarily depending on ADL
(https://abseil.io/tips/49).

Bug: None
Change-Id: I4f130fbd46bf3c7cc3b4313c9c85f1ac9dc64cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129764
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27340}
2019-03-28 13:20:00 +00:00
Steve Anton
efe4c92d54 Use RtpSender/RtpReceiver track ID for legacy GetStats
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.

This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
   the unsigneled RtpReceiver track ID for both Plan B and Unified
   Plan.
2) Removes a couple methods on PeerConnection that were only used by
   the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
   the code easier to understand.

Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
2019-03-27 18:14:00 +00:00
Harald Alvestrand
b9bb3718e3 Revert "Don't crash when a datachannel can't get an ID"
This reverts commit 77c442ca1946924b0acfc9c0ba469ef6a2c3178f.

Reason for revert: Breakage on internal build tests, random breakage elsewhere. Timing issues suspected.

Original change's description:
> Don't crash when a datachannel can't get an ID
> 
> This is exercised by WPT test RTCDataChannel-id.
> 
> Bug: chromium:945256
> Change-Id: I53781dc874134f8c68a49c201848377b93b8858f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128871
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27300}

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

Change-Id: Ib7eaf8880e8ce21226b84b3b2283be93acb8dc8b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:945256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129766
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27302}
2019-03-27 07:23:41 +00:00
Harald Alvestrand
77c442ca19 Don't crash when a datachannel can't get an ID
This is exercised by WPT test RTCDataChannel-id.

Bug: chromium:945256
Change-Id: I53781dc874134f8c68a49c201848377b93b8858f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128871
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27300}
2019-03-26 22:06:24 +00:00
Danil Chapovalov
07122bc87e Use TaskQueueForTest instead or TaskQueue in unittests
To avoid hidden dependency on GlobalTaskQueueFactory used to construct TaskQueue

Bug: webrtc:10284
Change-Id: Iaa08be2827198e16aeb5538ea188d54cab60c1d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128879
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27291}
2019-03-26 14:42:49 +00:00
Karl Wiberg
a58e169269 Add thread safety annotations for some more PeerConnection members (part 8)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I452c17f52302fb28d37d9b570ef3b7ab3d023f77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129443
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27289}
2019-03-26 13:17:19 +00:00
Karl Wiberg
ac025898e1 Fix misunderstanding: OnTransportChanged is called on network thread
Earlier CLs assumed that the object pointed to by call_ had to be
accessed on the worker thread. While this is generally the case,
Call::MediaTransportChange is explicitly thread safe, so
PeerConnection::OnTransportChanged doesn't have to run on the worker
thread for that reason.

Which is fortunate, because it actually runs on the network thread.
The RTC_RUN_ON(worker_thread()) annotation on the method declaration
was ineffective because this method is being called via a base class
pointer; replacing it with a call to
RTC_DCHECK_RUN_ON(worker_thread()) in the function body immediately
triggered assertions in the unit tests.

Bug: webrtc:9987
Change-Id: I08cf558a74f4ca2b2eff8ef4810ebbd1287a9726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129442
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27287}
2019-03-26 12:51:34 +00:00
Karl Wiberg
12ba3adcaf Move unique_ptr into task instead of using a raw pointer
The raw pointer would have leaked if the task was ever destroyed
without being run.

Bug: webrtc:9987
Change-Id: Iddeb1adf0f836b8fec3056eab89bce7b9f034ca7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128865
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27284}
2019-03-26 11:28:25 +00:00
Karl Wiberg
6cab5c8718 Add thread safety annotations for some more PeerConnection members (part 5)
Plus all the annotations that were necessary to make things compile
again.

We needed a special twist for call_. The value it points to is owned
by the worker thread, but the signal thread needs to read the pointer.
We could have made the pointer const, except that we explicitly reset
it in the destructor (in an invoke to the worker thread).

Bug: webrtc:9987
Change-Id: I31f024547f4be0e50967133b0d452c80ae38d7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128863
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27278}
2019-03-26 09:35:20 +00:00
Niels Möller
0cd95081c0 Delete last traces of RtpTransportAdapter
Bug: None
Change-Id: I6bfac26ebd924c83f2f8a3adae983feb7d5bf00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128770
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27258}
2019-03-25 08:52:45 +00:00
Karl Wiberg
85a4a93e77 Add thread safety annotations for some more PeerConnection members (part 4)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: Ia421e4dc0e1bbc81c3976cc7530d44de934d33bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128882
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27257}
2019-03-23 06:54:31 +00:00
Karl Wiberg
c70999ed54 Add thread safety annotations for some more PeerConnection members (part 3)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: Ieb363d9ebb47658ecf9138552f44c5bcba6b9b80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128775
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27256}
2019-03-23 06:43:31 +00:00
Karl Wiberg
1e1e102380 Add thread safety annotations for some more PeerConnection members (part 2)
Plus all the annotations that were necessary to make things compile
again.

