Commit Graph

580 Commits

Author SHA1 Message Date
9551375c02 getStats: add relayProtocol
adds relayProtocol stats member.

BUG=webrtc:7063

Change-Id: Iedef61506cac1ab2e3e38c836881748965eeda3d
Reviewed-on: https://webrtc-review.googlesource.com/97780
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#24923}
2018-10-02 08:43:06 +00:00
892acf01f6 Add support for send_encodings parameters in addTransceiver
This will later allow simulcast to be set up without any SDP
manipulation. Currently limited to only one layer as the SDP
generated is not spec compliant and more work is required
to support simulcast.

Initial encoding parameters are deferred and applied when the ssrc
is set on the sender. This allows parameters to be changed before
negotiation is completed.

Bug: webrtc:7600
Change-Id: I0a31cd1c2bfc72ebb61ce0fa4fa69d87e3d8b74d
Reviewed-on: https://webrtc-review.googlesource.com/95488
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24917}
2018-10-01 22:56:30 +00:00
e0d455b409 Remove runtime_enabled_feature.
This features is not needed anymore, with this CL it is also possible
to address two issues:
- The need to pick a default implementation.
- The need to use -Wno-global-constructors.

Bug: webrtc:9631, webrtc:9693
Change-Id: Id3daf34179fbc8db26969fc701ccbfa7182c6a9b
Reviewed-on: https://webrtc-review.googlesource.com/102543
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24904}
2018-10-01 07:03:25 +00:00
07ba2b9445 Parse two-byte header extensions.
Bug: webrtc:7990
Change-Id: I967d2065b85d6a2ca938ac0e83035cb92b45a907
Reviewed-on: https://webrtc-review.googlesource.com/98160
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24881}
2018-09-28 08:32:17 +00:00
17f4878419 Remove deprecated field_trial_default and metrics_default.
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default

It also refreshes all the dependencies on field_trial.h and metrics.h.

A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm

Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 07:21:07 +00:00
49ac5959c2 Add GetSources to VideoRtpReceiver
BUG=webrtc:9770

Change-Id: I16143fce6eb727bbab0f6c621aa5b51bc6d28d6b
Reviewed-on: https://webrtc-review.googlesource.com/101600
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24858}
2018-09-27 10:00:40 +00:00
965e7942a3 Add sanity checks to UpdateDelayStatistics and patch unit tests.
RtpPacket::UpdateDelayStatistics was previously optimized with several
sanity checks added. These sanity checks caused many of the unit tests
in peerconnection_integration_unittests to fail and the CL was therefore
reverted. This CL contains the sanity checks along with patches so that
the unit tests pass.

Bug: webrtc:9439
Change-Id: Ia5f5e8b125e5f3f4b79d433e2282901143530a25
Reviewed-on: https://webrtc-review.googlesource.com/99802
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24813}
2018-09-24 23:13:02 +00:00
84df1c724e Make fewer copies when using StringBuilder.
Replace calls to .str() which copies with .Release which moves in cases where that's safe.

This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"

Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
2018-09-24 09:39:19 +00:00
8c1bf9595a Reland "Add initial support for RtpEncodingParameters max_framerate."
This reverts commit 948b7e37557af68b3bc9b81b29ae2daffb2784ad.

Reason for revert: downstream project fixed.

Original change's description:
> 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}

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

Bug: webrtc:9597
Change-Id: Ieed9d62787f3e9dcb439399bfe7529012292381e
Reviewed-on: https://webrtc-review.googlesource.com/100080
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24720}
2018-09-13 10:06:33 +00:00
941a07cca3 Remove all remaining non-test uses of std::stringstream.
Bug: webrtc:8982
Change-Id: I635a8545c46dc8c89663d64af351e22e65cbcb33
Reviewed-on: https://webrtc-review.googlesource.com/98880
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24715}
2018-09-13 08:52:05 +00:00
9c147ddc91 Revert "Add SSLConfig object to IceServer."
This reverts commit 4f085434b912060874d6697f17aaedd2adae7c49.

Reason for revert: breaks downstream projects.

Original change's description:
> Add SSLConfig object to IceServer.
> 
> This is a rollforward of https://webrtc-review.googlesource.com/c/src/+/96020,
> with the addition of setting the old tlsCertPolicy, tlsAlpnProtocols and
> tlsEllipticCurves in the RTCIceServer initializer, for backwards compatibility.
> 
> Bug: webrtc:9662
> Change-Id: I28706ed4ff5abe3f7f913f105779f0e5412aeac5
> Reviewed-on: https://webrtc-review.googlesource.com/98762
> Commit-Queue: Diogo Real <diogor@google.com>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24696}

TBR=steveanton@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,kthelgason@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org,diogor@google.com

Change-Id: I1cb64b63fec688b4ac90c2fa368eaf0bc11046af
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9662
Reviewed-on: https://webrtc-review.googlesource.com/99880
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24701}
2018-09-12 10:46:04 +00:00
4f085434b9 Add SSLConfig object to IceServer.
This is a rollforward of https://webrtc-review.googlesource.com/c/src/+/96020,
with the addition of setting the old tlsCertPolicy, tlsAlpnProtocols and
tlsEllipticCurves in the RTCIceServer initializer, for backwards compatibility.

