This is quite useful in many places where we need to restrict the range
of a DataRate. It makes it easier to read the intention than with:
value_ = std::max(some_lower_limit, std::min(value_, some_upper_limit));
The naming follows the naming for rtc::SafeClamp.
Bug: webrtc:9709
Change-Id: I08e05197acec325d85babd2a06806a8667f2fcb1
Reviewed-on: https://webrtc-review.googlesource.com/c/104040
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25023}
Automatic detection if one-byte header or two-byte header should be used based
on extension ID and extension length.
Bug: webrtc:7990
Change-Id: I9fc848ecc59458d1ca97bace0e57ea04d3d0ced6
Reviewed-on: https://webrtc-review.googlesource.com/c/103422
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25004}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.
If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.
Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}
This name will better describe what implementation should do and how
users will interact with this class. The real simulation is done
by FakeNetworkPipe and this class is just operates with packages
metadata, so it is more about describing behavior.
Bug: webrtc:9630
Change-Id: I00977e6be0ca84e7c233b4c35f0677e8263e4382
Reviewed-on: https://webrtc-review.googlesource.com/c/95944
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24989}
Some important NetEq information was not available in NetEqState, which
meant it was not available on the API. This CL adds additional
information.
Bug: webrtc:9667
Change-Id: I702707c7d60472f488047d48fb286f839c5608dc
Reviewed-on: https://webrtc-review.googlesource.com/c/102300
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24985}
This reverts commit 9e24dcff16.
Reason for revert: Breaks chromium.webrtc.fyi bots.
Original change's description:
> Export symbols needed by the Chromium component build (part 1).
>
> 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: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
> Reviewed-on: https://webrtc-review.googlesource.com/c/103505
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24969}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org
Change-Id: I01f6e18f0d2c0f0309cdaa6c943c3927e1f1f49f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/103720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24974}
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: I802abd32874d42d3aa5ecd3c8022e7cf5e043d99
Reviewed-on: https://webrtc-review.googlesource.com/c/103505
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24969}
Configuring different number of temporal layers per simulcast layer is not supported.
Bug: webrtc:9785
Change-Id: I5709b2235233420e22e68fb0ae512305ae87e36c
Reviewed-on: https://webrtc-review.googlesource.com/c/102120
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24942}
This CL is a step towards making the TemporalLayers landable in api/ :
* It splits TemporalLayers from TemporalLayersChecker
* It initially renames temporal_layer.h to vp8_temporal_layers.h and
moved it into the include/ folder
* It removes the dependency on VideoCodec, which was essentially only
used to determine if screenshare_layers or default_temporal_layers
should be used, and the number of temporal temporal layers to use.
Subsequent CLs will make further cleanup before attempting a move to api
Bug: webrtc:9012
Change-Id: I87ea7aac66d39284eaebd86aa9d015aba2eaaaea
Reviewed-on: https://webrtc-review.googlesource.com/94156
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24920}
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}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
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}
This reverts commit 3f4a4fad8c.
Reason for revert: Breaking internal tests
Original change's description:
> Added field trial WebRTC-GenericDescriptor for the new generic descriptor.
>
> Also parameterized tests to test the new generic descriptor and
> added --generic_descriptor flag to loopback tests.
>
> Bug: webrtc:9361
> Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
> Reviewed-on: https://webrtc-review.googlesource.com/101900
> Commit-Queue: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24835}
TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I4d4714a9f4ab0e95adf0f4130bc1a932efc448fa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9361
Reviewed-on: https://webrtc-review.googlesource.com/101940
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24839}
Also parameterized tests to test the new generic descriptor and
added --generic_descriptor flag to loopback tests.
Bug: webrtc:9361
Change-Id: I2b76889606fe2e81249ecdebb0b7b61151682485
Reviewed-on: https://webrtc-review.googlesource.com/101900
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24835}
This is experimental interface for media transport.
The goal is to refactor WebRTC codebase to send/receive frames via media transport interface. It will allow us to have different media transport implementations in the future, including QUIC-based media transport.
This change focuses on core video interfaces: sending frames, receiving
frames, requesting keyframes.
It also defines a 'state sink' which allows us to know when the
connection is fully established (so that we can send key frame right
away).
