This was previously not working because the answerer wasn't generating
ICE credentials when it should have been.
This was fixed inadvertently by:
https://webrtc-review.googlesource.com/c/src/+/46380
But we should really also have a PeerConnection-level regression test
for this.
Bug: webrtc:6023
Change-Id: I3da900edcc8db8034ed61a7bb981d9c0e616254e
Reviewed-on: https://webrtc-review.googlesource.com/69403
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22832}
This was working before, but somewhat by accident (because an error
wasn't being surfaced).
This CL also starts surfacing that error, from
JsepTransportController::AddRemoteCandidates to PeerConnection.
Bug: None
Change-Id: Ib48c9c00ea2a5baa5f7e3210c5dc7a339498b2d0
Reviewed-on: https://webrtc-review.googlesource.com/69015
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22830}
This changes the PeerConnection when in Unified Plan mode to reject
SDP applied with SetLocalDescription or SetRemoteDescription if the
SDP has multiple "Plan B tracks" (a=ssrc lines) in a media section.
The error is to inform developers that the given SDP will not be
interpreted as they might expect.
Bug: None
Change-Id: I7a0e11282fbf63dac06038cd22a66683517a87d0
Reviewed-on: https://webrtc-review.googlesource.com/68764
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22829}
This reverts commit 4feb2044db4b3563323c24e2372885cbf7708c73.
Reason for revert: Landscape video was not showing as aspect fit as before. .
Original change's description:
> Fix rendering on an iPhone X's tall screen.
>
> Bug: webrtc:8884
> Change-Id: I850e4ea1919837e15a78c90968a4879a1ccbd22c
> Reviewed-on: https://webrtc-review.googlesource.com/52761
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22011}
TBR=magjed@webrtc.org,kthelgason@webrtc.org,jtteh@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:8884
Change-Id: I17bcdaf945d74540538162934cd3265240cc9302
Reviewed-on: https://webrtc-review.googlesource.com/68841
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22828}
OnRatesUpdated() is called every time the bitrate estimate, or once per
second. However, since we don't want to reconfigure libvpx too often,
just in case it interferes with the rate controller, so
ScreenshareLayers contains a boolean |bitrate_update_| which indicate
if the configuration should be updated on a call to
UpdateConfiguration().
However, it two rate updates happened between two frames, the first of
which changes the rates and second one does not, |bitrate_update_| will
be reset to false and the encoder won't get the desired config.
This CL makes sure we update the configuration iff the rate has changed
at any time since the last call to UpdateConfiguration().
Bug: webrtc:9012
Change-Id: I62af36cffe20ecb7d3f403b3eb11f23a9692d719
Reviewed-on: https://webrtc-review.googlesource.com/69040
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22826}
Not including <iterator> creates problems on some build systems because
std::back_inserter is defined there.
Bug: None
Change-Id: I27180f72dd327e3a0caab35d3a33907f1e0c4296
Reviewed-on: https://webrtc-review.googlesource.com/69323
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22825}
Contents of the template class VideoMediaChannelTest copied into
VideoMediaChannelTest, the only user.
Bug: None
Change-Id: Ie43a7c4bc1e85e2df77361f43776d4d902b74cae
Reviewed-on: https://webrtc-review.googlesource.com/67400
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22822}
The code which triggered the memcheck failure is gone.
Bug: webrtc:5989
Change-Id: I35decfae1af4e988724ababa58c41fca7a2c4a67
Reviewed-on: https://webrtc-review.googlesource.com/69261
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22821}
Mainly remove CHECKinitialized_ macro and AGC functionality. Also make
actual behavior clearer in some functions.
Bug: webrtc:7452
Change-Id: I1eac86f4eaff7b14820d3e4192b15c20ab6acb45
Reviewed-on: https://webrtc-review.googlesource.com/69161
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22820}
Currently there are several checks against _lastDrawnFrame in RTCEAGLVideoView.mm but this variable is not assigned anywhere. Seems like it was missed in 13941912b1 during work on injecting custom shaders.
Bug: webrtc:9133
Change-Id: Ie979a63de343e7253e4b4e70e3b98ffb0880af04
Reviewed-on: https://webrtc-review.googlesource.com/68720
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22819}
This reverts commit fc43d11717e16dd427ac84fee614e5511e43cefd.
Reason for revert: Crashes downstream tests
Original change's description:
> Add thread checker to PortAllocator and its subclasses and fix a bug
> causing memory contention by threads.
>
> PortAllocator and its subclasses assume all of their methods except the
> constructor must be called on the same thread (the network thread in
> practice). This CL adds a thread checker to PortAllocator and its
> subclasses for thread safety, and fixes bugs of invoking some of their
> methods in PeerConnection on the signaling thread.
>
> Bug: webrtc:9112
> Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
> Reviewed-on: https://webrtc-review.googlesource.com/66945
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22814}
TBR=deadbeef@webrtc.org,pthatcher@google.com,pthatcher@webrtc.org,qingsi@google.com,honghaiz@webrtc.org
Change-Id: I2db6561d5d6366d38caa58c3e719d0d48eda70c2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9112
Reviewed-on: https://webrtc-review.googlesource.com/69200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22818}
A part of the Audio Processing Module interface is GetStatistics. The
call collects stats from submodules. We make sure these calls are made
by the fuzzer to cover that code path.
Bug: webrtc:7820
Change-Id: Ia8f89d9838602dcb2599f676bd5c43e815bbf791
Reviewed-on: https://webrtc-review.googlesource.com/68980
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22817}
causing memory contention by threads.