Bug: webrtc:9662
Change-Id: I28706ed4ff5abe3f7f913f105779f0e5412aeac5
Reviewed-on: https://webrtc-review.googlesource.com/98762
Commit-Queue: Diogo Real <diogor@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24696}
2018-09-11 23:28:46 +00:00
bfd412ef71 Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
This change passes a pointer (non-owning) down to the MediaChannel when set
in the RtpSender / RtpReceiver. This currently is not used to encrypt frames.

Bug: webrtc:9681
Change-Id: I385fa8b948427803cd3f9cef918c31d7754d1b4f
Reviewed-on: https://webrtc-review.googlesource.com/97000
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24694}
2018-09-11 20:10:44 +00:00
ab49982601 Fix no_exit_time_destructors in pc.
This CL fixes the following error:
pc/peerconnection.cc:396:7:
error: declaration requires an exit-time destructor
[-Werror,-Wexit-time-destructors]
      proto_media_counter_map = {

It moves the protocol to media map into PeerConnection's attributes, the
map is initialized during PeerConnection::Initialize.
This removes the need of using 'static' and it should not cause too much
overhead since the map is initialized only once for each PeerConnection.

Bug: webrtc:9693
Change-Id: Icd71a70204ccc6fb032af52c64afa59e9aa7af74
Reviewed-on: https://webrtc-review.googlesource.com/98780
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24674}
2018-09-11 09:32:14 +00:00
3a66edf3c3 Use C++11 for range loop in pc/mediasession.cc (and test)
Bug: webrtc:9732
Change-Id: I1fad3313c5ad627f7eca52ea907608d67af6891f
Reviewed-on: https://webrtc-review.googlesource.com/98924
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24664}
2018-09-10 20:27:34 +00:00
211856b956 Make HasAttribute handle partial matching of attribute names.
Improve HasAttribute to handle the case where the beginning of an
attribute name is also an attribute name in it self. Two attributes
that have this relation are extmap-allow-mixed and extmap.

Bug: webrtc:9712
Change-Id: Iee660cc6e3dc7f2e7c56664a4f0ffb298eca9208
Reviewed-on: https://webrtc-review.googlesource.com/97422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24640}
2018-09-10 08:43:52 +00:00
4daf66e71e Use C++11 style for loop in webrtcsdp.cc
Bug: webrtc:9732
Change-Id: I76eda7cca0aab6989bea819bfff4e06034c399f7
Reviewed-on: https://webrtc-review.googlesource.com/98922
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24636}
2018-09-07 23:17:04 +00:00
d7b79af9df Add "tones remaining" argument to DTMF ontonechange callback
Bug: webrtc:9725
Change-Id: I2ad3e57d7357a9bd7cfbfa675df36ec66ff7c851
Reviewed-on: https://webrtc-review.googlesource.com/98361
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@{#24633}
2018-09-07 17:29:37 +00:00
366a50c4ef Remove simple stringstream usages.
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.

The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.

Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
2018-09-06 12:53:19 +00:00
96ede16a4e Enable -Wexit-time-destructors and -Wglobal-constructors.
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.

It also adds the possibility to suppress these warnings because
they trigger in a few places.

The long term goal is to avoid regressions on this and remove all the
suppressions.

Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
2018-09-06 12:43:20 +00:00
ad18276754 Revert "Use AsyncInvoker in PeerConnection instead of MessageHandler"
This reverts commit bb19276a325a5f9fce4afa245aa14ec2a4b1a41d.

Reason for revert: breaks downstream project

Original change's description:
> 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}

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

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

Bug: webrtc:9702
Change-Id: Ibfe507cd1593f7000e11f9a17313a016307381cb
Reviewed-on: https://webrtc-review.googlesource.com/98302
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24591}
2018-09-05 21:12:18 +00:00
ec9e4923db Remove stringstreams from sdp parsing.
Bug: webrtc:8982
Change-Id: Id5e8aa22794206864c8ce4b1ba155e49606a1b2d
Reviewed-on: https://webrtc-review.googlesource.com/97360
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24564}
2018-09-05 08:16:40 +00:00
1d52d2c24d Revert "Add SSLConfig object to IceServer."
This reverts commit 7f1ffcccce563a086da037334aec2d4faa723edb.

Reason for revert: Speculative revert

Original change's description:
> Add SSLConfig object to IceServer.
> 
> This is being added to allow greater configurability to TLS connections.
> tlsAlpnProtocols, tlsEllipticCurves and tlsCertPolicy will be removed from IceServer in a
> follow-up CL.
> 
> Bug: webrtc:9662
> Change-Id: I33cb804b02c26c662ed2a28c76f9a9dc2df40f36
> Reviewed-on: https://webrtc-review.googlesource.com/96020
> Commit-Queue: Diogo Real <diogor@google.com>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24559}

TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,juberti@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org,diogor@google.com

Change-Id: Iae9fc68b77f743876bda36fc2a04f6d791aae8e2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9662
Reviewed-on: https://webrtc-review.googlesource.com/98000
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24563}
2018-09-05 08:15:29 +00:00
7f1ffcccce Add SSLConfig object to IceServer.
This is being added to allow greater configurability to TLS connections.
tlsAlpnProtocols, tlsEllipticCurves and tlsCertPolicy will be removed from IceServer in a
follow-up CL.

Bug: webrtc:9662
Change-Id: I33cb804b02c26c662ed2a28c76f9a9dc2df40f36
Reviewed-on: https://webrtc-review.googlesource.com/96020
Commit-Queue: Diogo Real <diogor@google.com>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24559}
2018-09-04 22:46:19 +00:00
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