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}
In the Discussion part of
https://developer.apple.com/documentation/metal/mtltexture/1515679-replaceregion
it seems like we should sync the calls to replaceRegion (inside
setupTexturesForFrame) in RTCMTLRenderer and not just the command
buffer.
This is a speculative fix for the linked bug, but we don't have any
clear repro case. Have done basic testing in AppRTCMobile and don't
see any obvious regressions, so might be worth trying.
Bug: webrtc:10024
Change-Id: Id6848691129fba8845f38c3dfe0ba53b9e5a27ce
Reviewed-on: https://webrtc-review.googlesource.com/c/123766
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26819}
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 modernize-use-override [1] to 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/modernize-use-override.html
Bug: webrtc:10252
Change-Id: I2bb8bd90fa8adb90aa33861fe7c788132a819a20
Reviewed-on: https://webrtc-review.googlesource.com/c/120412
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26461}
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}
When using H264 encoder with profile level 3.1, the encoder may crash.
The reason is that we set the expected frame rate using kVTCompressionPropertyKey_ExpectedFrameRate
to the VideoToolBox. However, by iOS implementation, if our setting violates the sample rate limit
[1], the encoder will crash.
This CL fixes the bug by capping the expected frame rate with max allowed frame rate computed from the sample rate limit.
Change-Id: I090d7be8c20713c6a5a4ec80ed243c8fa7b4aa14
Bug: webrtc:10172
Reviewed-on: https://webrtc-review.googlesource.com/c/116056
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26254}
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}
This causes a crash if the NV12 texture cache attempts to upload textures
for a frame with a NULL backing CVPixelBufferRef.
Bug: webrtc:10175
Change-Id: I6866dcde5ace745cbd95b762254294aa8406c2a5
Reviewed-on: https://webrtc-review.googlesource.com/c/115430
Commit-Queue: Chuck Hays <haysc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26140}
When using WebRTC in iOS this Warning is shown for every single call even if stereo is not being used at all.
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Bug: webrtc:10146
Change-Id: I0cc71620b9deb0692544101d78c0801968edbb26
Reviewed-on: https://webrtc-review.googlesource.com/c/85283
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26133}
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}
Previously we were setting the property again in it's setter. This is
obviously not a great idea. CL 109641 changed ivar accesses in blocks
to property accesses and this bug got introduced there.
Bug: webrtc:10110, webrtc:10127, webrtc:9971
Change-Id: I01abb0885b3bfc91fb741d82d1ece015ee9d3b58
Reviewed-on: https://webrtc-review.googlesource.com/c/116062
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26124}
This reverts commit cdc5eb0de1.
Reason for revert: Causes wrong CPU adaptation to be used for some HW codecs since GetEncoderInfo() is polled before InitEncode().
Original change's description:
> Replace VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo
>
> Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
> until it is removed downstream and remove all implementations of it.
>
> Bug: webrtc:10065
> Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
> Reviewed-on: https://webrtc-review.googlesource.com/c/113065
> Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25924}
TBR=brandtr@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org,mirtad@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10065
Change-Id: Idaa452e1d8c1c58cdb4ec69b88fce9042589cc3c
Reviewed-on: https://webrtc-review.googlesource.com/c/113800
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25943}
Make implementation of VideoEncoderFactory::QueryVideoEncoder optional
until it is removed downstream and remove all implementations of it.
Bug: webrtc:10065
Change-Id: Ibb1f9612234e536651ce53f05ee048a5d172a41f
Reviewed-on: https://webrtc-review.googlesource.com/c/113065
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25924}
It's currently used only by the VCMJitterBuffer and VCMReceiver
classes. Injection is needed by the VCMReceiverTimingTest test, which
defines a subclass(!) of EventWrapper.
Bug: webrtc:3380
Change-Id: I765be0ceac58e941928319cc426ba49f1cbdc5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25893}
This is the first step in moving the metadata and eventually replacing
VideoEncoderFactory::QueryVideoEncoder with VideoEncoder::GetEncoderInfo.
Bug: webrtc:10065
Change-Id: If925b895718e1b1225d2cf49bede1adb3ff281b8
Reviewed-on: https://webrtc-review.googlesource.com/c/112285
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25856}
It makes sense to clean up self.videoFrame in -teardownGL, but if
we happen to have a frame available in -setupGL then it's OK to
keep using that frame. Clearing the frame here frequently causes
the screen view to go black for a moment when the app returns from
the background.
Bug: webrtc:10059
Change-Id: Ic62f872a0a582c807cee1e30ea9bb32f31ada341
Reviewed-on: https://webrtc-review.googlesource.com/c/112213
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25816}
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 CL utilizes the input frame rate in the RTCVideoEncoderH264, by setting it into VT Property.