Bug: webrtc:9987
Change-Id: I7bd9793eb2d474f2ac7ce9e1ed590e67cc2e0a93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128881
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27255}
2019-03-23 06:40:11 +00:00
Karl Wiberg
fb3be3948d Add thread safety annotations for some more PeerConnection members
Plus all the annotations that were necessary to make things compile
again.

port_allocator_flags_ was accessed on both the signaling and the
network thread, but I was able to replace it with a return value.

Bug: webrtc:9987
Change-Id: Iab977a49d6588ce2240487475ec3588ae579caa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128772
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27254}
2019-03-23 06:14:11 +00:00
Philipp Hancke
fe0b499634 legacy stats: update timestamp on localcandidate/remotecandidate
updates the timestamp on the local and remote candidate stats for consistency
with other places. This also makes the graphs on chrome://webrtc-internals
work (even though most values don't update so showing graphs is not meaningful)

BUG=chromium:937833

Change-Id: I3267dd7a5f5a887dcd0756137077b8f02c201905
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128765
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27248}
2019-03-22 17:26:20 +00:00
Steve Anton
ddb930a83b Update per-file OWNERS to reflect renamed file names
Bug: None
No-Try: True
Change-Id: I2404bd14296286a832943226c20e947d1efb73fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128920
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27246}
2019-03-22 16:47:56 +00:00
Artem Titov
741daaf039 Move rtc::FunctionView to the public API
Bug: webrtc:10138
Change-Id: Icc25a2a277a9608701aaddd546882366739991ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127898
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27227}
2019-03-21 15:23:05 +00:00
Artem Titov
94b57c044e Cleanup BUILD.gn files from imports like foo:foo
Repalce all occurrences of foo:foo in deps with just foo in BUILD.gn
files.

Done with Sublime regex replace.
Find: \b([-a-zA-Z0-9_]+):+\1\b
In: *.gn
Replace with: \1

Bug: None
Change-Id: I40aba1b14face687a595b852ffe443cb20197611
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127899
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27225}
2019-03-21 13:05:28 +00:00
Niels Möller
ef1052a134 Reland "Move api/rtp_headers.h to its own build target."
This is a reland of a67050debcb5a3461a452a7928d7aaea1562747e

Original change's description:
> Move api/rtp_headers.h to its own build target.
>
> Reduces dependencies on the libjingle_peerconnection_api target from
> lower-level code.
>
> Bug: None
> Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27213}

Bug: None
Tbr: kwiberg@webrtc.org
Change-Id: If15b05957e50bb8f18a33c2ed1321e672311b626
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127895
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27216}
2019-03-21 09:17:07 +00:00
Steve Anton
2baef3509f Revert "Move api/rtp_headers.h to its own build target."
This reverts commit a67050debcb5a3461a452a7928d7aaea1562747e.

Reason for revert: breaks downstream projects

Original change's description:
> Move api/rtp_headers.h to its own build target.
> 
> Reduces dependencies on the libjingle_peerconnection_api target from
> lower-level code.
> 
> Bug: None
> Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27213}

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

Change-Id: I8cccaa8be1700ca8db141db7252eb6ce588ba2e0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128645
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27215}
2019-03-20 16:47:30 +00:00
Niels Möller
a67050debc Move api/rtp_headers.h to its own build target.
Reduces dependencies on the libjingle_peerconnection_api target from
lower-level code.

Bug: None
Change-Id: I98576fc718c396cc0f720c3770acd2b696b9df89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128565
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27213}
2019-03-20 16:00:49 +00:00
tzik
4e2d76c2cd Replace deprecated std::not2 with a lambda
std::not2 is deprecated in C++17, and that starts failing on C++17 mode
of ios_simulator build. This CL replaces it with a lambda to avoid the
warning.

