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}
These headers was lost in the cleanup CL for the Obj-C directories. This
puts them back in the framework headers.
Note that since the protocol and interface was split into two different
headers, and all public framework headers are put into a flat directory
structure, I had to rename the implementation files so they would not collide
in the framework header directory.
Bug: webrtc:9701
Change-Id: I42d4c1e02bdfa4e114575f527c4c42a19be8fb52
Reviewed-on: https://webrtc-review.googlesource.com/97330
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24539}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
The JS API supports two operations which have never been implemented in
the iOS counterpart:
- generate a new certificate
- use this certificate when creating a new PeerConnection
Both functions are illustrated in the generateCertificate example code:
- https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/generateCertificate
Currently, on iOS, a new certificate is automatically generated for
every PeerConnection with no programmatic way to set a specific
certificate.
Work sponsored by |pipe|
Bug: webrtc:9498
Change-Id: Ic1936c3de8b8bd18aef67c784727b72f90e7157c
Reviewed-on: https://webrtc-review.googlesource.com/87303
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24276}
This changeset adds the ability for API users to enable or disable GCM Cipher
suites from objective-c.
Bug: chromium:713701
Change-Id: I0ac7b60f55dd56bebbcfb315a542ef4843099802
Reviewed-on: https://webrtc-review.googlesource.com/89263
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24028}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Add a new flag to RtcConfiguration. By setting that flag to true, the
SRTP parameters will be reset whenever the DTLS transports are reset
after every offer/answer negotiation.
The flag is added to Android and Objc wrapper as well.
This should only be used as a workaround for the linked bug, if the
application knows that the other party is affected (for instance,
using a version number).
TBR=sakal@webrtc.org, denicija@webrtc.org
Bug: chromium:835958
Change-Id: I6db025e1c69bf83e1b1908f7df4627430db9920c
Reviewed-on: https://webrtc-review.googlesource.com/83101
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23587}
This is an ObjC followup to https://webrtc-review.googlesource.com/c/src/+/78741.
This CL only adds the field to the API, but does not wire it up.
Bug: webrtc:9341
Change-Id: Id6b1ac681324120bc90158029da7a80bf99aa512
Reviewed-on: https://webrtc-review.googlesource.com/81182
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23524}
This provides an environment for testing out using WebRTC from an iOS
extension. It implements a ReplayKit broadcast extension for live
streaming games and screensharing.
The extension is only supported on iOS 11+ and is guarded by a build
flag.
Bug: webrtc:9335
Change-Id: Id218d6c73ef7599f5953c5a1e0e62e5d0dc4f10b
Reviewed-on: https://webrtc-review.googlesource.com/80000
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23504}
Expose this functionality in the Obj-C SDK to make it nicer to use for
Obj-C clients.
Bug: None
Change-Id: I5cb511af8799ac0fda15153d16f2550b848b93b2
Reviewed-on: https://webrtc-review.googlesource.com/80481
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23488}
This reverts commit fc4a9c9333.
Reason for revert: Remote video is not showing in a video call.
Original change's description:
> Metal rendering should account for cropping.
>
> Also:
> - added a rotation override to allow ignoring frame rotation
> - fixed a couple of minor issues
> - made it possible to run the MTKView without the DisplayLink
>
> Bug: webrtc:9301
> Change-Id: Ia83c152d9b6d45d56ceb80d287b5d3eacfaebddd
> Reviewed-on: https://webrtc-review.googlesource.com/78282
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23452}
TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org
Change-Id: Iddf7793368531d2d7268c1ec138bb3a9874a4ab7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9301
Reviewed-on: https://webrtc-review.googlesource.com/80020
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23455}
Also:
- added a rotation override to allow ignoring frame rotation
- fixed a couple of minor issues
- made it possible to run the MTKView without the DisplayLink
Bug: webrtc:9301
Change-Id: Ia83c152d9b6d45d56ceb80d287b5d3eacfaebddd
Reviewed-on: https://webrtc-review.googlesource.com/78282
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23452}
This will return all the fmtp parameters for the codecs, except for
DTMF codes that don't fit the key=value pattern.
Bug: webrtc:7112
Change-Id: I06a203ff64df2c3bc9bc2082cd0f374718b23510
Reviewed-on: https://webrtc-review.googlesource.com/71801
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23250}
This is a reland of 5faf36ef3c
The issue in Chrome has been fixed and this should be safe to reland.
TBR=deadbeef
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
Bug: webrtc:7580
Change-Id: Iabd41fb21afdf452c039d5513824ae334f8d1d3f
Reviewed-on: https://webrtc-review.googlesource.com/76980
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23247}
This reverts commit 5faf36ef3c.
Reason for revert: fast/peerconnection/RTCRtpSender-setParameters.html
failing in webrtc roll, probably this CL? https://chromium-review.googlesource.com/c/chromium/src/+/1045889.
Original change's description:
> Implement RtpParameters.transaction_id for PC RtpSenderInterface
>
> The transaction_id field should be refreshed for every getParameters()
> call and checked at each setParameters() call.
> This also checks that getParameters() was ever called to return a proper
> error code.
>
> Bug: webrtc:7580
> Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
> Reviewed-on: https://webrtc-review.googlesource.com/70820
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23120}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,orphis@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7580
Change-Id: I86da108227f8fc8d235bb2e9559377c800595b8c
Reviewed-on: https://webrtc-review.googlesource.com/74740
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23134}
The transaction_id field should be refreshed for every getParameters()
call and checked at each setParameters() call.
This also checks that getParameters() was ever called to return a proper
error code.
