We get this automatically from the //build checkout now
Bug: chromium:1432399
Change-Id: I223d7c5448244ed62821207068f979555617da57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318560
Auto-Submit: Chong Gu <chonggu@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/main@{#40686}
This needs to be rolled back as soon as the deprecated declarations
diagnostic errors get fixed.
Bug: b/288827308
Change-Id: I9584e0f156b0bd0ba40809d43e1edd7ba9ff5674
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/310300
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40342}
Chromium has done this in 2021 (https://crrev.com/c/2405530).
This will allow the new BoringSSL to roll into WebRTC (since now
BoringSSL requires the NDK level to be >= 18).
NO_TRY=True
Bug: None
Change-Id: I4b722aef56cb1e189f4df6de64a87ab1e5b620a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297362
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39543}
This will unblock the roll
Bug: b/226557854
Change-Id: I7f5c5b6385e91bf55c5c39fa63ebb8b55c639734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256681
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#36319}
This should avoid the situation where WebRTC's GN check is green and
Chromium (which turns it ON for //third_party/webrtc) fails.
Bug: webrtc:12614
Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33900}
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.
Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
To unblock roll of newer build files for libaom that are using
differen build flags to include/exclude libaom
Bug: webrtc:11404
Change-Id: If06b63e0835e65113617efa29f34ba6bb309c16d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170630
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30810}
This is a canary CL to check if using c++14 feature breaks any webrtc user.
Bug: webrtc:10945
Change-Id: Iabaf8c06414c1ac960791bcb7cc46f5f5a5e1f14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151600
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29119}
After https://webrtc-review.googlesource.com/c/src/+/93733, WebRTC
should be able to use absl pretty printers again.
Bug: None
Change-Id: I1ea63707e59ad502e40df84ca0abb03054d25106
Reviewed-on: https://webrtc-review.googlesource.com/93741
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24272}
This is a copy of b8736d858c
We probably don't need most of this but the build loudly screams without all those files, so whatever...
Bug: webrtc:9118
Change-Id: I5df54b4857eee9a2bcf8dea05e36f009570a0e21
Reviewed-on: https://webrtc-review.googlesource.com/89586
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24043}
Starting from [1], gtest can pretty print absl types. In order to
enable the feature WebRTC has to set gtest_enable_absl_printers to true
in the .gn file.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1027711
Bug: None
Change-Id: I74eb9a48c361f1523dd8d45510297e101a4d14cd
Reviewed-on: https://webrtc-review.googlesource.com/85345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23781}
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}
Libyuv is now 'gn check' compatible and the fixed version has been
rolled into chromium (r1697).
Bug: webrtc:8850
Change-Id: Iaaeae229571fd02045322c4f8addadd75f889bdb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/50180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21966}
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.
NOPRESUBMIT=true
Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
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}
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.
Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.
A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.
Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
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}
Currently, WebRTC is setting this config via mac_sdk_min_build_override. The old
mechanism is deprecated, but cannot be removed until chromium is updated to no
longer require mac_sdk_min_build_override.
BUG=chromium:740693
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2979603002
Cr-Commit-Position: refs/heads/master@{#19006}
