Commit graph

37 commits

Author SHA1 Message Date
Mirko Bonadei
9f9e20a3dc Fix errorprone issues preventing Chromium Roll.
Some ErrorProne warnings have been enabled by [1], that broke the
Chromium Roll into WebRTC, this CL should have taken care of all the
problems.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1935889

Bug: None
Change-Id: I2670e948c320984a122fdb774b891c98e05f582e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160862
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29933}
2019-11-27 12:52:48 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00
Jonas Olsson
a4d873786f Format almost everything.
This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}
2019-07-08 13:45:15 +00:00
Niels Möller
5a96a0e516 Reland "Delete deprecated rtc::Thread default constructor"
This is a reland of fdd6d3e46e

Original change's description:
> Delete deprecated rtc::Thread default constructor
>
> Bug: None
> Change-Id: Ic0e2e94b174a49e5d20ebdea90568473e1b71d62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134640
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27958}

Tbr: tommi@webrtc.org
Bug: None
Change-Id: I9e4b1d06e79670b4efb9c9517d909a0562485e12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137504
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27982}
2019-05-20 07:49:30 +00:00
Erik Språng
6e70f18fbd Revert "Delete deprecated rtc::Thread default constructor"
This reverts commit fdd6d3e46e.

Reason for revert: Seems to be be breaking downstream project.

Original change's description:
> Delete deprecated rtc::Thread default constructor
> 
> Bug: None
> Change-Id: Ic0e2e94b174a49e5d20ebdea90568473e1b71d62
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134640
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27958}

TBR=nisse@webrtc.org,tommi@webrtc.org

Change-Id: Id8015805c28c08e5e5d97c0cdef3c17d6d281fb5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137042
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27959}
2019-05-16 10:27:25 +00:00
Niels Möller
fdd6d3e46e Delete deprecated rtc::Thread default constructor
Bug: None
Change-Id: Ic0e2e94b174a49e5d20ebdea90568473e1b71d62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134640
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27958}
2019-05-16 08:44:21 +00:00
Ilya Nikolaevskiy
a8507e359b Remove deprecated interface in I420BufferInterface::GetI420
Bug: none
Change-Id: I55895a360308fd0be79099f2466a7487ef10ce47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134463
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27841}
2019-05-03 12:16:34 +00:00
Mirko Bonadei
0813fd504d Fix examples/unityplugin docs.
TBR=qiangchen@chromium.org

No-Try: True
Bug: webrtc:10544
Change-Id: I7e691b5420215a920214147747139c4ffaceebfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132705
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27582}
2019-04-12 09:04:03 +00:00
Magnus Jedvert
167316b833 Remove proxy layer from AndroidVideoTrackSource
This layer is not needed since the methods are thread safe, and the
classes those method touches (VideoBroadcaster, cricket::VideoAdapter)
are thread safe.

Bug: webrtc:10247
Change-Id: Id4e309de4ac1b9669052aaa60d3bd1ed965aaa29
Reviewed-on: https://webrtc-review.googlesource.com/c/120801
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26543}
2019-02-05 10:50:40 +00:00
Steve Anton
f380284035 (7) Rename files to snake_case: remove forwarding headers
Bug: webrtc:10159
Change-Id: I2ba899e0283b953538c7941c8790213e36d7c70e
Reviewed-on: https://webrtc-review.googlesource.com/c/118561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26417}
2019-01-26 00:33:46 +00:00
Anders Carlsson
45340ca824 Remove legacy video codec factories.
Removes the deprecated video codec factories and the related flag and
helper classes.

Bug: webrtc:7925
Change-Id: I0a6d1666ece9ad074fefc79b626ba241765e1b98
Reviewed-on: https://webrtc-review.googlesource.com/c/113940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26245}
2019-01-14 14:56:40 +00:00
Steve Anton
aec15aa810 (5) Rename files to snake_case: install forwarding headers
Mechanically generated with this command:

tools_webrtc/do-renames.sh install all-renames.txt && git cl format

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: Ic8e99f71f2da62e5c99863c6d24a8cfe311466cd
Reviewed-on: https://webrtc-review.googlesource.com/c/115682
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26227}
2019-01-11 17:13:36 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Steve Anton
1c05765831 (3) Rename files to snake_case: move the files
Mechanically generated with this command:

tools_webrtc/do-rename.sh move all-renames.txt

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I8b05b6eab9b9d18b29c2199bbea239e9add1e690
Reviewed-on: https://webrtc-review.googlesource.com/c/115481
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26225}
2019-01-11 17:05:20 +00:00
Niels Möller
0a59535ce7 Update unityplugin to use VcmCapturer.
Replaces use of WebRtcVideoDeviceCapturerFactory.