Bug: webrtc:7580
Change-Id: I6c6fe289542e486fc422cdc61577982b0529d4c1
Reviewed-on: https://webrtc-review.googlesource.com/70820
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23120}
This CL removes the use of default built-in SW in the ObjC layer. If a
client want to depend on the video SW codecs, they must inject them
explicitly.
Bug: webrtc:7925
Change-Id: If752e7f02109ff768dc5ec38d935203de85987c2
Reviewed-on: https://webrtc-review.googlesource.com/69800
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23073}
We want to allow the application to set it's own content mode.
Bug: b/73147161
Change-Id: I60fab454353a4c39731e49b7b6066e51d8e9a94d
Reviewed-on: https://webrtc-review.googlesource.com/70501
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22962}
There currently are no Objective-C API's to create a buffer with that data.
This change allows us to create a buffer with yuv data.
Bug: webrtc:9167
Change-Id: I00f1b91b04bbaa013a88137d0f54bef44287c5aa
Reviewed-on: https://webrtc-review.googlesource.com/70563
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Commit-Queue: Peter Slatala <psla@google.com>
Cr-Commit-Position: refs/heads/master@{#22945}
The MTL renderer should also have a way to notify it's delegate
that it's content size changed.
The plan is to introduce this new protocol, move existing clients over
to implementing it in favour of RTCEAGLVideoViewDelegate, and then finally
removing the old protocol.
Bug: b/73147161
Change-Id: I908d7b2667e44e02a58066d701a48efec0e98d14
Reviewed-on: https://webrtc-review.googlesource.com/70243
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22944}
This adds confusion to the native API and is only needed for
Chromium UMA metrics, so the appropriate metrics have been moved
upstream and kDefault option removed.
Bug: chromium:811683
Change-Id: I666d7f7793765b8d6edcd99416c8b6c957766f00
Reviewed-on: https://webrtc-review.googlesource.com/59261
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22864}
This parameter is being removed from the C++ API, remove it from the
ObjC API also. It was never used for anything by the H264 decoder.
Bug: webrtc:9107
Change-Id: I5222eac932a4e7d4129d803f8126b5e8d0b027b6
Reviewed-on: https://webrtc-review.googlesource.com/66740
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22730}
This is a reland of 4ea50c2b42
Original change's description:
> Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource.
>
> This CL also fixes a couple of bugs found in the toI420 method for
> RTCCVPixelBuffers backed by RGB CVPixelBuffers.
>
> Bug: webrtc:9007
> Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
> Reviewed-on: https://webrtc-review.googlesource.com/64940
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22656}
Bug: webrtc:9007
Change-Id: I2a787c64f8d23ffc4ef2419fc258d965f8a9480b
Reviewed-on: https://webrtc-review.googlesource.com/66341
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22706}
This reverts commit 4ea50c2b42.
Reason for revert: This change is causing crashes in video calls.
RTCCVPixelBuffer.mm - line 120
Compare is asserting as 420f is not 420v
Original change's description:
> Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource.
>
> This CL also fixes a couple of bugs found in the toI420 method for
> RTCCVPixelBuffers backed by RGB CVPixelBuffers.
>
> Bug: webrtc:9007
> Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
> Reviewed-on: https://webrtc-review.googlesource.com/64940
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22656}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9007
Change-Id: I500514ce05dd0555f8c4a05010ad52bd67c2fed3
Reviewed-on: https://webrtc-review.googlesource.com/65561
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22686}
This CL also fixes a couple of bugs found in the toI420 method for
RTCCVPixelBuffers backed by RGB CVPixelBuffers.
Bug: webrtc:9007
Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
Reviewed-on: https://webrtc-review.googlesource.com/64940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22656}
This flag (added to CryptoOptions) will allow applications to opt-in to
use of this suite, before it's disabled by default later. See bug for
more details.
TBR=magjed@webrtc.org
Bug: webrtc:7670
Change-Id: I800bedd4b26d807b6b7ac66b505d419c3323e454
Reviewed-on: https://webrtc-review.googlesource.com/64390
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22586}
Demonstrates how to use the iOS native API to wrap components into
C++ classes.
This CL also introduces a native API wrapper for the capturer.
The C++ code is forked from the corresponding CL for Android at
https://webrtc-review.googlesource.com/c/src/+/60540
Bug: webrtc:8832
Change-Id: I12d9f30e701c0222628e329218f6d5bfca26e6e0
Reviewed-on: https://webrtc-review.googlesource.com/61422
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22484}
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.
Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This removes the routing for the deprecated audio control setting
Change-Id: Id83ff548625279d5b34c9e3cadc097c25a00ef05
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/58900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22279}
This CL adds wrappers for the following PeerConnection native
APIs to the Objective C API:
- SdpSemantics enum added to the RTCConfiguration
- RTCRtpTransceiver
- RTCPeerConnection.addTrack
- RTCPeerConnection.removeTrack
- RTCPeerConnection.addTransceiver
- RTCPeerConnection.transceivers
Bug: webrtc:8870
Change-Id: I9449df9742a59e90894712dc7749ca30b569d94b
Reviewed-on: https://webrtc-review.googlesource.com/54780
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22214}
This reverts commit 6780c51b23.
Reason for revert:
More details in crbug.com/810292
Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
>
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
>
> R=deadbeef@webrtc.org
>
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}
TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org
Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
from native apps if really necessary.
R=deadbeef@webrtc.org
Bug: webrtc:7670
Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
Reviewed-on: https://webrtc-review.googlesource.com/41420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21952}
