Commit Graph

77 Commits

Author SHA1 Message Date
c6c44268bc Moves network control interface to API.
This prepares for allowing injection of a network controller.

Bug: webrtc:9155
Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3
Reviewed-on: https://webrtc-review.googlesource.com/73100
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23188}
2018-05-09 11:01:36 +00:00
ae8d8a1d22 Remove audio_frame.h from module_common_types.h
PSA: https://groups.google.com/forum/?pli=1#!topic/discuss-webrtc/wVztouO08gw

Bug: webrtc:9139, webrtc:7504
Change-Id: I9587513509eb4609e8e4e2e112af58d920b4e334
Reviewed-on: https://webrtc-review.googlesource.com/70700
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22952}
2018-04-20 10:22:53 +00:00
3285897c1f Cleaning up modules_tests resources.
* Removed video files which were not used by any tests.
* Removed ConferenceMotion_1280_720_50.yuv for mobile builds.

Bug: webrtc:8936
Change-Id: I0539e9fce20470fcc2f0af84bd297faffc4b587a
Reviewed-on: https://webrtc-review.googlesource.com/60942
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22359}
2018-03-09 12:57:24 +00:00
f35c6667d6 Separate build targets for aec3 and aec3_unittests
Bug: webrtc:8844
Change-Id: Id6a98eae19aaedc87c3f402a004f58f0290d5c28
Reviewed-on: https://webrtc-review.googlesource.com/56580
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22173}
2018-02-23 13:16:16 +00:00
0dd1b0a4b2 Revert "Revert "Enables PeerConnectionFactory using external fec controller""
This reverts commit 00733015fafbbc61ddc12dfdc88b21a9fcd9d122.

Reason for revert: The reason for a downstream test failure on the original commit and a workaround has been found. Solution is to keep a PeerConnectionFactory constructor implementation as the same as before.

Original change's description:
> Revert "Enables PeerConnectionFactory using external fec controller"
>
> This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3.
>
> Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java
>
> Original change's description:
> > Enables PeerConnectionFactory using external fec controller
> >
> > Bug: webrtc:8799
> > Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
> > Reviewed-on: https://webrtc-review.googlesource.com/43961
> > Commit-Queue: Ying Wang <yinwa@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22038}
>
> TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org
>
> Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8799
> Reviewed-on: https://webrtc-review.googlesource.com/54080
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22040}

TBR=deadbeef@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org

Bug: webrtc:8799
Change-Id: If9f3292bfcc739782967530c49f006d0abbc38a8
Reviewed-on: https://webrtc-review.googlesource.com/55400
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22100}
2018-02-20 12:41:55 +00:00
2ae140ae7e BUILD.gn file for api/audio.
Targets containing files in api/audio are moved from api/BUILD.gn to
api/audio/BUILD.gn.

Bug: webrtc:8844
Change-Id: Ib7ea4b7eb3c2ea38ef8261a1fc5c2b4674985981
Reviewed-on: https://webrtc-review.googlesource.com/54360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22074}
2018-02-19 10:38:29 +00:00
00733015fa Revert "Enables PeerConnectionFactory using external fec controller"
This reverts commit 4f07bdb25567d8ef528311e0b50a62c61d543fc3.

Reason for revert: Speculatively reverting, because downstream test is now hitting "PeerConnectionFactory.initialize was not called before creating a PeerConnectionFactory" error, even though it did call initialize. I don't see how any change in this CL could cause that, but it's the only CL on the blamelist, and it does modify PeerConnectionFactory.java

Original change's description:
> Enables PeerConnectionFactory using external fec controller
> 
> Bug: webrtc:8799
> Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
> Reviewed-on: https://webrtc-review.googlesource.com/43961
> Commit-Queue: Ying Wang <yinwa@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22038}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,yinwa@webrtc.org