Bug: webrtc:6353
Change-Id: I3c1626af46cb56817190739a39842c4c5a51560d
Reviewed-on: https://webrtc-review.googlesource.com/c/115960
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26117}
2019-01-03 10:06:17 +00:00
Artem Titarenko
69540f4419 Use android Nullable instead of javax Nullable
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.

Original comment from upstream change:

> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.

Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
2018-12-10 15:03:58 +00:00
Mirko Bonadei
2ff3f49700 Move webrtc::CreatePeerConnectionFactory definition next to decl.
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.

In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory

Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
2018-11-22 09:07:51 +00:00
Magnus Jedvert
9514071500 Android: Support externally aligned timestamps
This support is needed if there is a big delay between the creation of
frames and the time they are delivered to the WebRTC C++ layer in
AndroidVideoTrackSource. This is the case if e.g. some heavy video
processing is applied to the frames that takes a couple of hundred
milliseconds. Currently, timestamps coming from Android video sources
are aligned to rtc::TimeMicros() once they reach the WebRTC C++ layer in
AndroidVideoTrackSource. At this point, we "forget" any latency that
might occur before this point, and audio/video sync consequently
suffers.

Bug: webrtc:9991
Change-Id: I7b1aaca9a60a978b9195dd5e5eed4779a0055607
Reviewed-on: https://webrtc-review.googlesource.com/c/110783
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25654}
2018-11-15 11:41:06 +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
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
Niels Möller
2d02e085de Delete deprecated CreateAudioSource method, with constraints.
Bug: webrtc:9239
Change-Id: I5025b7fd103247e0426ceabedc1216a4f0f0ab34
Reviewed-on: https://webrtc-review.googlesource.com/76560
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23501}
2018-06-04 08:19:30 +00:00
Harald Alvestrand
73771a893f Prepare to remove old OnFailure implementations
This removes usage of the old OnFailure methods on CreateSessionDescriptionObserver
and SetSessionDescriptionObserver, so that WebRTC will continue to compile
once all the default implementations are removed.

Bug: chromium:589455
Change-Id: Id67295b3ad0c30d24d79589c2041acdd507a19f3
Reviewed-on: https://webrtc-review.googlesource.com/78480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23427}
2018-05-29 10:34:14 +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
Magnus Jedvert
1a759c6354 Android: Only use Java VideoFrames internally
This CL removes internal support for anything else than Android frames
that are wrapped Java VideoFrames. This allows for a big internal
cleanup and we can remove the internal class AndroidTextureBuffer and
all logic related to that. Also, the C++ AndroidVideoTrackSource no
longer needs to hold on to a C++ SurfaceTextureHelper and we can
remove all JNI code related to SurfaceTextureHelper. Also, when these
methods are removed, it's possible to let VideoSource implement the
CapturerObserver interface directly and there is no longer any need for
AndroidVideoTrackSourceObserver. Clients can then initialize
VideoCapturers themselves outside the PeerConnectionFactory, and a new
method is added in the PeerConnectionFactory to allow clients to create
standalone VideoSources that can be connected to a VideoCapturer outside
the factory.

Bug: webrtc:9181
Change-Id: Ie292ea9214f382d44dce9120725c62602a646ed8
Reviewed-on: https://webrtc-review.googlesource.com/71666
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23004}
2018-04-24 13:51:11 +00:00
Sami Kalliomäki
dc52651911 Annotate rest of WebRTC with @Nullable.
Bug: webrtc:8881
Change-Id: Ic199efa73a0b3b9437df1e8fe5a1814a70380993
Reviewed-on: https://webrtc-review.googlesource.com/64884
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22639}
2018-03-28 08:30:06 +00:00
Seth Hampson
13b8bad235 Final name changing of MediaStreamInterface.label() to id().
Downstreams have been updated, and this now updates all uses of label()
to id() within WebRTC code. This change also makes id() pure virtual and
removes label().

