Commit graph

94 commits

Author SHA1 Message Date
Danil Chapovalov
196100efa6 Replace rtc::Optional with absl::optional
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}
2018-06-21 09:32:56 +00:00
Jiawei Ou
ae810c10b4 Create a peer connection factory builder
Similar to the builder on android: https://cs.chromium.org/chromium/src/third_party/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java?rcl=b90e63c620877712e45ee320cfa25cb825bf5373&l=134

1. A builder will allow us to choose what module factories to provide and use default for the others.
2. A helper category is added to provide helpers functions for creating common builders.

Bug: None
Change-Id: I5889bdd7dc2a2aeded62ef5f2c2381edd07089b3
Reviewed-on: https://webrtc-review.googlesource.com/83280
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23696}
2018-06-21 09:22:50 +00:00
Yura Yaroshevich
5297bd21b1 Fixed crash when PCF is destroyed before PC in ObjC
Bug: webrtc:9231
Change-Id: Iaf18257b8f38fa786d462bca5f860f9a7b1cc2d0
Reviewed-on: https://webrtc-review.googlesource.com/78800
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23674}
2018-06-20 06:45:17 +00:00
Florent Castelli
abe301fe6c Add HeaderExtensions to RtpParameters
Bug: webrtc:7580
Change-Id: I4fcf3e8bc4975a6b2baa6f24a17c254d2bf521d9
Reviewed-on: https://webrtc-review.googlesource.com/78288
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23584}
2018-06-12 17:01:40 +00:00
Anders Carlsson
358f2e0760 Broadcast extension for AppRTCMobile on iOS
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}
2018-06-04 08:49:21 +00:00
Anders Carlsson
79ce820a13 Obj-C SDK for parsing and generating H264 ProfileLevelIds.
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}
2018-06-01 11:23:31 +00:00
Anders Carlsson
533c7d0204 Add missing header file to WebRTC iOS SDK.
This header is missing from the public headers when building the
framework.

Bug: None
Change-Id: I7ce3a57d5fa34d344239dfddcc6e29aee35a2ded
Reviewed-on: https://webrtc-review.googlesource.com/79942
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23453}
2018-05-30 15:35:31 +00:00
Florent Castelli
dacec71b16 Add Rtcp parameters for PeerConnection senders
Bug: webrtc:7580
Change-Id: Ibcf5e849a1f11f21fa75f6d006fecf1cd54f8552
Reviewed-on: https://webrtc-review.googlesource.com/78063
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23407}
2018-05-28 09:28:59 +00:00
Peter Hanspers
8d95e3b211 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

(was: https://webrtc-review.googlesource.com/c/src/+/67300)

Bug: webrtc:9120
Change-Id: I46c09900246f75ca5285aeb38f7b8b295784ffac
Reviewed-on: https://webrtc-review.googlesource.com/76741
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23238}
2018-05-15 10:33:01 +00:00
Peter Hanspers
43619a4f4a Revert "Moving iOS Audio Device to sdk."
This reverts commit 08da28dd60.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Moving iOS Audio Device to sdk.
> 
> This change forks the existing iOS audio device module and audio device
> from modules/audio_device/ into sdk/objc/Framework. It also updates
> RTCPeerConnectionFactory to use the forked implementation.
> 
> The unit tests are re-implemented as XCTests.
> 
> (was: https://webrtc-review.googlesource.com/c/src/+/67300)
> 
> Bug: webrtc:9120
> Change-Id: I07340505137b16c2dd487569ad0112f984557bba
> Reviewed-on: https://webrtc-review.googlesource.com/75125
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23208}

TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org

Change-Id: Ibbf8d53eaef386bc3033dc71e9490d5e48911fc9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9120
Reviewed-on: https://webrtc-review.googlesource.com/76460
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23211}
2018-05-14 10:41:20 +00:00
Peter Hanspers
08da28dd60 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

(was: https://webrtc-review.googlesource.com/c/src/+/67300)

Bug: webrtc:9120
Change-Id: I07340505137b16c2dd487569ad0112f984557bba
Reviewed-on: https://webrtc-review.googlesource.com/75125
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23208}
2018-05-14 09:25:49 +00:00
Niels Möller
c6ce9c5938 New file api/video/BUILD.gn
Build targets involving files under api/video/ are moved into this
file, from api/BUILD.gn. In addition, drop "_api" part of target
names, and move the header file api/videosinkinterface.h to
api/video/video_sink_interface.h.

