Commit graph

91 commits

Author SHA1 Message Date
Anders Carlsson
7bca8ca4e2 Obj-C SDK Cleanup
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}
2018-08-30 10:42:41 +00:00
Magnus Jedvert
3e169ac18c Revert "Update video_quality_analysis to align videos instead of using barcodes"
This reverts commit d65e143801.

Reason for revert: Breaks perf bots. frame_analyzer is a prebuilt binary, so it won't automatically pick up changes in the .cc file.

Original change's description:
> Update video_quality_analysis to align videos instead of using barcodes
> 
> This CL is a follow-up to the previous CL
> https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
> logic for aligning videos. This will allow us to easily extend
> video_quality_analysis with new sophisticated video quality metrics.
> Also, we can use any kind of video that does not necessarily need to
> contain bar codes. Removing the need to decode barcodes also leads to a
> big speedup for the tests.
> 
> Bug: webrtc:9642
> Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
> Reviewed-on: https://webrtc-review.googlesource.com/94845
> Reviewed-by: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24423}

TBR=phoglund@webrtc.org,magjed@webrtc.org,phensman@webrtc.org

Change-Id: Ia590b465687b861fe37ed1b14756d4607ca90da1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9642
Reviewed-on: https://webrtc-review.googlesource.com/95946
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24428}
2018-08-24 12:45:13 +00:00
Magnus Jedvert
d65e143801 Update video_quality_analysis to align videos instead of using barcodes
This CL is a follow-up to the previous CL
https://webrtc-review.googlesource.com/c/src/+/94773 that added generic
logic for aligning videos. This will allow us to easily extend
video_quality_analysis with new sophisticated video quality metrics.
Also, we can use any kind of video that does not necessarily need to
contain bar codes. Removing the need to decode barcodes also leads to a
big speedup for the tests.

Bug: webrtc:9642
Change-Id: I74b0d630b3e1ed44781ad024115ded3143e28f50
Reviewed-on: https://webrtc-review.googlesource.com/94845
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24423}
2018-08-24 09:26:14 +00:00
Niels Möller
f06f923ef0 Delete almost all use of MediaConstraintsInterface in the PeerConnection API
Bug: webrtc:9239
Change-Id: I04f4370f624346bf72c7e4e090b57987b558213b
Reviewed-on: https://webrtc-review.googlesource.com/74420
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24396}
2018-08-23 07:14:37 +00:00
Sami Kalliomäki
508e23421f Remove unnecessary //base:base_java dependencies.
WebRTC code shouldn't depend on Chromium Android base code.

Bug: None
Change-Id: Ie094f26e4ee855769c9c5276bbb47242aae9c217
Reviewed-on: https://webrtc-review.googlesource.com/92387
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24194}
2018-08-06 12:04:35 +00:00
Sami Kalliomäki
d54f5f5c76 Rename rtc_instrumentation_test_apk targets to end with _test_apk.
This matches Chromium pattern of naming instrumentation test apks with
a name ending in _test_apk. The old naming confuses generate_gradle.py.

Renames:
 - AppRTCMobileTest
   -> AppRTCMobile_test_apk
 - AppRTCMobileTestStubbedVideoIO
   -> AppRTCMobile_stubbed_video_io_test_apk
 - libjingle_peerconnection_android_unittest
   -> android_instrumentation_test_apk

Bug: webrtc:9588
TBR: phoglund
Change-Id: Idb82dc4bd089bc7c90e9373f7c3d572f9fd2d95a
Reviewed-on: https://webrtc-review.googlesource.com/92380
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24184}
2018-08-03 12:44:40 +00:00
Artem Titov
e41c433502 Move sigslot to proper third_party directory
Extract sigslot into separate target and move it to proper third_party
directory.

Bug: webrtc:8366
Change-Id: Id2e0712bd020bfad811947803c94553dce06d976
Reviewed-on: https://webrtc-review.googlesource.com/84141
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24099}
2018-07-25 14:53:33 +00:00
Sami Kalliomäki
903bd4145f Move VideoCapturer to video_api and cleanup.
This should only be landed after clients have been given time to
upgrade to the new interface.

Bug: webrtc:9496, webrtc:9181
Change-Id: Ideb37637d9f0b9a3a9748811879c263c64f81d11
Reviewed-on: https://webrtc-review.googlesource.com/87308
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24080}
2018-07-24 10:38:57 +00:00
Mirko Bonadei
f957dd9c7f Remove -Wno-reorder from examples/BUILD.gn.
Bug: webrtc:9251
Change-Id: I447469e94ad938eaaa628edcd1a57cdeac6ed603
Reviewed-on: https://webrtc-review.googlesource.com/88123
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23928}
2018-07-11 09:31:24 +00:00
Mirko Bonadei
f165e4745e Removing -Wno-sentinel from examples/BUILD.gn.
Bug: webrtc:9251
Change-Id: I7fdaab101bdd79a9fbaf57df6e2df9368614cf78
Reviewed-on: https://webrtc-review.googlesource.com/87921
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23924}
2018-07-11 08:04:22 +00:00
Mirko Bonadei
879f7881b4 Add missing dependencies on //third_party/abseil-cpp/absl/memory.
Abseil uses -isystem and -imsvc because of some warnings, these two
flags are not used by "gn check", this introduced some regressions.

