This was an ICE configuration experiment added a couple years ago that did not end up being used.
Bug: webrtc:11316
Change-Id: Iafb7e1c4f7b4598815f045808dbf6e470172f119
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167680
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30395}
This removes the SetParameters method from AudioRtpReceiver and Video
RtpReceiver, which is currently not used and is not part of the
specifications.
Bug: webrtc:11111
Change-Id: I6f67773bfef2d4b51e9ab670bde17b5fbf5f94c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159307
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Saurav Das <dinosaurav@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29995}
And move related files into api/transport/ and api/transport/media/.
The moved files are unchanged, except that
congestion_control_interface.h and datagram_transport_interface.h
no longer include media_transport_interface.h, instead, they forward
declare the few MediaTransport* types they reference.
Bug: webrtc:8733
Change-Id: I4f4000d0d111f10d15a54c99af27ec26c46ae652
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152482
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29178}
Bug: webrtc:10419
Change-Id: I18528bf2526e933568bf052de76a434f012161da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148320
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28838}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
from a field trial to RTCConfiguration.
The test coverage is also expanded for the underlying feature.
Bug: None
Change-Id: Ic9c1362867e4a956c5453be7a9355083b6a442f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28143}
This CL adds the callback on changes of the ICE connection state
following the standardized transitions
(https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate) to the
Android and the iOS SDKs.
Bug: None
Change-Id: I6133391fa54dd4e09016f29dddb85e4a0e270878
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138181
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28127}
This was added in CL 135122, but the bitratePriority parameter is not
standard and not implemented in a way users would expect. So it should
actually not be exposed in the Obj-C SDK.
Bug: webrtc:10438
Change-Id: I801ce940a32701d2703e951ef2b601c606aa2111
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135287
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27861}
These parameters are nullable in the JS API.
This allows cleaner handling of "unset" vs "set" in Chrome.
Backwards compatibility note: Behavior should not change, even for users
who set the values explicitly to -1 in the DataChannelInit struct.
Those who try to read back the value will get a compile-time error.
Bug: chromium:854385
Change-Id: Ib488ca5f70bc24ba8b4a3f71b506434c4d2c60b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131381
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27507}
This adds bindings for RIDs in RtpEncodingParameters.
Bug: webrtc:10464
Change-Id: I3cc25db25a4d777b9d9573ba69c82127d1c9a597
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128826
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27322}
This cl deprecates the FrameType enum, and adds aliases AudioFrameType
and VideoFrameType.
After downstream usage is updated, the enums will be separated
and be moved out of common_types.h.
Bug: webrtc:6883
Change-Id: I2aaf660169da45f22574b4cbb16aea8522cc07a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123184
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27011}
Previously, the use of pass-by-value caused an issue in
ObjCVideoDecoder::Decode, where the EncodedImage was being copied upon
calling initWithNativeEncodedImage, which then created an NSData using
the copy's pointer; then the copy was destroyed, invalidating that
pointer.
Bug: webrtc:9378
Change-Id: Iac28b890c9902108ffc5ec54a607a99034159153
Reviewed-on: https://webrtc-review.googlesource.com/c/121922
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26598}
This CL applies clang-tidy's performance-for-range-copy [1] on the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/performance-for-range-copy.html
Bug: webrtc:10215
Change-Id: I7c83290b8866d76129bbec4e24e6701f5014102e
Reviewed-on: https://webrtc-review.googlesource.com/c/120043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26420}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
Use size() accessor function. Also replace most nearby uses of _buffer
with data().
Bug: webrtc:9378
Change-Id: I1ac3459612f7c6151bd057d05448da1c4e1c6e3d
Reviewed-on: https://webrtc-review.googlesource.com/c/116783
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26273}
Removes the deprecated video codec factories and the related flag and
helper classes.
Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
- Remove visibility of encoder target.
- Remove unnecessary dependency on task_queue.
- Remove CreateRtcEventLogFactory() declaration from the rtc_event_log_api target
since the function is not defined in that target.