The main purpose is to guide VT encoder to make correct decision of the encoded frame size.
Bug: webrtc:10015
Change-Id: Id5c89f2876539f3181030f49b546326fc40b8ea3
Reviewed-on: https://webrtc-review.googlesource.com/c/111420
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25724}
RTC_ENABLE_VP9 is more natural to deal with then RTC_DISABLE_VP9.
In all the places this macro is used, WebRTC needs to do more things
so it is easier to "do more if RTC_ENABLE_VP9 is defined" than
"do more if RTC_DISABLE_VP9 is not defined".
Bug: None
Change-Id: If992e5c554173e6af3f030f6e0fd21bd82acf9eb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/111242
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25679}
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}
Implicitly retaining self pointer (assuming this is intended behavior) causes compiler warning `-Wimplicit-retain-self`. We should do it explicitly.
Bug: webrtc:9971
Change-Id: If77a67168d8a65ced78d5119b9a7332391d20bc9
Reviewed-on: https://webrtc-review.googlesource.com/c/109641
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25609}
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}
The SetChannelParameters function was used when WebRTC supported decoding
with errors, which we no longer do.
This cleanup CL is related to the work tracked by 9946.
Bug: webrtc:9946
Change-Id: Id2d5ed23031388f890c42651bfbe5f79eda701e5
Reviewed-on: https://webrtc-review.googlesource.com/c/108861
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25505}
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}
Unsafe reference is no longer sufficient with newer versions of XCode. See
https://bugs.chromium.org/p/webrtc/issues/detail?id=9457#c23
Bug: webrtc:9457
Change-Id: I58ca4456c0abd450b8c42fa87ba4129c772d370d
Reviewed-on: https://webrtc-review.googlesource.com/c/104700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25058}
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 enables tests that were previously disabled and fixes the issues
that made them flaky.
Bug: webrtc:6889, webrtc:7888
Change-Id: I914b59200d7bf2973e8993b04de867cc3355b8a8
Reviewed-on: https://webrtc-review.googlesource.com/98381
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24930}
Symbols under sdk/ are now exported using RTC_OBJC_EXPORT, while
RTC_EXPORT is used for C++ symbols.
Bug: webrtc:9419
Change-Id: Icdf7ee0e7b3faf4d7fec33e9b33a3b13260f45b7
Reviewed-on: https://webrtc-review.googlesource.com/102461
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24886}
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 CL removes a set of DCHECKs in AudioDeviceBuffer (ADB) where the goal has been
to ensure that some methods are called on one and the same native I/O thread.
The implementation of the ADB is platform independent but the underlying (driving)
audio components differ between platforms. This combination has shown to generate complex
corner cases such as:
- OS dependent I/O-thread(s) changes while audio is active
- OS dependent audio device changes and it leads to restart of native I/O threads
- Start/Stop of audio has different timing depending on platform and possibly also usage of
JNI and/or emulators.
To summarize: the gain of maintaining the current strict thread checking (in Debug mode)
is not worth all the efforts trying to resolve complex dynamic cases where the native
I/O threads changes ID.
TBR=glaznev
Bug: b/115385789
Change-Id: I681c89adec497a18b97d2a40421c04ea218fd919
Reviewed-on: https://webrtc-review.googlesource.com/100200
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24723}
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}
This component was added to work around an issue in iOS 8, which is
no longer supported by WebRTC. It's removal is made more urgent by
the fact that it prevents WebRTC being used by iOS extensions.
Bug: webrtc:9335
Change-Id: I2a3327534fe6d5014c34a9e908096d825e8149e3
Reviewed-on: https://webrtc-review.googlesource.com/87822
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24688}
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 is a reland of 711a31aead
Changes since original landing:
Rename methods only used by tests, mainly via FakeClock,
MessageQueueManager::ProcessAllMessageQueues
--> ProcessAllMessageQueuesForTesting
MessageQueue::IsProcessingMessages
--> IsProcessingMessagesForTesting
Fix the handling of null rtc::Thread::Current() in
ProcessAllMessageQueuesInternal().
Add override Thread::IsProcessingMessagesForTesting() to return false
for the wrapped main thread, unless it's also the current thread. In
tests, the main thread is typically not processing any messages,
but blocked in an Event::Wait().
Original change's description:
> Explicitly wrap main thread in test_main.cc.
>
> Bug: webrtc:9714
> Change-Id: I6ee234f9a0b88b3656a683f2455c3e4b2acf0d54
> Reviewed-on: https://webrtc-review.googlesource.com/97683
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24560}
Bug: webrtc:9714
Change-Id: I6f022d46aaf1e28f86f09f2d68c1803b69770126
Reviewed-on: https://webrtc-review.googlesource.com/98060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24596}
Similarly to how it is done for iOS.
