This is a reland of 1a2cc0acba6a66f89249455d8e5775849b56f755
Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
>
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
>
> Original change's description:
> > Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
> >
> > We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> > to report the metrics in pc/ and p2p/ that are currently been reported
> > using MetricsObserverInterface.
> >
> > TBR=tommi@webrtc.org
> >
> > Bug: webrtc:9409
> > Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> > Reviewed-on: https://webrtc-review.googlesource.com/83782
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23914}
>
> TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}
TBR=steveanton@webrtc.org,tommi@webrtc.org
Bug: webrtc:9409
Change-Id: Ib55f0b6c9bcb9d9585924a4dfac5cf643ff4d76b
Reviewed-on: https://webrtc-review.googlesource.com/88343
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23957}
This is required to make the test non-flaky.
Bug: webrtc:9494
Change-Id: Iae26028233fa4d990f082cbc1b023253e783ccc8
Reviewed-on: https://webrtc-review.googlesource.com/87438
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23951}
This is an reland of 6f5b0f920af08d66e6b77ee4f91ade5797145368
Relanded after speculative revert without any changes.
TBR=ilnik@webrtc.org
Original change's description:
> Remove rtc::Optional alias and api:optional target
>
> Update left-overs where old target still was used.
>
> Bug: webrtc:9078
> Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
> Reviewed-on: https://webrtc-review.googlesource.com/84740
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23913}
Bug: webrtc:9078
Change-Id: Ia33c6438253c6ec49f45d938e8a3607b51c418be
Reviewed-on: https://webrtc-review.googlesource.com/88160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23941}
GetStats() is currently the only way to get smoothed audio levels. It's
expected to be called quite frequently. We use 100ms intervals, so a
call to GetStats() for any reason other than audio levels has a 50%
chance of triggering this log line. This makes it too noisy for LS_INFO.
The log line was added recently
(https://webrtc-review.googlesource.com/c/src/+/82260) and doesn't seem
very useful for diagnostic purposes, so remove it entirely.
Bug: webrtc:9519
Change-Id: I15700085c60b9929a4df2e2327012a4f16b505b6
Reviewed-on: https://webrtc-review.googlesource.com/88003
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23939}
This reverts commit 1a2cc0acba6a66f89249455d8e5775849b56f755.
Reason for revert: It breaks internal Android debug build. Need further investigation.
Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
>
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
>
> Original change's description:
> > Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
> >
> > We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> > to report the metrics in pc/ and p2p/ that are currently been reported
> > using MetricsObserverInterface.
> >
> > TBR=tommi@webrtc.org
> >
> > Bug: webrtc:9409
> > Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> > Reviewed-on: https://webrtc-review.googlesource.com/83782
> > Commit-Queue: Qingsi Wang <qingsi@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23914}
>
> TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,tommi@webrtc.org,hta@webrtc.org,qingsi@google.com,qingsi@webrtc.org
Change-Id: I4a75fc7f52bfd0780526537a5a9a016fb9c20d6a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88320
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23938}
This is the first step to removing streams from third_party/webrtc.
RtpReceiverInterface::streams() will have to be removed separately.
See https://crbug.com/webrtc/9480 for more information.
Bug: webrtc:9480
Change-Id: I6f9e6ddcda5e2245cc601d2cc6205b7b363f73ef
Reviewed-on: https://webrtc-review.googlesource.com/86840
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23929}
This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}
TBR=steveanton@webrtc.org,hta@webrtc.org,tommi@webrtc.org
Bug: webrtc:9409
Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
Reviewed-on: https://webrtc-review.googlesource.com/88060
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23919}
This reverts commit 870bca1f418a1abf445169a638a61f9a649d557f.
Reason for revert: it breaks internal tests and builds
Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hta@webrtc.org,tommi@webrtc.org
Change-Id: I1afd92d44f3b8cf3ae9aa6e6daa9a3a272e8097f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23916}
We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
to report the metrics in pc/ and p2p/ that are currently been reported
using MetricsObserverInterface.
TBR=tommi@webrtc.org
Bug: webrtc:9409
Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
Reviewed-on: https://webrtc-review.googlesource.com/83782
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23914}
Update left-overs where old target still was used.
Bug: webrtc:9078
Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206
Reviewed-on: https://webrtc-review.googlesource.com/84740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23913}
Generated using gmock_gen.py with some editing.
This mock doesn't seem to be used by unittest in webrtc, but we need to use it in downstream unittests.
Bug: None
Change-Id: Ia7904ffdd22f3d16fe5fd515fa68833817b44481
Reviewed-on: https://webrtc-review.googlesource.com/85780
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23900}
With deadbeef removed from these OWNERS files, Steve is the only OWNER
on our team. I'm adding myself, because I have worked in these
directories and it makes sense to be able to distribute the code
reviews.