Bug: None
Change-Id: Id9edee86f358d08ea063d62bd96e9653c5b06d55
Reviewed-on: https://webrtc-review.googlesource.com/c/116060
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26215}
When landed, the FileRotatingStream class can be made write-only.
Bug: webrtc:7811
Change-Id: I6dcd2a869301b9b8273b48d47df51a1065767ffd
Reviewed-on: https://webrtc-review.googlesource.com/c/115302
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26126}
This change removes the ability to set CryptoOptions through the PeerConnection
Factory in both Java and IOS. Native will be removed after the Chromium change
lands. The semantics have been changed such that these options should only be
set on individual PeerConnections and not directly on the Factory itself. This
allows for more flexibility in setting CryptoOptions for PeerConnections which
are created as part of a factory.
Bug: webrtc:10020
Change-Id: I9ef3d431e728927b9ced5de6188cedeb2671254b
Reviewed-on: https://webrtc-review.googlesource.com/c/111560
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25736}
This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
Originally reviewed as https://webrtc-review.googlesource.com/c/110502, with an added check to prevent calling unimplemented optional method.
Bug: webrtc:9977
Change-Id: Iebac8ce58d435e38450add51b8915575d0ffd934
Reviewed-on: https://webrtc-review.googlesource.com/c/111084
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25727}
This reverts commit 586725dc9a.
Reason for revert: misses a check to see if the optional callback is implemented.
Original change's description:
> Add ios bindings for PeerConnectionState.
>
> This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
>
> Bug: webrtc:9977
> Change-Id: Icf69bb1faa0383ae239cb7508f2a740a2d489697
> Reviewed-on: https://webrtc-review.googlesource.com/c/110502
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25651}
TBR=kthelgason@webrtc.org,jonasolsson@webrtc.org
Change-Id: Iff919e9876e6b8dddc6d8ab7df302081d0cfa917
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9977
Reviewed-on: https://webrtc-review.googlesource.com/c/111062
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25659}
This change makes it possible for ios apps to use the new standards-compliant PeerConnectionState.
Bug: webrtc:9977
Change-Id: Icf69bb1faa0383ae239cb7508f2a740a2d489697
Reviewed-on: https://webrtc-review.googlesource.com/c/110502
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25651}
Adds a field |use_media_transport_for_data_channels| to RTCConfiguration.
PeerConnection requires a media transport factory to be set if this bit
is set. As with |use_media_transport|, the value may not be modified
after setting the local or remote description.
If either |use_media_transport| or |use_media_transport_for_data_channel| is
set, PeerConnection uses its media transport factory when creating a JSEP
transport controller.
PeerConnection stops unconditionally using media transport in
CreateVoiceChannel, as it may be present only for use in data channels. It uses
the media transport if it is present and |use_media_transport| is set.
Bug: webrtc:9719
Change-Id: I59d4ce8f7531fd19d9c17eefe033f063f663ebcc
Reviewed-on: https://webrtc-review.googlesource.com/c/109041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25507}
With the expanding use cases for webrtc::CryptoOptions it makes more sense for
it to be be available per peer connection instead of only as a factory option.
To support backwards compatability for now this code will support the factory
method of setting crypto options by default. However it will completely
overwrite these settings if an RTCConfiguration.crypto_options is provided.
Got LGTM offline from Sami, adding him to TBR if he has any further comments.
TBR=sakal@webrtc.org
Bug: webrtc:9891
Change-Id: I86914cab69284ad82afd7285fd84ec5f4f2c4986
Reviewed-on: https://webrtc-review.googlesource.com/c/107029
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25375}
This change adds a new subcategory to the public native webrtc::CryptoOptions
structure: webrtc::CryptoOptions::Frame.
This new structure has a single off by default property:
crypto_options.frame.require_frame_encryption.
This new flag if set prevents RtpSenders from sending outgoing payloads unless
a frame_encryptor_ is attached and prevents RtpReceivers from receiving
incoming payloads unless a frame_decryptor_ is attached.
This option is important to enforce no unencrypted data can ever leave the
device or be received.
I have also attached bindings for Java and Objective-C.