Change-Id: I95868c35d6f9973e0ebf563814cd71d0fcbd433d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8799
Reviewed-on: https://webrtc-review.googlesource.com/54080
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22040}
2018-02-15 20:07:24 +00:00
4f07bdb255 Enables PeerConnectionFactory using external fec controller
Bug: webrtc:8799
Change-Id: Ieb2cf6163b9a83844ab9ed4822b4a7f1db4c24b8
Reviewed-on: https://webrtc-review.googlesource.com/43961
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22038}
2018-02-15 16:58:26 +00:00
d377f04194 Move AudioFrame to its own header file and target in api/.
This breaks the dependency api:audio_mixer_api --> modules:module_api,
and allows peerconnectioninterface.h to include audio_mixer.h, without
introducing a dependency cycle.

In addition, un-inline all AudioFrame methods, moving implementations
to audio_frame.cc, and replace assert by RTC_CHECK_*.

Bug: webrtc:7504
Change-Id: I11e3d3d22716e9b98976bf830103fbb06e7bbb77
Reviewed-on: https://webrtc-review.googlesource.com/51860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22016}
2018-02-14 11:01:53 +00:00
10d9d59db1 Adding simulcast/spatial layering support to VideoProcessor.
Encoded frames are preserved and decoded after all layers are
encoded.
Each spatial layer is decoded with separate decoder.
For quality evaluation of lowres layers original input frame is
downscaled with bilinear interpolation.
Encoded and decoded frames are dumped into separate files.

For async codecs encoded frames are passed to decoder in encode
callback, as before.

Bug: webrtc:8524
Change-Id: Idb0c92c7274c1915cff9a011a2794f1cf4bc8cb1
Reviewed-on: https://webrtc-review.googlesource.com/43381
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21844}
2018-02-01 13:28:46 +00:00
e48c61fca7 Delete unused MediaFile module.
Delete the subdirectory modules/media_file, and all references to it.

Bug: none
Change-Id: I19d86420a7d1d51cb6174c914a90484918106c5a
Reviewed-on: https://webrtc-review.googlesource.com/40540
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21790}
2018-01-29 11:18:18 +00:00
65ce31158f Removing useless dependencies on //testing/gmock.
If a WebRTC build target requires gmock it has to include
test/gmock.h and just depend on //test:test_support.

Unfortunately //testtest_support was a leaky abstraction because it
wasn't propagating the correct -I compiler flag. To make everything
work, all the targets that use gmock started also to depend on
//testing/gmock (even if they were not including any gmock header
directly).

This CL makes //testtest_support propagate the include path up in the
dependency chain so it is possible to remove unused dependencies.

Note: all_dependent_configs should probably be used in the original
gmock target. There is an ongoing discussion about it. This CL solves
the problem on WebRTC side and it is forward compatible.

TBR=phoglund@webrtc.org

Bug: webrtc:8603
Change-Id: If08daf2ce9a6431a6e881a236743b4ec33b59ea7
Reviewed-on: https://webrtc-review.googlesource.com/44340
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21776}
2018-01-26 13:34:12 +00:00
34924c236c Fix warning 4373.
Looks like all the current warnings were because of a MSVC bug:
https://github.com/google/googletest/blob/master/googlemock/docs/FrequentlyAskedQuestions.md

We can just disable this one for all tests and be done with it.

Bug: webrtc:261
Change-Id: I882a577f832ff71ac61936abebe0ca537088bab8
Reviewed-on: https://webrtc-review.googlesource.com/40840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21691}
2018-01-19 10:37:44 +00:00
9c68613080 Update gn files to support Mozilla build
Bug: webrtc:8670
No-Presubmit: true
Change-Id: I085dc63daa8274b5068540cbf56b6330f40643fa
Reviewed-on: https://webrtc-review.googlesource.com/38920
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21624}
2018-01-16 07:51:23 +00:00
a7f2d84ad1 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""
This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.

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 588c548657b3ddf76e7b3f241263eef7f5799f16.
> 
> 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
c73e1f4378 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.

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
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
75f18fca8e Make building with X11 libraries optional.
Desktop capturing on Linux will be disabled in this case, but everything
can be built without any X11 development libraries installed.