Bug: webrtc:9253
Change-Id: I2896d3f063db8dff902bc29738578395b2fcc155
Reviewed-on: https://webrtc-review.googlesource.com/75500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23207}
2018-05-14 06:57:38 +00:00
Max Morin
826738b78c Revert "Moving iOS Audio Device to sdk."
This reverts commit a167212657.

Reason for revert: Breaks Chromium build.
Log:
https://ci.chromium.org/buildbot/chromium.webrtc.fyi/ios-device/
Writing """\
additional_target_cpus = [ "arm64" ]
goma_dir = "/b/c/goma_client"
ios_enable_code_signing = false
is_component_build = false
is_debug = false
target_cpu = "arm"
target_os = "ios"
use_goma = true
""" to /b/c/b/ios_device/src/out/Release-iphoneos/args.gn.
/b/c/b/ios_device/src/buildtools/mac/gn gen //out/Release-iphoneos --check
  -> returned 1
ERROR at //third_party/webrtc/sdk/BUILD.gn:108:9: Can't load input file.
        "../../rtc_base:checks",
        ^----------------------
Unable to load:
  /b/c/b/ios_device/src/third_party/rtc_base/BUILD.gn
I also checked in the secondary tree for:
  /b/c/b/ios_device/src/build/secondary/third_party/rtc_base/BUILD.gn

Original change's description:
> Moving iOS Audio Device to sdk.
> 
> This change forks the existing iOS audio device module and audio device
> from modules/audio_device/ into sdk/objc/Framework. It also updates
> RTCPeerConnectionFactory to use the forked implementation.
> 
> The unit tests are re-implemented as XCTests.
> 
> Bug: webrtc:9120
> Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
> Reviewed-on: https://webrtc-review.googlesource.com/67300
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23163}

TBR=andersc@webrtc.org,kthelgason@webrtc.org,peterhanspers@webrtc.org

Change-Id: Iebe52e9775409a3bdd6d5e44f4f985d56b859cbe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9120
Reviewed-on: https://webrtc-review.googlesource.com/75220
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23166}
2018-05-08 11:00:37 +00:00
Peter Hanspers
a167212657 Moving iOS Audio Device to sdk.
This change forks the existing iOS audio device module and audio device
from modules/audio_device/ into sdk/objc/Framework. It also updates
RTCPeerConnectionFactory to use the forked implementation.

The unit tests are re-implemented as XCTests.

Bug: webrtc:9120
Change-Id: Ie60cafae796efbd7966d21ff6877c92cbe850fb7
Reviewed-on: https://webrtc-review.googlesource.com/67300
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23163}
2018-05-08 08:46:25 +00:00
Karl Wiberg
7ba22b8eea Break out the part of the iSAC codec that's used for Voice Activity Detection
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.

Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
2018-05-04 08:53:34 +00:00
Peter Hanspers
1c62b986d9 Adding a Metal RGB renderer.
The new RTCMTLRGBRenderer dynamically handles both the kCVPixelFormatType_32BGRA
and the kCVPixelFormatType_32ARGB pixel formats.

Change-Id: I935532f762eff74c4b84fea9b855191f4c321fb7
Bug: webrtc:9200
Reviewed-on: https://webrtc-review.googlesource.com/72482
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@{#23100}
2018-05-03 12:57:41 +00:00
Magnus Jedvert
8b4e92d0a5 ObjC SDK: Stop using built-in SW video codecs
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}
2018-05-02 10:15:56 +00:00
Yura Yaroshevich
0f77feae6d Init max supported H.264 profile at runtime on iOS
Bug: webrtc:9134, webrtc:7992
Change-Id: Id24c570bf3296298901f61ee817a3d7c3f8c6347
Reviewed-on: https://webrtc-review.googlesource.com/71560
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23034}
2018-04-26 15:01:07 +00:00
Anders Carlsson
1f433e46db Mark built-in software video codecs as poisonous.
The goal is to make these injectable, and only VP8 and VP9 specific
targets should depend on them.

