This reverts commit 69241a93fb.
Reason for revert: Breaks WebRTC roll into Chromium.
Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
> break a circular dependency (is has been extracted from
> //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
> break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}
TBR=mbonadei@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.
This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).
The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
break a circular dependency (is has been extracted from
//rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
break another circular dependency.
Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
- This CL also affects some return type handling in Android Voip demo
app due to changes in return type handling.
Bug: webrtc:12193
Change-Id: Id76faf7c871476ed1f2d08fb587211ae234ae8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32821}
`create_srcjar = false` was needed during the transition to moving
R.java generation to android_library targets. Now this variable is
unused (the variable is asserted to be false), clean up all references.
Bug: chromium:1073476
Change-Id: I4c09ea05ded27ea2360392aacbce036bc1a2f928
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185160
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mohamed Heikal <mheikal@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32178}
R.java file creation responsibilities will be moved to android_library
and android_apk targets and creating R.java files in the
android_resources targets is now deprecated. This cl migrates webrtc
targets to the new way.
Bug: chromium:1073476
Change-Id: I0a2fa759d3ff1d8e201e5719c9238701a58171e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183060
Commit-Queue: Mohamed Heikal <mheikal@chromium.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32026}
Moved asynchronicity from Java to C++.
Bug: webrtc:11723
Change-Id: I985693dc7d4312b6072314088716167b9cdd9999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180774
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31958}
The app showcased the ability to send real-time voice data between two endpoints using the VoIP API.
Users can also configure session parameters such as the endpoint information and codec used.
Bug: webrtc:11723
Change-Id: I682f4aa743b707759536bce59e598789a77b7ec6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178467
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31775}