CL https://chromium-review.googlesource.com/c/chromium/src/+/1124478
will try to switch back absl to -I.

Bug: None
Change-Id: I52e857ef1d11831393c35a1bee09479b83827bad
Reviewed-on: https://webrtc-review.googlesource.com/88121
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23923}
2018-07-11 07:56:07 +00:00
Mirko Bonadei
3e603ec4a6 Remove -Wno-sign-compare, -Wno-unused-variable and -Wno-format.
Bug: webrtc:9251
Change-Id: Ide719100cb99fdf5fe6700af9d5f4a1884ac2b67
Reviewed-on: https://webrtc-review.googlesource.com/87842
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23905}
2018-07-10 12:55:07 +00:00
Anders Carlsson
1e06d885dc Migrate legacy Android video codec factories.
Prepare for building without built-in software codecs. When passing
null, inject the new type of factories but wrap them in the built-in
software codecs outside the videoengine.

Bug: webrtc:7925
Change-Id: I7408e6e46e6b9efdf346852954bf51a97e023b5c
Reviewed-on: https://webrtc-review.googlesource.com/83729
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23897}
2018-07-10 08:01:05 +00:00
Mirko Bonadei
bd216f2854 Removing /wd4245 from examples/.
Bug: webrtc:9251
Change-Id: I335648b622ba49ca4a5bd501db2fb4dad924f2ea
Reviewed-on: https://webrtc-review.googlesource.com/87143
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23895}
2018-07-10 07:14:55 +00:00
Karl Wiberg
918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
Instead of going through our wrappers in ptr_util.h.

This CL was generated by the following script:

  git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
  git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
  git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
  git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
  git cl format

Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.

Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 10:59:49 +00:00
Kári Helgason
aaa483bff6 Revert "Remove deprecated mac capture code."
This reverts commit 3f1d15b352.

Reason for revert: Removing this breaks a debugging tool that people relied on. I will update that tool to use the new capturer before relanding this.

Original change's description:
> Remove deprecated mac capture code.
> 
> Bug: webrtc:6898, webrtc:6333, webrtc:7861
> Change-Id: Ie33eaa47585012f98b59ccffc0c849c1d9da54da
> Reviewed-on: https://webrtc-review.googlesource.com/79920
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23454}

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

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:6898, webrtc:6333, webrtc:7861
Change-Id: Ifc367eecfe92a2b2e4a826a820dc9c3c970ea01e
Reviewed-on: https://webrtc-review.googlesource.com/84380
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23681}
2018-06-20 12:24:16 +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
Kári Tristan Helgason
3f1d15b352 Remove deprecated mac capture code.
Bug: webrtc:6898, webrtc:6333, webrtc:7861
Change-Id: Ie33eaa47585012f98b59ccffc0c849c1d9da54da
Reviewed-on: https://webrtc-review.googlesource.com/79920
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23454}
2018-05-30 15:40:01 +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
Anders Carlsson
6753795409 Built in video codec factories.
To prepare for making the software codecs optional and injectable, these
codec factories provide a way to pass in identical factories as were the
default old behaviour.

Bug: webrtc:7925
Change-Id: I0c70fa3c56c999e9d1af6e172eff2fbba849e921
Reviewed-on: https://webrtc-review.googlesource.com/71162
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23096}
2018-05-03 11:49:42 +00:00
Sami Kalliomäki
7d8f5949b2 Make depending on a specific audio implementation optional.
Splits out audio_java into audio_api_java and
java_audio_device_module_java.

Makes depending on java_audio_device_module_jni optional for clients
that do not use it. It is only necessary to depend on this target if
depending on java_audio_device_module_java.

Also some cleanup.

Bug: webrtc:7452
Change-Id: Ic6c4dbe11db3ed8330802a8e90203acb8ef18e72
Reviewed-on: https://webrtc-review.googlesource.com/70220
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22981}
2018-04-23 14:18:47 +00:00
Sami Kalliomäki
1641ca3dd3 Split out video targets from //sdk/android:base_java.
Bug: webrtc:9048
Change-Id: Icda0fabf41610f99254d244e0b11d321eee345f7
Reviewed-on: https://webrtc-review.googlesource.com/65120
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22752}
2018-04-05 16:02:09 +00:00
Magnus Jedvert
e2be7ee2e8 Android: Split out audio device targets
This CL splits out the audio device module Java code into a separate
target, and also splits up the audio device module implementations into
three different build targets, one for OpenSLES, AAudio, and the Java
based implementation.

