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8 Commits

Author SHA1 Message Date
Honghai Zhang
f8998cf8c4 Add a turn port prune policy to keep the first ready turn port. 
Bug: webrtc:11026
Change-Id: I6222e9613ee4ce2dcfbb717e2430ea833c0dc373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155542
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29470}
 2019-10-14 19:08:23 +00:00
Bjorn A Mellem
bc3eebc722 Reland "Reland "Refactor SCTP data channels to use DataChannelTransportInterface."" 
This is a reland of 487f9a17e426fd14bb06b13e861071b3f15d119b

Original change's description:
> Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
> 
> Also clears SctpTransport before deleting JsepTransport.
> 
> SctpTransport is ref-counted, but the underlying transport is deleted when
> JsepTransport clears the rtp_dtls_transport.  This results in crashes when
> usrsctp attempts to send outgoing packets through a dangling pointer to the
> underlying transport.
> 
> Clearing SctpTransport before DtlsTransport removes the pointer to the
> underlying transport before it becomes invalid.
> 
> This fixes a crash in chromium's web platform tests (see
> https://chromium-review.googlesource.com/c/chromium/src/+/1776711).
> 
> Original change's description:
> > Refactor SCTP data channels to use DataChannelTransportInterface.
> >
> > This change moves SctpTransport to be owned by JsepTransport, which now
> > holds a DataChannelTransport implementation for SCTP when it is used for
> > data channels.
> >
> > This simplifies negotiation and fallback to SCTP.  Negotiation can now
> > use a composite DataChannelTransport, just as negotiation for RTP uses a
> > composite RTP transport.
> >
> > PeerConnection also has one fewer way it needs to manage data channels.
> > It now handles SCTP and datagram- or media-transport-based data channels
> > the same way.
> >
> > There are a few leaky abstractions left.  For example, PeerConnection
> > calls Start() on the SctpTransport at a particular point in negotiation,
> > but does not need to call this for other transports.  Similarly, PC
> > exposes an interface to the SCTP transport directly to the user; there
> > is no equivalent for other transports.
> 
> Bug: webrtc:9719
> Change-Id: I64e94b88afb119fdbf5f22750f88c8a084d53937
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29120}

Bug: webrtc:9719
Change-Id: I28481a3de64a3506bc57748106383eeba4ef205c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152740
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29290}
 2019-09-24 17:10:52 +00:00
Sebastian Jansson
1b83a9e400 Only handle each RTCP once. 
Previously, each RTCP packet was handled several times in a row, once
per m-section. This caused various weirdness and log warning spam, in
particular when using unified plan.

The cause was that the packets were wired trough each BaseChannel
instance up to the Call class. With this fix, the RTCP packets are wired
once per RtpTransportInternal via the common peer connection class.

Bug: chromium:1002875
Change-Id: I41c4eb3b68e215ebe0f2c6fb93ae0ee73335b89a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152668
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29226}
 2019-09-18 16:54:39 +00:00
Sebastian Jansson
86314cfb5d Cleaning up C++14 move into lambda TODOs. 
Bug: webrtc:10945
Change-Id: I4d2f358b0e33b37e4b4f7bfcf3f6cd55e8d46bf9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153240
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29212}
 2019-09-17 19:18:26 +00:00
Mirko Bonadei
317a1f09ed Use std::make_unique instead of absl::make_unique. 
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
 2019-09-17 15:47:29 +00:00
Sebastian Jansson
47287d546d Reland "Adds peer scenario connection interface." 
This is a reland of d181ee798da57ce5b955f09e8dcb755fba70b51b

Original change's description:
> Adds peer scenario connection interface.
>
> This allows implementing custom clients for test in peer connection
> scenario tests. For example server side behavior.
>
> Bug: webrtc:10839
> Change-Id: I5627b7a4d967d401f31d2e9a8f861d0849eb0184
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151907
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29125}

TBR=perkj@webrtc.org

Bug: webrtc:10839
Change-Id: I5e0857dc7647587eab2a9b61965f627bf310b88c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152481
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29147}
 2019-09-11 07:10:52 +00:00
Qingsi Wang
467073a0c1 Revert "Adds peer scenario connection interface." 
This reverts commit d181ee798da57ce5b955f09e8dcb755fba70b51b.

Reason for revert: the dependent API changing cl is reverted

Original change's description:
> Adds peer scenario connection interface.
> 
> This allows implementing custom clients for test in peer connection
> scenario tests. For example server side behavior.
> 
> Bug: webrtc:10839
> Change-Id: I5627b7a4d967d401f31d2e9a8f861d0849eb0184
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151907
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29125}

TBR=srte@webrtc.org,perkj@webrtc.org

Change-Id: I8bc5dd4fdc9d72288baa74ff94c1ad8b3e7772a6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152423
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29142}
 2019-09-10 18:19:48 +00:00
Sebastian Jansson
d181ee798d Adds peer scenario connection interface. 
This allows implementing custom clients for test in peer connection
scenario tests. For example server side behavior.

Bug: webrtc:10839
Change-Id: I5627b7a4d967d401f31d2e9a8f861d0849eb0184
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151907
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29125}
 2019-09-10 08:10:37 +00:00