Bug: webrtc:8977
Change-Id: Ib045ea4fabba6f14447c64875c7aeba87dc2be24
Reviewed-on: https://webrtc-review.googlesource.com/60382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22431}
2018-03-14 20:30:52 +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
Seth Hampson
513449eab9 Changes name of RtpTransceiverInit's stream_labels to stream_ids.
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.

Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
2018-03-06 23:42:01 +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
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
Qiang Chen
43fb912318 Reland: Add Alpha Channel Support For WebRTC Unity Plugin
This CL make webrtc unity plugin compatible with alpha channel support.

TBR=gyzhou@chromium.org,magjed@webrtc.org

Bug: webrtc:8645
Change-Id: Ic1c11f8c82f8244b84b8ab67c623ad2002b940e8
Reviewed-on: https://webrtc-review.googlesource.com/35421
Reviewed-by: George Zhou <gyzhou@chromium.org>
Reviewed-by: Qiang Chen <qiangchen@chromium.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21398}
2017-12-20 21:56:17 +00:00
Lu Liu
ec8410796a Revert "Add Alpha Channel Support For WebRTC Unity Plugin"
This reverts commit 7ed2af5b46.

Reason for revert: breaking buildbot

Original change's description:
> Add Alpha Channel Support For WebRTC Unity Plugin
> 
> This CL make webrtc unity plugin compatible with alpha channel support.
> 
> Bug: webrtc:8645
> Change-Id: I3250aede47b31c4685e57d11fb2b2e86b824f9c4
> Reviewed-on: https://webrtc-review.googlesource.com/32325
> Commit-Queue: Qiang Chen <qiangchen@chromium.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: George Zhou <gyzhou@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#21394}

TBR=magjed@webrtc.org,gyzhou@chromium.org,qiangchen@chromium.org

Change-Id: I6994d7e87170f97216886a747548a988ca71b7d0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8645
Reviewed-on: https://webrtc-review.googlesource.com/35420
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21396}
2017-12-20 18:37:27 +00:00
Qiang Chen
7ed2af5b46 Add Alpha Channel Support For WebRTC Unity Plugin
This CL make webrtc unity plugin compatible with alpha channel support.

Bug: webrtc:8645
Change-Id: I3250aede47b31c4685e57d11fb2b2e86b824f9c4
Reviewed-on: https://webrtc-review.googlesource.com/32325
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21394}
2017-12-20 18:23:16 +00:00
Qiang Chen
51e2046dbc Bug Fix: WebRTC Unity Plugin Audio One Way
When audio_only is on for the webrtc unity plugin, there is a bug that
the audio from hologram cannot be heard at the remote side.

Actually we found the audio is transmitted to the remote side, but the
remote side wants video data also to playout everything. So without
video data, the remote side will drop all the audio data.

Thus, on the hologram (using webrtc unity plugin) side, we should not
hook up a dummy camera, but instead we should use media constraint to
request the remote side to send video data.

This CL fixes the bug.

Bug: webrtc:8555
Change-Id: I21ddda65185b645088aa4ac15f47b3f8ffad1873
Reviewed-on: https://webrtc-review.googlesource.com/24680
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Reviewed-by: George Zhou <gyzhou@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21094}
2017-12-05 20:36:12 +00:00
Mirko Bonadei
675513b96a Stop using LOG macros in favor of RTC_ prefixed macros.
This CL has been generated with the following script:

for m in PLOG \
  LOG_TAG \
  LOG_GLEM \
  LOG_GLE_EX \
  LOG_GLE \
  LAST_SYSTEM_ERROR \
  LOG_ERRNO_EX \
  LOG_ERRNO \
  LOG_ERR_EX \
  LOG_ERR \
  LOG_V \
  LOG_F \
  LOG_T_F \
  LOG_E \
  LOG_T \
  LOG_CHECK_LEVEL_V \
  LOG_CHECK_LEVEL \
  LOG
do
  git grep -l $m | xargs sed -i "s,\b$m\b,RTC_$m,g"
done
git checkout rtc_base/logging.h
git cl format

Bug: webrtc:8452
Change-Id: I1a53ef3e0a5ef6e244e62b2e012b864914784600
Reviewed-on: https://webrtc-review.googlesource.com/21325
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20617}
2017-11-09 11:56:32 +00:00
Mirko Bonadei
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
Mirko Bonadei
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