Bug: webrtc:7452, webrtc:9048
Change-Id: I8ec09c73580b468837223ddd420fb29ca61fdea5
Reviewed-on: https://webrtc-review.googlesource.com/66461
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22727}
2018-04-04 11:50:30 +00:00
Sami Kalliomäki
725c106d8c Split out peerconnection, swcodec and hwcodec targets from Android SDK.
Bug: webrtc:9048
Change-Id: I3094ca8993cd754a0ea799e325f941d3ffd5578b
Reviewed-on: https://webrtc-review.googlesource.com/63700
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22644}
2018-03-28 10:46:26 +00:00
Paulina Hensman
17071682f7 Add setting to switch between new and legacy AudioDeviceModule in AppRTC
To facilitate testing both the old and new AudioDeviceModule path, a
setting is added to AppRTC. Enable "Use legacy audio device" to use
the old path.

Bug: webrtc:7452
Change-Id: I221378ac7bb0fa4e543c3fd081c7a322621621a0
Reviewed-on: https://webrtc-review.googlesource.com/64760
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22609}
2018-03-26 14:57:00 +00:00
Sami Kalliomäki
06e363a6e6 Annotate AppRTCMobile with @Nulllable.
Bug: webrtc:8881
Change-Id: I32390b1e2586415757f0453b60d35b23160d2862
Reviewed-on: https://webrtc-review.googlesource.com/63641
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22580}
2018-03-23 12:42:05 +00:00
Magnus Jedvert
08006d4133 Android AppRTCMobile: Use new audio device code
This CL contains some follow-up fixes for
https://webrtc-review.googlesource.com/c/src/+/60541. It removes all use
of the old voiceengine implementation from AppRTCMobile.

Bug: webrtc:7452
Change-Id: Iea21a4b3be1f3cbb5062831164fffb2c8051d858
Reviewed-on: https://webrtc-review.googlesource.com/63480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22530}
2018-03-21 12:40:18 +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
Sami Kalliomäki
3e77afd0d2 Add an example app for Android native API.
The app is a simple loopback demo demonstrating the usage of Android
native API. This is an initial version and I will add support for
HW codecs etc. in the future.

Bug: webrtc:8769
Change-Id: Ifb6209769dabeb8ca3185b969a1ef8afd6d84390
Reviewed-on: https://webrtc-review.googlesource.com/60540
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22385}
2018-03-12 14:22:59 +00:00
George Zhou
2770c3df91 Synced webrtc_unity_plugin to the latest webrtc and fixed bugs.
1, Let targets libwebrtc_unity and webrtc_unity_plugin built with Ninja -C out/***.
2, Fixed compile issue of libwebrtc_unity.
3, Built libwebrtc_unity classes into Java 7 instead of Java 8 for android.
4, Added an interface to enable peerconnectionFactory for android in Unity.

Bug: webrtc:8986
Change-Id: I2a206a77ab38895ec9ac845ce89507d61076d396
Reviewed-on: https://webrtc-review.googlesource.com/59000
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Commit-Queue: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22373}
2018-03-09 18:34:14 +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
henrika
5641fbb5ec Add support for saving local audio input to file in AppRTCMobile
Uses new WebRtcAudioRecordSamplesReadyCallback which was added recently in
https://webrtc-review.googlesource.com/c/src/+/49981.

This CL:
- Serves as a test of new WebRtcAudioRecordSamplesReadyCallback.
- Useful for debugging purposes since it records the most native raw audio.

Bug: None
Change-Id: I57375cbf237c171e045b0bdb05f7ae1401930fbc
Reviewed-on: https://webrtc-review.googlesource.com/53120
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22128}
2018-02-21 14:09:56 +00:00
Patrik Höglund
a425184a04 Fix override warnings.
Bug: webrtc:6306
Change-Id: I5c01139475a75d56a9642943eff527eaf036c738
Reviewed-on: https://webrtc-review.googlesource.com/55522
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22126}
2018-02-21 13:48:57 +00:00
Mirko Bonadei
cbaaaed418 Re-enabling 'gn check' on //examples/*.
Bug: webrtc:8850
Change-Id: I7397bc17e0f790a31d417d1bf5db6c5f9e32088d
No-Presubmit: true
Reviewed-on: https://webrtc-review.googlesource.com/54314
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22085}
2018-02-19 15:07:45 +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
Qingsi Wang
a4e72b7914 Add RtcEventLog to AppRTCMobile with preference setting.
Enable diagnostic packet and event recording as in the "webrtc-internal"
setting in Chromium.

