In its stead, there's now a SetEventCallback() method.
Bug: webrtc:11943
Change-Id: If936d6e1e23e8a584f06feb123ecf2d450ea4145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319040
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42187}
This is a step towards removing StreamInterface::SignalEvent.
Downstream dependency will need to be updated to call FireEvent()
before further changes can land in webrtc.
Bug: webrtc:11943
Change-Id: Ia7d3f1c43fda52b7cf5bfa082aef3f462553cd67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347884
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42143}
This cleans up the last vestiges of the old interface for rtc::StreamInterface
and will cause builds to refer to the old functions to fail.
Bug: webrtc:14632
Change-Id: I569b16677754d7f9e08449e273672a59a86e6498
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283844
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38784}
Updates the code to use the new interfaces
Bug: webrtc:14632
Change-Id: I33b2a25b5968de0251e3cbc84076afc013ecef6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282680
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38601}
A few usages in ssl_stream_adapter_unittests are converted to make
sure the aliases are usable.
Next steps are:
- Change all usages inside WebRTC to the new form
- Deprecate the old API
- Remove the old API
Pipewire failures believed to be unrelated, so No-try.
No-try: true
Bug: webrtc:14632
Change-Id: I618551e61a05d53e524e97483d3c7cef59b88a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282221
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38577}
This includes relying on related types such as MessageData and
PostEvent functionality inside the StreamInterface itself.
This affects mostly tests but OpenSSLStreamAdapter
requires special attention.
Bug: webrtc:11988
Change-Id: Ib5c895f1bdf77bb49e3162bd49718f8a98812d91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185505
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32290}
As documented in webrtc:11908 this cleanup is fairly invasive and
when a part of a frequently executed code path, can be quite costly
in terms of performance overhead. This is currently the case with
synchronous calls between threads (Thread) as well with our proxy
api classes.
With this CL, all code in WebRTC should now either be using MessageHandlerAutoCleanup
or calling MessageHandler(false) explicitly.
Next steps will be to update external code to either depend on the
AutoCleanup variant, or call MessageHandler(false).
Changing the proxy classes to use TaskQueue set of concepts instead of
MessageHandler. This avoids the perf overhead related to the cleanup
above as well as incompatibility with the thread policy checks in
Thread that some current external users of the proxies would otherwise
run into (if we were to use Thread::Send() for synchronous call).
Following this we'll move the cleanup step into the AutoCleanup class
and an RTC_DCHECK that all calls to the MessageHandler are setting
the flag to false, before eventually removing the flag and make
MessageHandler pure virtual.
Bug: webrtc:11908
Change-Id: Idf4ff9bcc8438cb8c583777e282005e0bc511c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183442
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32049}
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.
Bug: webrtc:9883
Change-Id: I3cb857cc707d5e897759366d1478cc1ec19bce9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165344
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30180}
Used only by test code and by pseudo_tcp.
Bug: webrtc:6424
Change-Id: I28903e74f7b69cbdd8c368f4444c8a233eb76868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128868
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27341}
Keep methods on subclasses where they are used: FifoBuffer and
MemoryStream. Also FileStream gets to keep SetPosition, because it's
used by a downstream subclass.
Bug: webrtc:6424
Change-Id: If2a00855aba7c2c9dc0909cda7c8a8ef00e0b9af
Reviewed-on: https://webrtc-review.googlesource.com/c/116487
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26237}
Also delete the base class MemoryStreamBase.
Bug: webrtc:6424
Change-Id: I7eec40ecff17c7c65fdf55882bccb2abaf1d5d84
Reviewed-on: https://webrtc-review.googlesource.com/c/108622
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25494}
Moved methods: GetReadData, ConsumeReadData, GetWriteBuffer,
ConsumeWriteBuffer, GetWriteRemaining.
These methods represented an optional interface for reading and
writing streams, intended to optimize certain use cases. However,
it was implemented only in the FifoBuffer subclass, and the few
users of that class all have a concrete FifoBuffer, and hence
don't need the methods on the abstract StreamInterface.
Bug: webrtc:6424
Change-Id: I6de74d1a9205fcb7037ad84e24679d4a27c1d219
Reviewed-on: https://webrtc-review.googlesource.com/c/108621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25446}
Refactor only remaining user, IsDefaultRoute (helper function
called from BasicNetworkManager::IsIgnoredNetwork) to use a
FILE* and fgets instead.
Bug: webrtc:6424
Change-Id: I57652f664b9a6965c19575c1b5d7f7de24f2ed44
Reviewed-on: https://webrtc-review.googlesource.com/c/108089
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25433}
Deleted methods HttpData::setContent and
HttpData::setDocumentAndLength, as well as the
StreamInterface::GetAvailable method which becomes unused.
Bug: webrtc:6424
Change-Id: I6f360b68327d5964b2a18a9c4055255d774f6cbc
Reviewed-on: https://webrtc-review.googlesource.com/101180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24793}
Deleted features include HttpBase::GetDocumentStream(), and support for
other operations than GET.
Bug: webrtc:6424
Change-Id: Ib16537cd1db87de53150f8e9e30dd89778a20c2e
Reviewed-on: https://webrtc-review.googlesource.com/84140
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24102}
Also move StringStream to the only test using it.
Bug: webrtc:6424
Change-Id: Iad79c7becaa2764ac954c18711eaae4faf46ae72
Reviewed-on: https://webrtc-review.googlesource.com/84320
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23721}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Deletes the ALIGNP and RTC_ALIGNED_P macros from basictypes.h.
ALIGNP was used by MemoryStream, supposedly to make it more efficient.
If it really provided an efficiency improvement is unclear, and in any
case, MemoryStream is used for tests only, and doesn't need high
performance.
Bug: webrtc:6853
Change-Id: If835e881e3857dcc22c7a544491b92829a81d1b3
Reviewed-on: https://webrtc-review.googlesource.com/78021
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23350}
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}