Bug: webrtc:7925
Change-Id: Ie9239a54d197fe70c93de0582797211fef6997a2
Reviewed-on: https://webrtc-review.googlesource.com/72082
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23021}
2018-04-25 11:34:33 +00:00
Karl Wiberg
bb23c838f5 GN hack to tag targets as poisonous (and use it with audio codecs)
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.

This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).

Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
2018-04-23 13:41:47 +00:00
Kári Tristan Helgason
06d094f3e6 Add renderer-agnostic delegate protocol.
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}
2018-04-19 16:18:49 +00:00
Niels Möller
2cb7b5ebef Convert BitrateAdjuster from webrtc::Clock to rtc::TimeMillis.
We can then also drop the system_wrappers dependency from the common_video
build target.

Bug: webrtc:6733
Change-Id: I501113d100322d1ebc51b2286970697a24b70a43
Reviewed-on: https://webrtc-review.googlesource.com/70381
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22934}
2018-04-19 09:22:08 +00:00
Karl Wiberg
5817d3dfaa AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.

BUG=webrtc:5801, webrtc:8396

Change-Id: I80749ec3b86cba73994307046d05964f59167d44
Reviewed-on: https://webrtc-review.googlesource.com/18440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22774}
2018-04-06 15:10:27 +00:00
Anders Carlsson
fe9d8178df Reland "Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource."
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}
2018-04-03 11:35:40 +00:00
JT Teh
35d052c2a3 Revert "Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource."
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}
2018-03-30 00:49:48 +00:00
Anders Carlsson
4ea50c2b42 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}
2018-03-28 16:47:06 +00:00
Anders Carlsson
c3d1e09a25 Make sure RTCMTLVideoView.h ends up in framework headers.
Needs to be added to the array before the array is copied to the
sources and public_headers arrays.

Bug: None
Change-Id: If41fd1c882dd17e4007b62c9c7a49f196849dd12
Reviewed-on: https://webrtc-review.googlesource.com/63640
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22531}
2018-03-21 12:44:58 +00:00
Anders Carlsson
7311918269 Add an example app for iOS native API.
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}
2018-03-19 09:31:06 +00:00
Anders Carlsson
9823ee47d3 Fix native api in preparation for native_api example.
Add native api conversions for video frames and video renderer. This
also requires some changes to sdk/BUILD to avoid cyclic dependencies.

Bug: webrtc:8832
Change-Id: Ibf21e63bdcae195dcb61d63f9262e6a8dc4fa790
Reviewed-on: https://webrtc-review.googlesource.com/57142
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22340}
2018-03-08 13:22:13 +00:00
Kári Tristan Helgason
a2d89fc9f5 Remove ObjC static library target.
It's been unused for a while and starting to be a maintainance burden.

Bug: webrtc:8943
Change-Id: Ie49d6b06bdeb002496007725009ea194b8130f2b
Reviewed-on: https://webrtc-review.googlesource.com/60160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22339}
2018-03-08 12:22:13 +00:00
Steve Anton
8cb344acfd Add new PeerConnection APIs to the ObjC SDK
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}
2018-02-28 01:28:57 +00:00
Mirko Bonadei
d68956db0e sdk:metal_objc should be publicly visible.
TBR=kthelgason@webrtc.org,phoglund@webrtc.org

Bug: webrtc:8855
Change-Id: I1a1fdb57d908b479bac1870c4709ebfbeedc1565
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/54305
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22058}
2018-02-16 17:05:28 +00:00
Kári Tristan Helgason
99bf77c851 Fix issues found by gn check.
It turns out that some headers were not owned by any targets.
These were:
RTCVideoCodec.h
RTCVideoCodecFactory.h
RTCVideoCodecH264.h
RTCVideoEncoderVP8.h
RTCVideoDecoderVP8.h
RTCVideoEncoderVP9.h
RTCVideoDecoderVP9.h

And some were owned by multiple targets, namely:
RTCPeerConnectionFactory+Native.h
RTCPeerConnectionFactory+Private.h
RTCVideoFrameBuffer.h

These have all been moved to their appropriate homes.