PortAllocator and its subclasses assume all of their methods except the
constructor must be called on the same thread (the network thread in
practice). This CL adds a thread checker to PortAllocator and its
subclasses for thread safety, and fixes bugs of invoking some of their
methods in PeerConnection on the signaling thread.
Bug: webrtc:9112
Change-Id: I33ba9bae72ec09a45ec70435962f3f25cd31583c
Reviewed-on: https://webrtc-review.googlesource.com/66945
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22814}
Reject the local/remote description trying to change the pre-negotiated
BUNDLE tag.
Reject an answer containing a BUNDLE group that's not a subset of the offered group.
Reject an offer/answer with a BUNDLE group containing a MID that no m= section has.
Reject an answer removes an m= section from an established BUNDLE group without
rejecting it.
Bug: chromium:827917
Change-Id: If334eefb00b1c1c1e24f9afba0cb00b5867f5590
Reviewed-on: https://webrtc-review.googlesource.com/67190
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22813}
These two new histograms relate to the packet-loss concealment that
happens when audio packets are lost or late for decoding, and the
NetEq must resort to extrapolating audio from the previously
decoded data.
Bug: webrtc:9126
Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b
Reviewed-on: https://webrtc-review.googlesource.com/67861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22812}
This is a reland of dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8
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}
Bug: webrtc:8415
Change-Id: I0ea7d4d679e7845907cfbe9a120f128ff2180e4b
Reviewed-on: https://webrtc-review.googlesource.com/68580
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22810}
This CL changes the handling of saturated microphone signals in AEC3.
Some of the changes included are
-Make the detection of saturated echoes depend on the echo path gain
estimate.
-Remove redundant code related to echo saturation.
-Correct the computation of residual echoes when the echo is saturated.
-Soften the echo removal during echo saturation.
Bug: webrtc:9119
Change-Id: I5cb11cd449de552ab670beeb24ed8112f8beb734
Reviewed-on: https://webrtc-review.googlesource.com/67220
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22809}
The ostream operators does not work consistently with gtest since the
definitions would have to be included before the first include of gtest.
This is tricky to ensure and the end result is that the operators will
sometimes work and sometimes not without obvious explanations.
To avoid causing confusing behavior, this CL removes the operators
pending a better solution.
Bug: None
Change-Id: I66bead0efb7246d368359ddf9e9bfad9d67c05da
Reviewed-on: https://webrtc-review.googlesource.com/68640
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22808}
The static initializers caused issues with chromium build. This should
fix it for now.
Bug: None
Change-Id: I4592df0e2bd02980421bb6319c24e7b6983d2252
Reviewed-on: https://webrtc-review.googlesource.com/69020
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22807}
Set the flag based on coded length of buffered frame which is reset
after picture is encoded and, thus, is equal to zero when encoder
delivers first frame of next picture.
Before this change first_frame_in_picture was set based on index of
spatial layer of encoded frame. This is not right anymore since encoder
can drop base layer but deliver upper layers.
Bug: chromium:828350
Change-Id: I12c7534240de8bc4905f04ff368cc3704720a70b
Reviewed-on: https://webrtc-review.googlesource.com/68561
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22805}
Recent Clang versions fixed a bug which had previously allowed some casts that
removed qualifiers to go undiagnosed.
This fixes the following kind of error:
./../third_party/webrtc/api/optional.h:41:35: error: reinterpret_cast from
'const int *' to 'void *' casts away qualifiers
FunctionThatDoesNothingImpl(reinterpret_cast<void*>(x)));
^~~~~~~~~~~~~~~~~~~~~~~~~~
../../third_party/webrtc/api/optional.h:280:45: note: in instantiation of
function template specialization
'rtc::optional_internal::FunctionThatDoesNothing<const int>' requested here
return has_value_ ? *optional_internal::FunctionThatDoesNothing(&value_)
^
../../third_party/webrtc/call/rtp_bitrate_configurator.cc:70:53:
note: in instantiation of member function 'rtc::Optional<int>::value_or'
requested here
std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0),
^
Bug: chromium:831081
Change-Id: I032ebd1f052fa2a50548e984febb7fa462df42ea
Reviewed-on: https://webrtc-review.googlesource.com/68941
Commit-Queue: Hans Wennborg <hans@chromium.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22804}
instead of relying on optional.h to included these 2 headers.
Bug: webrtc:9078
Change-Id: I7a4b3facd81690b8f107640487e129986c1f5ff6
Reviewed-on: https://webrtc-review.googlesource.com/68602
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22803}
The given IOSurfaceRef was ignored until now. Wrap it into the new
DesktopFrameIOSurface. The new DesktopFrameProvider object is there
to manage them as it has to be done per display id.
From initial measurement this speed-up the frame capture by 2.
Disabled by default for now but it can be enabled by calling
options.set_use_iosurface. This CL will allow to do some advanced
tests.
Bug: webrtc:8652
Change-Id: Ia9ac0b69b30098774941cb378804b45cb1710119
Reviewed-on: https://webrtc-review.googlesource.com/33014
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22801}
This reverts commit a8f13ccad400eb8ff84a379042c0595951ca9658.
Reason for revert: It's causing no video to be shown after the 1st call.
Original change's description:
> Improve thread-safety of MTL Renderer.
>
> Bug: b/77579859
> Change-Id: I427d0f41593155dc5cbf98a09d7ec826497b803c
> Reviewed-on: https://webrtc-review.googlesource.com/67040
> Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22795}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
Change-Id: Ia8f33995e087178f1c3be7753f70be8ba18447f8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: b/77579859
Reviewed-on: https://webrtc-review.googlesource.com/68860
Reviewed-by: JT Teh <jtteh@webrtc.org>
Commit-Queue: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22800}
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}