BUG=webrtc:5716,webrtc:8319

Change-Id: I01bd6a4b02816b407be19476e22ff073d264b496
Reviewed-on: https://webrtc-review.googlesource.com/32360
Reviewed-by: Henrik Andreassson (OOO until Jan 2) <henrika@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21462}
2017-12-31 14:31:08 +00:00
3e113438b1 Fix circular dependencies in webrtc_common.
One reason for the circular deps is that common_types.h is a
historical dumping ground for various structs and defines that
are believed to be generally useful. I tried moving things out
that did not appear to be used downstream (StreamCounters,
RtpCounters etc) and moved the things that seemed used
(RtpHeader + supporting structs) to a new file api/rtp_headers.h.
This makes their place in the api more clear while moving out
the things that don't belong in the API in the first place.

I had to extract out typedefs.h from webrtc_common to resolve
another circular dependency. I believe checks includes typedefs,
but common depends on checks.

Bug: webrtc:7745
Change-Id: I725d49616b1ec0cdc8b74be7c078f7a4d46f084b
Reviewed-on: https://webrtc-review.googlesource.com/33001
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21295}
2017-12-15 14:33:26 +00:00
28fe510b2f Stop using public_deps in modules/BUILD.gn.
Bug: webrtc:8603
Change-Id: I508473d840d069725f73524a36536633db13e93b
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/33200
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21288}
2017-12-15 09:42:16 +00:00
e9619f8f81 Add a new NetEq decoding unit test for Opus with DTX
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.

Also adding a new resource file which is encoded using Opus with DTX.

Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
2017-11-28 10:45:38 +00:00
558cabf670 Refactor RtpToNtpEstimator and MovingMedianFilter
In preparation for a new remote clock estimation:
RtpToNtpEstimator will now store unwrapped rtp timestamps in
|measurements| and |calculated| flag is out of |Parameters| struct.

Additional missing |Reset| method added to MovingMedianFilter and
PercentileFilter

Bug: webrtc:8468
Change-Id: I1bb454ba0c93720cbb30bfded19677daaf9f721c
Reviewed-on: https://webrtc-review.googlesource.com/22020
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20669}
2017-11-14 10:34:56 +00:00
b5b5bcee72 Separate i420 and i444 implementations to separate targets.
This means we can properly declare the dependency between
libjingle_peerconnection_api and video_frame_api. i420
pulls in system_wrappers, which can't be a dependency of
the public API.

Plan:
1) Land this CL + send out PSA
2) Make all direct users of i420_buffer depend on the
   new video_frame_api_i420 target
3) Move i420_buffer.cc to the new target
4) Make libjingle_peerconnection_api depend on
   video_frame_api, since it no longer contains i420 code

Bug: webrtc:7504
Change-Id: I30d90f2ac7af53748859bbde8aed92386d5501f9
Reviewed-on: https://webrtc-review.googlesource.com/9382
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20656}
2017-11-13 14:27:39 +00:00
2397b9a114 Remove voe::OutputMixer and AudioConferenceMixer.
This code path is not used anymore.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/3015553002
Cr-Commit-Position: refs/heads/master@{#19929}
2017-09-22 13:48:10 +00:00
cb728ea83a Fix Gn Untracked headers in webrtc/modules/video_coding.
Fixed following headers in this CL
===================================

src/webrtc/modules/video_coding/sequence_number_util.h
src/webrtc/modules/video_coding/codecs/interface/common_constants.h
src/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h

src/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
src/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
src/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h

src/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h

src/webrtc/modules/video_coding/test/test_util.h
src/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
src/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
src/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h

Remaining:
===========
src/webrtc/modules/video_coding/include/video_codec_interface.h
src/webrtc/modules/video_coding/include/video_error_codes.h

BUG=webrtc:7620

Review-Url: https://codereview.webrtc.org/3012323002
Cr-Commit-Position: refs/heads/master@{#19886}
2017-09-18 10:08:08 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00