This CL also fixes a lot of cyclic interdependencies in the iOS sdk build files.

Bug: webrtc:8855
Change-Id: I1b7ddb6c2a93868d1510ccf0a64bd3dd169ec3e7
Reviewed-on: https://webrtc-review.googlesource.com/49060
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22052}
2018-02-16 12:36:08 +00:00
Niels Möller
8366e177e7 Rename Call::Config to CallConfig, keep old name as alias.
We want api/peerconnectioninterface.h (and corresponding build target)
to not depend on call.h, and generally we treat Call as an internal,
non-api, class. But we need CallFactoryInterface in the api in order to
enable use of PeerConnection with or without support for media.

Making CallConfig a top-level class makes it possible to forward declare
it, together with Call, for use in callfactoryinterface.h and
peerconnectioninterface.h.

Delete the peerconnection_and_implicit_call_api target, replaced by
new target callfactory_api, to link between Call and Peerconnection.

Bug: webrtc:7504
Change-Id: I5e3978ef89bcd6705e94536f8676bcf89fc82fe1
Reviewed-on: https://webrtc-review.googlesource.com/46201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22020}
2018-02-14 15:14:39 +00:00
Kári Tristan Helgason
0d3c9a3f2b Delete RTCAVFoundationVideoSource and related classes.
Bug: webrtc:8852
Change-Id: Ie073fe3f7bafc3d22fafef51f659e340d5a9250f
Reviewed-on: https://webrtc-review.googlesource.com/48620
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21985}
2018-02-12 14:41:25 +00:00
Anders Carlsson
c974b4bbc7 Remove corevideoframebuffer target.
This target is deprecated and downstream projects have been updated.

This CL replaces https://webrtc-review.googlesource.com/c/src/+/46521

Bug: webrtc:8470
Change-Id: Icf4696c946fd0a1aeeb687c4960586ba0cc52dc0
Reviewed-on: https://webrtc-review.googlesource.com/48362
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21908}
2018-02-06 11:03:39 +00:00
Anders Carlsson
3ff50fba59 Create experimental Obj-C++ API.
This can be used to wrap Objective-C components in C++ classes, so users
can use the WebRTC C++ API directly together with the iOS specific
components provided by our SDK.

Bug: webrtc:8832
Change-Id: I6d34f7ec62d51df8d3a5340a2e17d30ae73e13e8
Reviewed-on: https://webrtc-review.googlesource.com/46162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21850}
2018-02-01 16:36:24 +00:00
Anders Carlsson
3b3364ed2e Add new Logger to iOS SDK to be able to intercept logs and report them to the appropriate system
Bug: webrtc:8695
Change-Id: I4f81529222dcaf3dded1a0ab93fabf5cda1fdef8
Reviewed-on: https://webrtc-review.googlesource.com/37001
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21821}
2018-01-31 09:35:29 +00:00
Anders Carlsson
dd8c16574e Enable building WebRTC without built-in software codecs
This CL adds a GN build flag to include builtin software codecs
(enabled by default).

When setting the flag to false, libvpx can also be excluded. The
benefit is that the resulting binary is smaller.

Replaces https://webrtc-review.googlesource.com/c/src/+/29203

Bug: webrtc:7925
Change-Id: Id330ea8a43169e449ee139eca18e4557cc932e10
Reviewed-on: https://webrtc-review.googlesource.com/36340
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21818}
2018-01-31 08:33:59 +00:00
Anders Carlsson
e7dd83f2a7 Add tests for starting and stopping RTCCameraVideoCapturer.
Bug: webrtc:8755
Change-Id: I07d9a203276359069af7ba384c58612df7f2b467
Reviewed-on: https://webrtc-review.googlesource.com/40240
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21692}
2018-01-19 10:54:12 +00:00
Ilya Nikolaevskiy
2ffe3e80db Reland Use runtime enabled features API to enable dual stream mode
This is an unchanged patch after dependency fixes in downstream projects are implemented.