I have implemented this functionality for E2EE audio but not E2EE video
since the changes are still in review.
Bug: webrtc:9681
Change-Id: Ie184711190e0cdf5ac781f69e9489ceec904736f
Reviewed-on: https://webrtc-review.googlesource.com/c/105540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25238}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@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@{#25135}
This reverts commit ac2f3d14e4.
Reason for revert: Breaks downstream project
Original change's description:
> Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h
>
> Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
> that only handles SRTP configuration to a more generic structure that can be
> used and extended for all per peer connection CryptoOptions that can be on a
> given PeerConnection.
>
> Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
> accessed as crypto_options.srtp.whatever_option_name. This is more inline with
> other structures we have in WebRTC such as VideoConfig. As additional features
> are added over time this will allow the structure to remain compartmentalized
> and concerned components can only request a subset of the overall configuration
> structure e.g:
>
> void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
>
> In addition to this it made little sense for sslstreamadapter.h to hold all
> Srtp related configuration options. The header has become loo large and takes on
> too many responsibilities and spilting this up will lead to more maintainable
> code going forward.
>
> This will be used in a future CL to enable configuration options for the newly
> supported Frame Crypto.
>
> Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
> Bug: webrtc:9681
> Reviewed-on: https://webrtc-review.googlesource.com/c/105180
> Reviewed-by: Emad Omara <emadomara@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25130}
TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
Bug: webrtc:9681
Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff
Reviewed-on: https://webrtc-review.googlesource.com/c/105541
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25133}
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Change-Id: I99d1be36740c59548c8e62db52d68d738649707f
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/c/105180
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25130}
And use RTCConfiguration to enable/disable it on a per connection basis.
With the advent of MediaTransportInterface, we need to be able to enable
it on the per PeerConnection basis.
At this point PeerConnection will not take any action when the
MediaTransportInterface is set; this code will land a bit later, and
will be accompanied by the tests that verify correct setup (hence no tests right now).
At this point this is just a method stub to enable further development.
Bug: webrtc:9719
Change-Id: I1f77d650cb03bf1191aa0b35669cd32f1b68446f
Reviewed-on: https://webrtc-review.googlesource.com/c/103860
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25053}
I missed adding these headers in my inital check-in. This change simply adds
these headers.
Bug: webrtc:9681
Change-Id: Ic2265105cd401d59fac124c2dc1963f0163c5af6
Reviewed-on: https://webrtc-review.googlesource.com/c/103304
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24956}
This change adds bindings so that native FrameEncryptor and native FrameDecryptor
objects can be set on the objective C RTCRtpSender and RTCRtpReceiver objects.
Bug: webrtc:9681
Change-Id: Iec4006ea020d6ab6adcc0ad068dcd8fb2738063d
Reviewed-on: https://webrtc-review.googlesource.com/c/103020
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24936}
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}
A new version of RTC_EXPORT will be introduced by [1] and it will be
used by WebRTC native code.
This CL renames the current RTC_EXPORT to RTC_OBJC_EXPORT in order
to avoid to mix them. It has been decided to avoid to unify them because
RTC_OBJC_EXPORT always marks symbols with default visibility, while
RTC_EXPORT will do it only when COMPONENT_BUILD is defined.
[1] - https://webrtc-review.googlesource.com/c/src/+/97960 is
Bug: webrtc:9419
Change-Id: I56a3fc6601c72d3ad6a58f9961a00e3761dfb5da
Reviewed-on: https://webrtc-review.googlesource.com/100521
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24754}
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}
With "aggressive" preset the default bundlePolicy is set to "maxBundle" when it shoud be "balanced" according to spec.
Bug: webrtc:9458
Change-Id: Ifbdd76be3a6d9968574cba857f178d5f859dcb87
Reviewed-on: https://webrtc-review.googlesource.com/88567
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24650}
This method is allowed to return nil but was not annotated as such.
Bug: webrtc:8560
Change-Id: If54aa94d6ff83b7bdb87b526244616e2627a8999
Reviewed-on: https://webrtc-review.googlesource.com/97380
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24568}
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}
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}
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}
