Moving the responsibility for calling callbacks from implementations
of NetworkControllerInterface to SendSideCongestionController. This
decreases the coupling and makes the callbacks more explicit.
Bug: webrtc:8415
Change-Id: Ie75effbde01533106080bb6c40308b0c20064c45
Reviewed-on: https://webrtc-review.googlesource.com/66882
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22793}
This CL adds a field trial to enable the BBR congestion control method.
Since BBR is only implemented to handle per packet feedback,
SendSideCongestionController is modified to recreate network controllers
when the packet feedback availability changes and the BBR experiment is
enabled.
This also means that the periodic task used for process updates in the
network controllers has to recreated.
Bug: webrtc:8415
Change-Id: Ia24f7ad35336d2cc7a02bb3a445f1a84b8643475
Reviewed-on: https://webrtc-review.googlesource.com/61520
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22791}
When building test:test_support_unittests with is_official_build=true,
the linker fails with the following error:
duplicate symbol: webrtc::videocapturemodule::VideoCaptureImpl::Create(
char const*)
>>> defined in obj/modules/video_capture/video_capture_internal_impl/\
video_capture_linux.o
>>> defined in obj/modules/video_capture/libvideo_capture.a(\
video_capture_external.o)
After looking at both test:test_support_unittests and test:test_support,
it seems these targets had unused dependenicies. This CL removes them
and fixes the duplicated symbol error.
The GN flag is_official_build changes some configurations down in the
toolchain, that is probably why building with is_official_build=false
was not triggering the problem.
In any case, build targets in test/ need to be cleaned up because they
depend on too many things.
Bug: webrtc:9117
Change-Id: Icfdae3b5610f1c873ccdd0292c12ef946dea79af
Reviewed-on: https://webrtc-review.googlesource.com/67161
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22789}
This reverts commit dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8.
Reason for revert: breaks downstream projects
Original change's description:
> Storing frame if encoder is paused.
>
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
>
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
>
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}
TBR=sprang@webrtc.org,srte@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8415
Change-Id: I4449eca65a64e2bc2fb25d866e5775e9a085cee9
Reviewed-on: https://webrtc-review.googlesource.com/68280
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22788}
Adding Stop method of periodic tasks in SendSideCongestionController
(SSCC). This is utilized in a later CL enabling switching the network
controller which requires stopping the old periodic task and starting a
new one with a new update period.
Bug: webrtc:8415
Change-Id: I2e56c1e1fe10d88c038b2f290d94c08723ddf4e4
Reviewed-on: https://webrtc-review.googlesource.com/67280
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22786}
This argument was previously used to implement track muting
(black frames) in the video engine, but that now happens in
the VideoTrack/VideoBroadcaster upstream.
Bug: webrtc:6983
Change-Id: Ib721b297d9fbe55b641c56690dbbd37a52edbb2f
Reviewed-on: https://webrtc-review.googlesource.com/67341
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22785}
It takes some time for rate controller to adapt to content. Quality of first
frames is usually worse than quality of following frames. It makes sense to
exclude first frames from analysis and, thus, avoid negative affect of this
interval on overall results.
Bug: none
Change-Id: Ib0a258889750cf794c7d6fdff26af958f7bbe48a
Reviewed-on: https://webrtc-review.googlesource.com/66100
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22782}
Adds a pending frame to VideoStreamEncoder that is used to store frames
that are not sent because the encoder is paused. If the encoder is
resumed within 200 ms, the pending frame will be encoded and sent. This
ensures that resuming a stream instantly starts sending frames if it is
possible.
This also protects against a race between submitting the first frame
and enabling the encoder that caused flakiness in end to end tests
when using the task queue based congestion controller.
Bug: webrtc:8415
Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
Reviewed-on: https://webrtc-review.googlesource.com/67141
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22781}
more standard optional<T> inlines compares instead of converting second argument to T.
that leads to warnings about comparing unsigned to signed integers.
Bug: webrtc:9078
Change-Id: I43cc729d3b85d789b0c394064dc7e11dc27a37aa
Reviewed-on: https://webrtc-review.googlesource.com/66782
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22779}
This removes the optimization that would stop sending the MID RTP
header extension when an RTCP report block is received. The old
implementation was not flexible enough for the API, and making
those changes is too involved at this time as we need this to work
now to unblock other work.
Bug: webrtc:4050
Change-Id: I099f8e9047a40993d93bcda9164eb82fdf810387
Reviewed-on: https://webrtc-review.googlesource.com/67192
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22776}
Replacing observer interface with polling for pending probe clusters.
