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}
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}
To address this, this CL introduces a PeerConnectionFactoryDependencies
structure to encapsulate all mandatory and optional dependencies (where a
dependency is defined as non trivial executable code that an API user may want
to provide to the native API). This allows adding a new injectable dependency
by simply adding a new field to the struct, avoiding the hassle described above.
Bug: webrtc:7913
Change-Id: Ice58fa72e8c578b250084a1629499fabda66dabf
Reviewed-on: https://webrtc-review.googlesource.com/79720
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23480}
The proper closing procedure is:
1. Alice resets outgoing stream.
2. Bob receives incoming stream reset, resets his outgoing stream.
3. Alice receives incoming stream reset; channel closed!
4. Bob receives acknowledgement of reset; channel closed!
https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.7
However, up until now we've been sending both an incoming and outgoing reset
from the side initiating the closing procedure, and doing nothing on the remote
side.
This means that if you call "Close" and the remote endpoint is using an old
version of WebRTC, the channel's state will be stuck at "closing" since the
remote endpoint won't send a reset. Which is already what happens when Firefox
is talking to Chrome.
This CL also fixes an issue where the DataChannel's state prematurely went to
"closed" before the closing procedure was complete. Which could result in a
new DataChannel attempting to re-use the ID and failing.
TBR=magjed@webrtc.org
Bug: chromium:449934, webrtc:4453
Change-Id: Ic1ba813e46538c6c65868961aae6a9780d68a5e2
Reviewed-on: https://webrtc-review.googlesource.com/79061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23478}
Both AudioRtpSender and VideoRtpSender receive stream_ids in their
constructor, no need to call set_stream_ids again.
Bug: None
Change-Id: I6238a6d6e31076a0b3245c89e2825d8dee5166c0
Reviewed-on: https://webrtc-review.googlesource.com/80220
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23476}
This generates a number that represent a set of bits that
indicates how a PeerConnection has been used over time.
Bug: chromium:718508
Change-Id: I6df177684c50bc825bc41ea97996574292084d41
Reviewed-on: https://webrtc-review.googlesource.com/79823
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23471}
Only use replaced with FakePeriodicVideoTrackSource.
Bug: webrtc:6353
Change-Id: Iee38b98a5242a292a848738bde05de18d96de7f4
Reviewed-on: https://webrtc-review.googlesource.com/79441
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23441}
This removes usage of the old OnFailure methods on CreateSessionDescriptionObserver
and SetSessionDescriptionObserver, so that WebRTC will continue to compile
once all the default implementations are removed.
Bug: chromium:589455
Change-Id: Id67295b3ad0c30d24d79589c2041acdd507a19f3
Reviewed-on: https://webrtc-review.googlesource.com/78480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23427}
Experiment flag added to PeerConnectionInterface::RtcConfiguration and
propagated down to VideoStreamEncoder.
Artificial Sdp parameter is added to the sdp format if the flag is set.
Additionally, sdp format is propagated in vp8 simulcast adapters.
Bug: chromium:794608
Change-Id: I2dec54d19ae7bfbd5f2777ec682da5a84194da94
Reviewed-on: https://webrtc-review.googlesource.com/78500
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23412}
It was being parsed, but not serialized. Meaning that if you set a
remote description with a=ice-lite, and then read the remoteDescription
attribute, it doesn't contain a=ice-lite.
NOTRY=True
Bug: webrtc:6668
Change-Id: Ia3c56d876c317b5af71a1f383f238d1e86f06a01
Reviewed-on: https://webrtc-review.googlesource.com/78821
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23391}