Original patch was reviewed here:
https://webrtc-review.googlesource.com/c/src/+/39008

TBR=phoglund@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org,lliuu@webrtc.org

Change-Id: I648bbf63d34282a48cabc854615005ec65b28cb3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8287
Reviewed-on: https://webrtc-review.googlesource.com/40420
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21674}
2018-01-18 12:22:49 +00:00
Lu Liu
c1094eb81d Revert "Use runtime enabled features API to enable dual stream mode"
This reverts commit 6f011dfdd4.

Reason for revert: Broke internal builds

Original change's description:
> Use runtime enabled features API to enable dual stream mode
> 
> Bug: webrtc:8287
> Change-Id: I1a366d959a8b7f2a704baa7ea8ace64c1c398d52
> Reviewed-on: https://webrtc-review.googlesource.com/39008
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21661}

TBR=phoglund@webrtc.org,ilnik@webrtc.org,nisse@webrtc.org,philipel@webrtc.org

Change-Id: I0af406066231b67dd0b8eb6808bdc3e3f77560b6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8287
Reviewed-on: https://webrtc-review.googlesource.com/40321
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21663}
2018-01-17 19:45:48 +00:00
Ilya Nikolaevskiy
6f011dfdd4 Use runtime enabled features API to enable dual stream mode
Bug: webrtc:8287
Change-Id: I1a366d959a8b7f2a704baa7ea8ace64c1c398d52
Reviewed-on: https://webrtc-review.googlesource.com/39008
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21661}
2018-01-17 16:29:37 +00:00
Anders Carlsson
dc6b477fb4 Generate iOS framework umbrella header.
Instead of keeping the umbrella header in sync manually and needing
ifdefs to make it include the correct headers depending on platform,
generate it based on the headers we include in the framework target.

Can also be used to only include internal software codec headers when
compiling with support for them.

Bug: webrtc:7925
Change-Id: I63f97af1efc8710cfd62d527fcb343fed05daae2
Reviewed-on: https://webrtc-review.googlesource.com/38702
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@{#21613}
2018-01-15 13:43:59 +00:00
Mirko Bonadei
75baa498fa Stop using public_deps in media/.
Bug: webrtc:8603
Change-Id: I7a6dad323ac298dc784feb5aa1fdc2ae5876cb5c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/33180
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21591}
2018-01-12 08:15:01 +00:00
Mirko Bonadei
34814c728f Stop using public_deps in api/.
Bug: webrtc:8603
Change-Id: I12f91365fa9a3ed12be1a9658999b6f6aad7e5e3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/33220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21584}
2018-01-11 14:50:11 +00:00
Per Kjellander
a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
This reverts commit c73e1f4378.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
2018-01-10 15:55:04 +00:00
Per Kjellander
c73e1f4378 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
This reverts commit 588c548657.

Reason for revert: 

Breaks Chrome FYI:

/b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
  -> returned 1
ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
    static_library(target_name) {
    ^----------------------------
The item //content/renderer:renderer
can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
  //third_party/webrtc/*
  //third_party/webrtc_overrides/*
]

 https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout

Original change's description:
> GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> 
> This means that by default, targets are visible to everything under
> the WebRTC root, but not visible to anything else.
> 
> API targets are manually tagged with visibility "*", so that targets
> outside the WebRTC tree can see them.
> 
> BUG=webrtc:8254
> 
> Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> Reviewed-on: https://webrtc-review.googlesource.com/24140
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21548}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38760
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21555}
2018-01-10 15:14:54 +00:00
Karl Wiberg
588c548657 GN rtc_* templates: Set default visibility to webrtc_root + "/*"
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.

API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.

BUG=webrtc:8254

Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
2018-01-10 13:08:11 +00:00
Peter Hanspers
8020ae82f5 Adding test to check for c++ in framework headers.
Bug: webrtc:8469
Change-Id: I336a2aa75638920901c2bddf07fb03cb00ccb83e
Reviewed-on: https://webrtc-review.googlesource.com/38020
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21536}
2018-01-09 15:17:30 +00:00