Bug: webrtc:9627
Change-Id: I7e4e3495d28a0a098531250bfdcf93d272e27b9d
Reviewed-on: https://webrtc-review.googlesource.com/98162
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24594}
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}
Intention is to make the member private, but downstream callers
must be updated to use the accessor methods first.
Bug: webrtc:9378
Change-Id: I3495bd8d545b7234fbea10abfd14f082caa420b6
Reviewed-on: https://webrtc-review.googlesource.com/82160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24352}
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 code never executes as we always get passed a nil codecSpecificInfo.
Bug: webrtc:9580
Change-Id: I5c5311c20877494978df45d409a53ad5b0e86a9b
Reviewed-on: https://webrtc-review.googlesource.com/92083
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24177}
RTCNativeAudioSessionDelegateAdapter has a raw pointer to AudioDeviceIOS,
and receives callbacks from RTCAudioSession and forwards them to AudioDeviceIOS.
During teardown of these components the situation can occur that the dtor for
AudioDeviceIOS has been called but the ObjC runtime has not yet dealloced
RTCNativeAudioSessionDelegateAdapter, so it's still receiving callbacks while
the pointer it keeps to AudioDeviceIOS has been invalidated.
This occasionally triggers a crash when WebRTC is shutting down.
The fix in this CL is to make sure to deregister the adapter from RTCAudioSession
_before_ the dtor for AudioDeviceIOS returns.
Bug: webrtc:9523
Change-Id: Ica85420d76efc63940472bc43e3ec71d16036ccf
Reviewed-on: https://webrtc-review.googlesource.com/90245
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24173}
After this change, it is only recreated when needed.
This change also clarifies the relation between the compression
session and the pixel buffer pool, and handles invalid sessions
explicitly.
Change-Id: Iae4aa02b60b0d5c153db3ae2d4cd2a0cfa05757b
Bug: webrtc:9562
Reviewed-on: https://webrtc-review.googlesource.com/90403
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24161}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I6f03c46e772ccf4d15951a4b9d4e12015d539e58
Reviewed-on: https://webrtc-review.googlesource.com/90408
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24113}
BGRA RTCCVPixelBuffers were cropped and scaled incorrectly. Libyuv’s
`ARGBScale` method is used in RTCCVPixelBuffer to scale and crop the
pixel buffer. To crop by `cropX` and `cropY` pixels, pointer
arithmetic is used to offset the src pointer of the original pixel
buffer bytes. There is a bug in how this offset is calculated.
The offset is done by `src += srcStride * _cropY + _cropX`. Libyuv
expects that the src pointer will point to the start of a new pixel.
However, if _cropX is a not a multiple of 4 (4 bytes for BGRA), the src
pointer will point to a byte in the middle of a pixel and thus libyuv
will incorrectly treat the data as the start of pixel (incorrectly
treating the first byte as red when it is actually green, etc...). To
fix this, the src pointer needs to be offset to always point to the
start of a new pixel.
Before this change:
Original Test Gradient image with a cropX of 2:
https://i.imgur.com/gSIgwGV.jpg
Scaled image (notice the colors are incorrect):
https://i.imgur.com/oPxbTEK.jpg
After this change:
Scaled image (notice the colors are correct):
https://i.imgur.com/dqBsmsH.jpg
A new unit test which tests scaling with cropX and cropY values has been
added. The test fails without this change and now passes with the
correct src pointer offsetting.
Bug: webrtc:9555
Change-Id: I87cbd7b91bc139d51fb4e11cc50ccb014cfa8051
Reviewed-on: https://webrtc-review.googlesource.com/89220
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24076}
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}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script passing top level directories except rtc_base and api
find $@ -type f \( -name \*.h -o -name \*.cc -o -name \*.mm \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I9465c172e65ba6e6ed4e4fdc35b0b265038d6f71
Reviewed-on: https://webrtc-review.googlesource.com/84584
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23697}
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 CL adds support targeting iOS 10 as a min version.
Bug: None
Change-Id: I353a9884eb907e97387553fd73427fd7cb0dbfc2
Reviewed-on: https://webrtc-review.googlesource.com/79921
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23461}
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}
Made some minor changes to resolve the issue. Only affects Debug builds.
NOTRY=TRUE
Bug: webrtc:9310
Change-Id: Ieeeb57d24b559282b2eefd4d8785f7cfe4f44e40
Reviewed-on: https://webrtc-review.googlesource.com/79624
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23434}
This removes usage of the old OnFailure methods on CreateSessionDescriptionObserver
and SetSessionDescriptionObserver, so that WebRTC will continue to compile
once all the default implementations are removed.
Bug: chromium:589455
Change-Id: Id67295b3ad0c30d24d79589c2041acdd507a19f3
Reviewed-on: https://webrtc-review.googlesource.com/78480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23427}
