And delete corresponding dependencies on :webrtc_common. After this
change, common_types.h is included directly only from code in the
following directories:
api/
api/video/
api/video_codecs/
common_video/libyuv/include/
media/base/
modules/remote_bitrate_estimator/
modules/rtp_rtcp/source/
modules/video_coding/codecs/vp9/
There remains plenty of indirect dependencies on the types declared in
common_types.h, but the fewer direct dependencies should make it
easier to find the proper place for each type.
Bug: webrtc:5876
Change-Id: I93e8f214025ecb613c19fdec2015bd3f96c59aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27376}
This is a reland of 171df93262
Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}
Tbr: danilchap@webrtc.org
Bug: webrtc:6883
Change-Id: I30771b86bbe50de609353e23e80dc532dc884ad4
Reviewed-on: https://webrtc-review.googlesource.com/c/119661
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26394}
This reverts commit 171df93262.
Reason for revert: Breaks downstream project
Original change's description:
> Delete RtpUtility::Payload, and refactor RTPSender to not use it
>
> Replaced by a payload type --> video codec map in RTPSenderVideo,
> where it is used to select the right packetizer.
>
> Bug: webrtc:6883
> Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
> Reviewed-on: https://webrtc-review.googlesource.com/c/119263
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26380}
TBR=danilchap@webrtc.org,brandtr@webrtc.org,nisse@webrtc.org
Change-Id: I76489c29541827aaba72515a76db54bdb7495e28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6883
Reviewed-on: https://webrtc-review.googlesource.com/c/119640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26385}
Replaced by a payload type --> video codec map in RTPSenderVideo,
where it is used to select the right packetizer.
Bug: webrtc:6883
Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f
Reviewed-on: https://webrtc-review.googlesource.com/c/119263
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26380}
The replacements are absl::EqualsIgnoreCase and
absl::StartsWithIgnoreCase. Also delete the alias
RtpUtility::StringCompare.
Bug: webrtc:6424
Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f
Reviewed-on: https://webrtc-review.googlesource.com/c/109006
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25481}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
Remove PayloadUnion's public member variables, so that the outside
world has to go through the accessors.
This is good code hygiene in general. For example, it makes it
possible to make the audio and video states Optional, so that exactly
one of them can be live at any one time.
BUG=webrtc:8159
Change-Id: Ie617b9038f961b329bd67b45478ff33d97148447
Reviewed-on: https://webrtc-review.googlesource.com/4428
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20064}
I need to replace the audio part of PayloadUnion with SdpAudioFormat,
but that's a non-trivially-deletable class and those just don't work
well in unions, especially unions that don't have a discriminator that
says which member is currently active.
This CL converts the union to a class, adds a discriminator, and
provides accessor functions. CL #2 in the series will change all
outsiders to use the accessors instead of the public member variables
directly, and CL #3 will remove the public member variables. (It needs
to be done in separate steps like this because PayloadUnion is
unfortunately part of the API, and just changing it all in one go
would break users.)
BUG=webrtc:8159
Change-Id: I38c44bbb21a2d38600cff59bf37d8d47dfdbce21
Reviewed-on: https://webrtc-review.googlesource.com/4340
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20025}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
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}
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}