Bug: webrtc:8859
Change-Id: I1d4a19e0dd60133cdd0d4e18a55780623b65653c
Reviewed-on: https://webrtc-review.googlesource.com/49541
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21987}
2018-02-12 19:12:05 +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
Edward Lemur
70ba9b453c Reland "Make it possible to run video_quality_loopback_test in swarming."
This is a reland of 1175ecd095.

Original change's description:
> Make it possible to run video_quality_loopback_test in swarming.
> 
> I made two changes to video_quality_loopback_test to make it possible
> to run it on swarming.
> 
> 1. We need to know the path to frame_analyzer when we're generating
>    the build files, and it must be already present.
>    I made frame_analyzer a resource, so it's downloaded to a known path
>    before generating the build files.
> 2. The .zip files for apprtc and golang are downloaded and isolated.
>    The script now extracts them and installs AppRTC.
> 
> Passing task:
>  https://chromium-swarm.appspot.com/task?id=3b230bcc04128210
> 
> Bug: chromium:755660
> Change-Id: I34090897402421d5b7e29f21fbed354551197f92
> Reviewed-on: https://webrtc-review.googlesource.com/40920
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21716}

Bug: chromium:755660
Change-Id: Ie3ca62d64b4fe856485287b7d9d3d9e3f75dc091
Reviewed-on: https://webrtc-review.googlesource.com/42860
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21733}
2018-01-23 13:03:17 +00:00
Edward Lemur
9c1848952e Revert "Make it possible to run video_quality_loopback_test in swarming."
This reverts commit 1175ecd095.

Reason for revert: Breaks the bots.

Original change's description:
> Make it possible to run video_quality_loopback_test in swarming.
> 
> I made two changes to video_quality_loopback_test to make it possible
> to run it on swarming.
> 
> 1. We need to know the path to frame_analyzer when we're generating
>    the build files, and it must be already present.
>    I made frame_analyzer a resource, so it's downloaded to a known path
>    before generating the build files.
> 2. The .zip files for apprtc and golang are downloaded and isolated.
>    The script now extracts them and installs AppRTC.
> 
> Passing task:
>  https://chromium-swarm.appspot.com/task?id=3b230bcc04128210
> 
> Bug: chromium:755660
> Change-Id: I34090897402421d5b7e29f21fbed354551197f92
> Reviewed-on: https://webrtc-review.googlesource.com/40920
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21716}

TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org

Change-Id: Id25d26adc547ff6f9ab178601e37527459c8b5ef
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/42800
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21723}
2018-01-22 16:12:10 +00:00
Edward Lemur
1175ecd095 Make it possible to run video_quality_loopback_test in swarming.
I made two changes to video_quality_loopback_test to make it possible
to run it on swarming.

1. We need to know the path to frame_analyzer when we're generating
   the build files, and it must be already present.
   I made frame_analyzer a resource, so it's downloaded to a known path
   before generating the build files.
2. The .zip files for apprtc and golang are downloaded and isolated.
   The script now extracts them and installs AppRTC.

Passing task:
 https://chromium-swarm.appspot.com/task?id=3b230bcc04128210

Bug: chromium:755660
Change-Id: I34090897402421d5b7e29f21fbed354551197f92
Reviewed-on: https://webrtc-review.googlesource.com/40920
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21716}
2018-01-22 14:40:28 +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
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
Anders Carlsson
a114c88e78 Reland "Reland "Reland "Put internal video codec factories into separate target"""
This is a reland of 727b7d0470
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
> 
> This is a reland of 0efd1e8b7e
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> > 
> > This is a reland of 51698aefd4
> > Original change's description:
> > > Put internal video codec factories into separate target
> > > 
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > > 
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > > 
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> > 
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
> 
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}

Bug: webrtc:7925
Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454
Reviewed-on: https://webrtc-review.googlesource.com/37000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21513}
2018-01-08 09:29:52 +00:00
Patrik Höglund
be214a26f8 Move videosinkinterface.h to common_video to solve a circular dep.
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.

Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
2018-01-04 13:19:49 +00:00
Mirko Bonadei
4f024efe7f Renaming android_library android_manifest to android_manifest_for_lint.
The attribute android_manifest for android_library targets has been
removed in [1]. This CL renames it to android_manifest_for_lint (to
avoid lint errors) in all the rtc_android_library targets.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/848079

Bug: webrtc:8707
Change-Id: Ifa127790937fa49ed52d6aab0c7ce5ab03e1177b
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/37440
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21493}
2018-01-04 12:55:49 +00:00
Patrik Höglund
49acb1e8ca Add ARDAppDelegate.h.
Bug: webrtc:7643
Change-Id: Ib1980c82c64ffa3acc009a855a7f077509872548
Reviewed-on: https://webrtc-review.googlesource.com/37002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21476}
2018-01-02 18:13:40 +00:00