NOTRY=True
Bug: None
Change-Id: I48e88a07ee42254d937391f500f273656853d98b
Reviewed-on: https://webrtc-review.googlesource.com/86980
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23826}
ADAPTER_TYPE_ANY can be used to set the network ignore mask if an
application does not want candidates from the any address ports, the
underlying network interface types of which are not determined in
gathering. The ADAPTER_TYPE_ANY is also given the maximum network cost
so that when there are candidates from explicit network interfaces,
these candidates from the any address ports as backups, if they ever
surface, are not preferred if the other candidates have at least the
same network condition.
Bug: webrtc:9468
Change-Id: I20c3a40e9a75b8fb34fad741ba5f835ecc3b0d92
Reviewed-on: https://webrtc-review.googlesource.com/85880
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23807}
Also adds tests, and adds a bit of logging in ParseIceServers.
Bug: chromium:718508
Change-Id: Id41ccb7cccbdab5af76e380b32b4d8ba0c4a0a72
Reviewed-on: https://webrtc-review.googlesource.com/86121
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23769}
Those are static functions in the spec, so implemented as member functions
of the PeerConnectionFactory instead.
Bug: webrtc:7577, webrtc:9441
Change-Id: Iccb24180e096e713d24e7e25ecfd5d7bbd7638f9
Reviewed-on: https://webrtc-review.googlesource.com/85341
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23768}
Both incoming and outgoing datachannels should cause
the DATA_ADDED flag to be set.
This CL also moves all tests into their own file, and
improves scaffolding.
Bug: chromium:718508
Change-Id: I5c4c257ccb6f26799f7593bce8b27ebf59015b1e
Reviewed-on: https://webrtc-review.googlesource.com/85348
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23766}
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.
Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}
This can happen with the following sequence of API calls:
1) AddTrack(track) + offer/answer
2) RemoveTrack(track's sender) + offer/answer
3) AddTrack(same track)
Since the first transceiver had already been used to send, it will
not get re-used by the second call to AddTrack. Another RtpSender
will be created with its ID = the track ID. But the code hits a
DCHECK when CreateOffer is later called since both m= sections will
offer the same track ID component of the MSID.
The fix implemented here is to randomly generate a sender ID if
there is already an RtpSender with the track's ID.
Bug: webrtc:8734
Change-Id: Ic2dda23d66e364e77ff7505e1c37e53105a17dae
Reviewed-on: https://webrtc-review.googlesource.com/84249
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23748}
This change also standardizes the RtpSender to a single constructor
and moves the |track| and |stream_ids| arguments to setter methods.
Bug: webrtc:8734
Change-Id: I227a84868a80797f6cc2a1af6eec6d76da8ea159
Reviewed-on: https://webrtc-review.googlesource.com/84248
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23730}
The purpose is to make the fixture reusable in downstream
projects. The CL adds the following things to API:
- api/test/video_quality_test_fixture.h
- api/test/create_video_quality_test_fixture.h
The following things are moved to API:
- call/bitrate_constraints.h (api/bitrate_constraints.h)
- call/simulated_network.h (api/test/simulated_network.h)
- call/media_type.h (api/mediatypes.h)
These are required by the params struct passed to the
fixture. I didn't attempt to split the params struct into
an internal-only and public version in this CL, and as
a result we need to pull in the above things. They are
quite harmless though, so I think it's worth it in order
to avoid splitting up the test config struct.
This CL doesn't solve all the problems we need to
implement downstream tests; we probably need to upstream
tracing variants of FakeNetworkPipe for instance, but
that will come later. This puts in place the basic
structure for now.
Bug: None
Change-Id: I35e26ed126fad27bc7b2a465400291084f6ac911
Reviewed-on: https://webrtc-review.googlesource.com/69601
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23714}
This involves treating it just like "detailed", for now.
At a later stage we might want to modify codec parameters for it.
Bug: chromium:852701
Change-Id: I24678e1f7711bf03ca22273afaaf338e9e3ba1fe
Reviewed-on: https://webrtc-review.googlesource.com/83582
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Peter Boström <pbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23701}
Target bitrate is set to 0.75 of the max bitrate.
Bug: webrtc:9341, webrtc:8655
Change-Id: I9a8c8bb95bb1532d45f05578832418464452340e
Reviewed-on: https://webrtc-review.googlesource.com/79821
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23676}
EndToEndConnectionTimeWithTurnTurnPair was failing intermittently due to
a DCHECK being hit in ports.cc. This was caused by the ScopedFakeClock
being destroyed before the ports. The ports miscalculated a large
negative number for the rtt of a STUN request/response due to the global
clock changing. This fixes the problem by closing the PeerConnections
before the ScopedFakeClock goes out of scope.
