This reverts commit 264547d084.
Refactor AsyncTcpSocket(s) to use rtc::ReceivedPackets
Patchset 1 contains original cl.
Newer patchsets contains fix of the problem from pathset 1.
Bug: webrtc:15368, webrtc:11943
Change-Id: Ib8c4c06daf502a5dec8c31beea78eacac8c3c644
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328820
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Cr-Commit-Position: refs/heads/main@{#41255}
This reverts commit 211daadb66.
Reason for revert: AsyncStunTCPSocket::ProcessInput , Bug introduced, not reading length of each stun message in a tcp fetch
Original change's description:
> Refactor AsyncTcpSocket(s) to use rtc::ReceivedPackets
>
> Instead of using raw pointers.
>
> Bug: webrtc:15368, webrtc:11943
> Change-Id: Id28a0a4fc3d00680e972bd95e0c60344c7886892
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328500
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41237}
Bug: webrtc:15368, webrtc:11943
Change-Id: Id15261579a61dd200e7c3b1a013877575b87db2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41245}
Instead of using raw pointers.
Bug: webrtc:15368, webrtc:11943
Change-Id: Id28a0a4fc3d00680e972bd95e0c60344c7886892
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328500
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41237}
This is done instead of directly using AsyncPacketSocket::SignalReceived.
Bug: webrtc:15368, webrtc:11943
Change-Id: I5671e66b270355188b1252138eced8e6c78ba7ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327521
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41180}
This time, hit the BUILD files too (where possible).
Bug: webrtc:11943
Change-Id: Ic8f2d77e1ba66f740efe0ef73b1ea6051356dedc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40654}
This removes use of the SignalClose sigslot. This CL includes thread
checks for the callback list and updates some call sites to unsubscribe
from events before deletion or detaching from a socket instance.
Bug: webrtc:11943
Change-Id: Ib66d39aa5cc795b750c9e3eaa85ed6af8b55b2b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258561
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36540}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
The AsyncListenSocket::SetOption method then gets unused, and can be
deleted.
Bug: webrtc:13065
Change-Id: Idcf70a75b96036290fdceff6e0f96a8d5617f87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35302}
This is a reland of b141c162ee
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
Bug: webrtc:13065
Change-Id: I88bebdd80ebe6bcf6ac635023924d79fbfb76813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35260}
This reverts commit b141c162ee.
Reason for revert: Breaking WebRTC rolls. See https://ci.chromium.org/ui/b/8832847811929676465 for an example failed build.
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13065
Change-Id: Id5d5b35cb21704ca4e3006caf1636906df062609
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235824
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35249}
The AsyncTCPSocket is an AsyncPacketSocket which means it
emulates UDP-like (packet) semantics via a TCP stream. When
sending, if the entire packet could not be written then the
packet socket should indicate it wrote the whole thing and
flush out the remaining later when the socket is available.
The WriteEvent signal was already wired up but was not getting
fired (at least with the virtual sockets) since it would not
call Send() enough on the underlying socket to get an
EWOULDBLOCK that would register the async event.
This changes AsyncTCPSocket to repeatedly call Send() on the
underlying socket until the entire packet has been written
or EWOULDBLOCK was returned.
Bug: webrtc:6655
Change-Id: I41e81e0c106c9b3e712a8a0f792d28745d93f2d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168083
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30449}