Bug: chromium:752720
Change-Id: Id7ef847df0fbe0c44583ef3320e06f44644de929
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128620
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taiju Tsuiki <tzik@chromium.org>
Cr-Commit-Position: refs/heads/master@{#27198}
2019-03-20 06:10:31 +00:00
Danil Chapovalov
f5258be6f4 Make PeerConnectionFactory constructor taking dependency the only constructor
Bug: None
Change-Id: I19e9fab1ecec3799cc7b8573ab3fd6b258114cce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128601
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27194}
2019-03-19 19:37:15 +00:00
Amit Hilbuch
edd2054562 Minor fixes and refactoring for RtpTransport until the Demux.
This change fixes some inefficiencies and quirks in the code that
originates in RtpTransport leading up to the demux.

This work is in preparation for more refactoring of the Demux stage
onwards.

Bug: webrtc:10297
Change-Id: I7b8f00134657d62c722939618a55a91a2b6040bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128220
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27185}
2019-03-19 16:48:47 +00:00
Niels Möller
db4def9f59 Update parsing of stun and turn urls for RFC 7064-7065
Main change is deleting support for @userinfo in turn urls. This was
specified in early internet drafts, but never made it into RFC 7065.

Bug: webrtc:6663, webrtc:10422
Change-Id: Idd315a9e6001326f3104be62be3bd0991adc7db4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128423
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27171}
2019-03-19 08:13:13 +00:00
Piotr (Peter) Slatala
7fbfaa49d2 PeerConnection::SetBitrate now also configures media transport.
(so far SetBitrate did not do anything for media transport)

Bug: webrtc:9719
Change-Id: I48e669341ffe6c9e4697ff9146c314be7796a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27169}
2019-03-18 19:38:21 +00:00
Amit Hilbuch
e7a5f7bfae Modifying MediaChannel to accept CopyOnWriteBuffer by value.
MediaChannel accepted the RtpPacket buffers through non-const pointer.
This is both unclear and introduces questions regarding if the buffer is
actually copied or not.
This change modifies the method to accept by value to reduce ambiguity.
Usage of the non-const data() method which could potentially copy the
buffer contents is also reduced in favor of cdata() which never copies.

Bug: None
Change-Id: I3b2daef0d31cb6aacceb46c86da3a40ce836242b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127340
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27090}
2019-03-12 23:49:57 +00:00
Sebastian Jansson
d155d686f8 Removes rtp level keep alive support.
This is not used in practice as there's functionality on
other levels that serves the same purpose.

Bug: None
Change-Id: I0488dc42459b07607363eba0f2b06f4c50f7cda4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27061}
2019-03-11 14:47:15 +00:00
Marina Ciocea
e448a3fb54 Update DataChannel bufferedamount implementation.
Call DataChannelObserver::OnBufferedAmountChange on each successful send.
Previously, the observer would get notified of buffered amount changes only when
queued send data is consumed. Data gets queued only if it cannot be sent right
away. According to the WebRTC standard[1], bufferedamount should be increased
before each sent and decreased after each successful sent. Update implementation
to be standard compliant.

Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90

[1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount

Bug: chromium:878682
Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27057}
2019-03-11 13:21:36 +00:00
Johannes Kron
8cc711a7e1 Update URI of TransportSequenceNumberV2
The previous URI was a placeholder and is not valid. The URI
https://webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02/
must be used instead.

Bug: webrtc:10264
Change-Id: Ibabde599b5bbd116c1c5e86ba0c9c64019bf7026
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126360
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27051}
2019-03-11 10:15:52 +00:00
Niels Möller
abea6e5114 Delete always-true member is_media_transport_factory_enabled_
Member was added to JsepTransportController in
https://webrtc-review.googlesource.com/c/119911, but only code path
setting it to false was deleted in
https://webrtc-review.googlesource.com/c/125040

Bug: webrtc:9719
Change-Id: I9d2c5f338dfc30a769ed54d64e7f5bf27c230e31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126521
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27048}
2019-03-11 08:04:09 +00:00
Danil Chapovalov
4423c36448 Migrate RepeatingTask to take raw pointer to TaskQueueBase instead of TaskQueue
In particular replace call rtc::TaskQueue::Current with TaskQueueBase::Current

Bug: webrtc:10191
Change-Id: I19d42a716d27f0aba087dc70ac65b4ee6249408f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125085
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27005}
2019-03-06 18:44:35 +00:00
Jakob Ivarsson
232b3fda92 Expose relative packet arrival delay metric in stats API.
The metric is non-standard and documented in: https://github.com/henbos/webrtc-provisional-stats/pull/14

Bug: webrtc:10333
Change-Id: Ie5b4bbad5b1e2c9104742931529bab8f48f51f8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26999}
2019-03-06 16:35:16 +00:00