Bug: webrtc:9719
Change-Id: I0480337c699b337cabd13c27de8987ad06241b3a
Reviewed-on: https://webrtc-review.googlesource.com/99304
Commit-Queue: Peter Slatala <psla@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24832}
Update uri to include version.
Bug: webrtc:9361
Change-Id: I71164879d2a5b88c052fdab9f74202f758e85be6
Reviewed-on: https://webrtc-review.googlesource.com/101762
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24827}
The proper signature of the function should return an RTCError,
but due to all the classes in external projects implementing it with
this signature, we need a many steps process to update this.
- Add an implementation for the pure virtual method
- Update projects not to override it
- Update the function signature
- Update projects to override it with the right signature
- Remove the dummy implementation from the interface
Bug: webrtc:9777
Change-Id: Idf99216792b4ad13339e4e8be6f7b735bb6b64e7
Reviewed-on: https://webrtc-review.googlesource.com/101564
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24807}
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}
This reverts commit 958ed23860.
Reason for revert: Will need additional work to handle clock-drift.
Original change's description:
> AEC3: Reduce filter divergence during low-echo double-talk
>
> Bug: webrtc:9746,chromium:883264
> Change-Id: Ie3faf106fd1fd835e67d9e6794c679703af54fea
> Reviewed-on: https://webrtc-review.googlesource.com/99920
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24706}
TBR=gustaf@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9746, chromium:883264
Change-Id: Ib039eb80e2ddfc43ec52183086da2474baef65e0
Reviewed-on: https://webrtc-review.googlesource.com/100480
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24741}
This change enables NetEq to use the packet concealment audio (aka
PLC) produced by a decoder. The change also includes a new API to the
AudioDecoder interface, which lets the decoder implementation generate
and deliver concealment audio.
Bug: webrtc:9180
Change-Id: Icaacebccf645d4694b0d2d6310f6f2c7132881c4
Reviewed-on: https://webrtc-review.googlesource.com/96340
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24738}
This change allows supporting additional data for authentication and adds a
requirement for the contributing source to be provided during decryption.
Change-Id: Ifc19cb2d8a7d6c3715c83c95cf12f64df0bca454
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/100001
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24712}
This reverts commit 4f085434b9.
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}
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}
It is problematic to set field trials more than once, so to avoid running into problems, this functionality has been placed in the main function of neteq_rtpplay.
Bug: webrtc:9667
Change-Id: Ib9b9990f30a1715b50889dbfc4d74787bcbe5dae
Reviewed-on: https://webrtc-review.googlesource.com/98541
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24673}
This move constructor causes downstream issues, so it needs to be removed for now.
Bug: webrtc:9667
Change-Id: Ic15bfdf6b392a95e05bf75bc2c1dd32ce132d32b
Reviewed-on: https://webrtc-review.googlesource.com/99121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24672}
This is necessary to migrate some clients so that we can move forward
with removal of cricket::WebRtcVideoDecoderFactory.
TBR=stefan@webrtc.org
Bug: webrtc:7925
Change-Id: Icc2949e3f7f3137d1b68eb30874f14a33168e41f
Reviewed-on: https://webrtc-review.googlesource.com/97500
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24671}
This CL adds more useful information to NetEqState, and implements setting action_times_ms, which can be used to get a better idea of what actually happened during a timestep.
Bug: webrtc:9667
Change-Id: I789a3e1ad852066fdf4e9b4c96b8fb6033dacb27
Reviewed-on: https://webrtc-review.googlesource.com/98163
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24643}
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}
This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.
To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.
Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
This change allows an implementer to provide a custom error code to be returned
on failure with 0 being reserved for success. It also modifies the output size
APIs to be more specific and requests the number of bytes written to be returned
so we can determine how many bytes we need to send.
Bug: webrtc:9681
Change-Id: I13d34861bf851527fcbb350d0cfb480c0f95a6b3
Reviewed-on: https://webrtc-review.googlesource.com/98720
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24613}
This is experimental interface for media transport.
The goal is to refactor WebRTC codebase to send/receive frames via media transport interface. It will allow us to have different media transport implementations in the future, including QUIC-based media transport.
Bug: webrtc:9719
Change-Id: I64e0b69d18c212e1ed0a08c6904578c3dfbe3af7
Reviewed-on: https://webrtc-review.googlesource.com/95960
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24612}
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}
This reverts commit 7f1ffcccce.
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}