The purpose is to make it easier to reason about and control side
effects and to prepare for a similar change in the network controller
interface.
Bug: webrtc:8415
Change-Id: I8101cfda22e640a8e0fa75f3f6e63876db826a89
Reviewed-on: https://webrtc-review.googlesource.com/66881
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22775}
So that we don't have to be capable of creating one ourselves, which
requires a dependency on the audio decoders.
BUG=webrtc:5801, webrtc:8396
Change-Id: I80749ec3b86cba73994307046d05964f59167d44
Reviewed-on: https://webrtc-review.googlesource.com/18440
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22774}
Setting a default value for a class members prevents memory sanitizer
to behave correctly and may confuse the reader.
Instead, one should use rtc::MsanUninitialized, which creates an object of
a given type and marks its memory as uninitialized.
This prevents issues in production (due to uninitialized memory) and
allows MemorySantizier to catch invalid access patterns.
Bug: webrtc:8762
Change-Id: I74c79caa9c19ea85708e89e24bc5516c4d9d12a1
Reviewed-on: https://webrtc-review.googlesource.com/52342
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22773}
This reverts commit 3fb3939896f6270d48aff34eee2946bd7661bd63.
Reason for revert: Downstream projects failures.
Original change's description:
> Floating-point exception observer for unit tests
>
> This CL adds a simple tool that let a unit test fail if a floating
> point exception occurs. It is possible to focus on specific exceptions.
> Note that FloatingPointExceptionObserver is only effective in debug
> mode. For this reason, the related unit tests only run in debug mode.
> Plus, due to some platform-specific limitations, not all the floating
> point exceptions are available on Android.
>
> Bug: webrtc:8948
> Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
> Reviewed-on: https://webrtc-review.googlesource.com/58097
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22768}
TBR=phoglund@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org
Change-Id: I0fd3d114ab4a348fd46339e98273e19c1ac1c6dc
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8948
Reviewed-on: https://webrtc-review.googlesource.com/67380
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22769}
This CL adds a simple tool that let a unit test fail if a floating
point exception occurs. It is possible to focus on specific exceptions.
Note that FloatingPointExceptionObserver is only effective in debug
mode. For this reason, the related unit tests only run in debug mode.
Plus, due to some platform-specific limitations, not all the floating
point exceptions are available on Android.
Bug: webrtc:8948
Change-Id: I0956e27f2f3aa68771dd647169fba7968ccbd771
Reviewed-on: https://webrtc-review.googlesource.com/58097
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22768}
This allows clients to enable Receiver reference time reports via
PeerConnection.
RRTR is not enabled by default but can be added to SDP string.
Bug: webrtc:9108
Change-Id: I851f0d65152875bf115553a851b839f83e3d241e
Reviewed-on: https://webrtc-review.googlesource.com/66861
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22767}
This is a reland of 8ac9bb4d52a687b34158dc52c8c25830b23b8333
Original change's description:
> Added BBR network controller.
>
> BBR is a congestion control method that is initially developed for TCP.
> This CL adds an implementation of BBR ported from QUIC for use with
> WebRTC. An upcoming CL enables it via a field trial.
>
> Bug: webrtc:8415
> Change-Id: Ie4261d2e43bafa15aa928a7cadcfec256107cdbc
> Reviewed-on: https://webrtc-review.googlesource.com/39788
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22647}
Bug: webrtc:8415
Change-Id: I090e4116d1f470acbd64af31520654e1bd8dfcda
Reviewed-on: https://webrtc-review.googlesource.com/65200
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22766}
This CL introduces sdk/android/api/org/webrtc/audio/AudioDeviceModule.java,
which is the new interface for audio device modules on Android.
This CL also refactors the main AudioDeviceModule implementation, which
is sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java and makes
it conform to the new interface. The old code used global static methods
to configure the audio device code. This CL gets rid of all that and uses
a builder pattern in JavaAudioDeviceModule instead. The only two dynamic
methods left in the interface are setSpeakerMute() and setMicrophoneMute().
Removing the global static methods allowed a significant cleanup, and e.g.
the file sdk/android/src/jni/audio_device/audio_manager.cc has been
completely removed.
The PeerConnectionFactory interface is also updated to allow passing in
an external AudioDeviceModule. The current built-in ADM is encapsulated
under LegacyAudioDeviceModule.java, which is the default for now to
ensure backwards compatibility.