Bug: webrtc:9422
Change-Id: Ia4aa3f638dff5da4317a35cf1514ec61472d0d74
Reviewed-on: https://webrtc-review.googlesource.com/84241
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23670}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'pc'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Ide3b9eb32df7f25991f898ac58fcb119c9f8ae12
Reviewed-on: https://webrtc-review.googlesource.com/84181
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23669}
This is the simplest possible fix, returning SSRC stats with a missing
track ID instead of returning no SSRC stats at all.
This means calling GetStats with the track selector argument will still not
work in this case.
Bug: webrtc:3342
Change-Id: I6b58fd5ac15b49274d3f1655e78ae36c4575e5fd
Reviewed-on: https://webrtc-review.googlesource.com/82260
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23667}
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}
Extract rtc_base/base64.h and rtc_base/base64.cc into separate target
to prepare to move them into third_party
Bug: webrtc:8366
Change-Id: I477e6da2b9d09307439b3272261f31042f479d74
Reviewed-on: https://webrtc-review.googlesource.com/83980
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23645}
This experiment is now wired up inside of chrome using field trial and
this passthrough is now obsolete.
Bug: chromium:794608
Change-Id: I1407e391d39c7e8696add9f656f059e7d8a27a08
Reviewed-on: https://webrtc-review.googlesource.com/82780
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23625}
Besides using the MetricsObserverInterface, using RTC_HISTOGRAM_ENUMERATION
directly using RTC_HISTOGRAM_ENUMERATION to report the error which is
needed by internal projects.
Bug: b/110121202, webrtc:9409
Change-Id: I1aaece91200905ea0495229dc2b5e62b1d61279b
Reviewed-on: https://webrtc-review.googlesource.com/83565
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23616}
This fixes a crash that occurs with this sequence of events:
1. AddTrack. SetLocalDescription(CreateOffer())
2. RemoveTrack. SetLocalDescription(CreateOffer())
3. AddTrack.
When AddTrack is called again it re-uses the RtpTransceiver/
RtpSender and try to configure the underlying MediaChannel. But the
MediaChannel would DCHECK since the send stream had been destroyed
by the SLD in 2. and would not get created until SLD is called
again.
Bug: webrtc:9401
Change-Id: I4b5572886e17263aaa4ce0408663444d72e09243
Reviewed-on: https://webrtc-review.googlesource.com/83420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23605}
This policy is not implemented.
Bug: None
Change-Id: I6c162d61c2488a4726c20df5c14439f83633a198
Reviewed-on: https://webrtc-review.googlesource.com/76041
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23591}
The PeerConnection integration test was creating TurnServers on the
stack on the signaling thread. This could cause a race condition problem
when the test was being taken down. Since the turn server was destructed
on the signaling thread, a socket might still try and send to it after
it was destroyed causing a seg fault. This change creates/destroys the
TestTurnServers on the network thread to fix this issue.
Bug: None
Change-Id: I080098502b737f0972ce2fa5357920de057a3312
Reviewed-on: https://webrtc-review.googlesource.com/81301
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23590}
Add a new flag to RtcConfiguration. By setting that flag to true, the
SRTP parameters will be reset whenever the DTLS transports are reset
after every offer/answer negotiation.
The flag is added to Android and Objc wrapper as well.
This should only be used as a workaround for the linked bug, if the
application knows that the other party is affected (for instance,
using a version number).
TBR=sakal@webrtc.org, denicija@webrtc.org
Bug: chromium:835958
Change-Id: I6db025e1c69bf83e1b1908f7df4627430db9920c
Reviewed-on: https://webrtc-review.googlesource.com/83101
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23587}
This CL fixes a bug that the RtcEventLog owned by PeerConnection was not
passed to P2PTransportChannel after JsepTransportController was
introduced to deprecate the legacy TransportController.
Bug: webrtc:9337
Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5
Reviewed-on: https://webrtc-review.googlesource.com/79964
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23572}
This reverts commit bae103126c5bdaf1361bcff4750eb5ebe10020ee.
Reason for revert: Merge native code change with Android and Objc wrapper.
Original change's description:
> Add a flag to actively reset the SRTP parameters
>
> Add a new flag to RtcConfiguration. By setting that flag to true, the
> SRTP parameters will be reset whenever the DTLS transports are reset
> after every offer/answer negotiation.
>
> This should only be used as a workaround for the linked bug, if the
> application knows that the other party is affected (for instance,
> using a version number).
>
> Bug: chromium:835958
> Change-Id: Ifb4b99f68dc272507728ab59c07627f0d1b9c605
> Reviewed-on: https://webrtc-review.googlesource.com/81642
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23570}
TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
Change-Id: Ibd7a3b8f45ff8df4af33d758f8fd3e2d5158e8e2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:835958
Reviewed-on: https://webrtc-review.googlesource.com/83080
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23571}