Bug: webrtc:7452
Change-Id: I64d5f4dba9a004da001f1acb2bd0c1b1f2b64f21
Reviewed-on: https://webrtc-review.googlesource.com/65360
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22765}
Added functionality on the FakeNetworkPipe to introduce arbitrary
clock offsets. This offset is added to the reported receive time of
all packets. This prepares for a later CL using this to test correction
of receive time stamps.
Bug: webrtc:9054
Change-Id: I811b3aa8359bc917f59443088d8a418368242db9
Reviewed-on: https://webrtc-review.googlesource.com/64726
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22763}
This will enable changing thresholds when switching between hardware
and software encoders. It is also a partial revert of
https://webrtc-review.googlesource.com/33340: construction of the
OveruseFrameDetector is still in VideoSendStream, but configuration is
moved back to VideoStreamEncoder.
Longer term, information about HW vs SW, or generally, about resources
consumed by the encoder, should be passed in the per-frame callbacks
to OveruseFrameDetector, and then the CpuOveruseOptions could move
back to construction time.
Bug: webrtc:8504, webrtc:8830
Change-Id: I44577519d4e05356730cac9bd9ae3c74bfc17ed7
Reviewed-on: https://webrtc-review.googlesource.com/65163
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22761}
A downstream bug ocurred because of a lack of symmetry when adding and
removing a remote sender in Plan B that specifies SSRCs, but doesn't
specify stream IDs. The issue when the first remote description is
applied "default" for the stream ID on the remote sender, but the
second time it's applied the current remote sender's "default" stream
ID does not match the new remote description's empty stream ID. This
was incorrectly interpreted as a new remote sender (which removed/added
the sender).
Bug: webrtc:7933
Change-Id: I87191b9e887b3450ef15111b5e867023c723a86e
Reviewed-on: https://webrtc-review.googlesource.com/67191
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22760}
This changeset refactors the OpenSSLSessionCache out of the Factory. Instead of
directly injecting a pointer to the factory to each OpenSSLAdapter instead just
a pointer to the OpenSSLSessionCache is submitted which the Factory is the sole
owner of. This provides a cleaner dependency injection interface and allows the
OpenSSLSessionCache to be tested independently of the factory that uses it. It
also allows for the factories role to be more clearly defined allowing for
additional dependency injection in future updates.
This change also removes the habit of having OpenSSL typedefs around certain
functions and instead uses the standardised ossl_typ.h header which contains
these typedefs. This makes the headers more directly tied to just what they are
responsible for doing.
Bug: webrtc:9085
Change-Id: I7938178b70acc613856139d387a1b46928dca6ad
Reviewed-on: https://webrtc-review.googlesource.com/66941
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22758}
This profile will now not be used unless the application explicitly
sets the flag in CryptoOptions to true. As a result, an 80-bit
authentication tag will be used instead of a 32-bit one. See bug for
more details.
Bug: webrtc:7670
Change-Id: I7c0a118fd7b1e7aac23b9eb8717099f055de0441
Reviewed-on: https://webrtc-review.googlesource.com/66600
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22757}
This changeset addresses concerns about how the OpenSSLAdapter does certificate
name matching. The current approach has a number of issues which are outlined
in the bug description. The approach taken in this changeset is to use the
standard function X509_check_host which should correctly parse the wildcard
expansions and is directly supported in OpenSSL instead of attempting my own
implementation. This changeset uses this as an opportunity to add additional
parameter checking and refactoring logging code out of the main code path.
Bug: webrtc:8888
Change-Id: Iaffe1daddcd52193ba674489f613ce8515b81e91
Reviewed-on: https://webrtc-review.googlesource.com/65022
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22755}
With the latest usrsctp roll, the MTU value you provide is the space
avaiable for chunks in the packet. We previously specified this to be the
MTU for the entire SCTP packet, so we were logging errors when the SCTP
packets were 12 bytes larger than expected (the size of the SCTP header).
This fix updates our MTU specified to account for the SCTP header size
as well.
Bug: webrtc:9082
Change-Id: Id3bfa839d4e7662230111ebbdf33bd81ccdc7cf4
Reviewed-on: https://webrtc-review.googlesource.com/66943
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22754}
|is_posix| will be switched to false for Fuchsia, this is a preliminary change.
Bug: chromium:812974
Change-Id: I3bfda3e056ad1e5229834286ce5d095d9204a428
Reviewed-on: https://webrtc-review.googlesource.com/65782
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Fabrice de Gans-Riberi <fdegans@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22753}
This CL implements the functions related to decoding.
Bug: webrtc:8909
Change-Id: Iefa3c1565a9b9ae93f14992b4a1cca141b7c5193
Reviewed-on: https://webrtc-review.googlesource.com/66403
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22747}
