This ensures that the video_frame_buffer method never can return a
null pointer.
BUG=webrtc:6591
Review-Url: https://codereview.webrtc.org/2541863002
Cr-Commit-Position: refs/heads/master@{#15574}
AddCodec represents better what this function actually does.
BUG=None
Review-Url: https://codereview.webrtc.org/2573593003
Cr-Commit-Position: refs/heads/master@{#15565}
The Chromium mock implementation implements the new GetStats API, so we
can remove this default implementation.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2566143002
Cr-Commit-Position: refs/heads/master@{#15563}
This CL doesn't start *using* a=bundle-only; it just adds support for
parsing it. We need to do this first, because otherwise old versions of
WebRTC will interpret a zero port value as a rejected m= section.
BUG=webrtc:4674
Review-Url: https://codereview.webrtc.org/2562183002
Cr-Commit-Position: refs/heads/master@{#15558}
Reason for revert:
A interface change broke some downstream code in google3.
Original issue's description:
> Support external audio mixer in webrtc.
>
> An external audio mixer will be passed from PeerConnectionFactory to
> AudioTransportProxy.
>
> BUG=webrtc:6457
>
> Committed: https://crrev.com/f6bcac59e88c3be5ffd73bbb1098f2d82ee316a1
> Cr-Commit-Position: refs/heads/master@{#15556}
TBR=solenberg@webrtc.org,aleloi@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2562333003
Cr-Commit-Position: refs/heads/master@{#15557}
An external audio mixer will be passed from PeerConnectionFactory to
AudioTransportProxy.
BUG=webrtc:6457
Review-Url: https://codereview.webrtc.org/2539213003
Cr-Commit-Position: refs/heads/master@{#15556}
It was using a non-WebRTC-named header guard, which could conflict with
other similarly named/intended headers.
BUG=None
NO_DEPENDENCY_CHECKS=true
Review-Url: https://codereview.webrtc.org/2548113002
Cr-Commit-Position: refs/heads/master@{#15554}
SimulcastEncoderAdapter calls Release() on a failed sub-encoder init,
but Release only knows how to clean up encoders that have registered
stream info. Since failed ones don't register, they aren't currently
cleaned up.
BUG=None
Review-Url: https://codereview.webrtc.org/2544003005
Cr-Commit-Position: refs/heads/master@{#15553}
Was added for video initially, but not for audio.
BUG=webrtc:6862
Review-Url: https://codereview.webrtc.org/2568553002
Cr-Commit-Position: refs/heads/master@{#15552}
The packetization parts of this class are accessed from the
encoder thread, which might change under different occasions.
The use of a sequenced task checker here is thus incorrect, since
that requires the access to always be on the same thread, whenever
a task queue is not used.
The access to the instantiated object of this class, at least when
it comes to the RTP packetization parts, is however synchronized
using the lock in PayloadRouter::OnEncodedImage. We can therefore
safely remove the sequenced task checker.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2562983002
Cr-Commit-Position: refs/heads/master@{#15549}
The file was aldready pruned down to the point where it only included
webrtc/typedefs.h. Therefore, all includes of
voice_engine_configurations.h are replaced with typedefs.h, except on
two occasions where it was obvously not needed.
BUG=webrtc:6506
Review-Url: https://codereview.webrtc.org/2553583002
Cr-Commit-Position: refs/heads/master@{#15547}
adaptReason in webrtcvideoengine2.h only defines NONE=0, CPU=1 and BANDWIDTH=2 so &0x4 can not happen anymore.
This was probably never implemented in videoengine2
BUG=webrtc:6870
Review-Url: https://codereview.webrtc.org/1887773002
Cr-Commit-Position: refs/heads/master@{#15546}
The changes here are the same as in https://codereview.webrtc.org/2523993002/:
- reduce packet loss from 50% to 5%, to allow the BWE to ramp up better.
- Do not wait for 100 packets to be sent before letting the test pass.
BUG=webrtc:6851
Review-Url: https://codereview.webrtc.org/2558303002
Cr-Commit-Position: refs/heads/master@{#15542}
This also deletes unused features of the video_capturer interface, the classes
VideoCaptureFeedBack, VideoCaptureEncodeInterface and related methods,
and the module id which used to be passed as an argument to the
VideoCaptureDataCallback.
In theory the module id could have been used to let a single
VideoCaptureDataCallback serve several capturers, and demultiplex
on the id, but in practice, it was unused. With this change, it is
required to use a separate VideoSinkInterface for each capturer.
BUG=webrtc:6789
Review-Url: https://codereview.webrtc.org/2534553002
Cr-Commit-Position: refs/heads/master@{#15540}
This is to allow application to pass an audio network adaptor config string to WebRTC.
BUG=webrtc:6303
Review-Url: https://codereview.webrtc.org/2437803004
Cr-Commit-Position: refs/heads/master@{#15532}
Changing the configuration will cause subsequently generated offers to change
the ufrag/pwd as necessary, so that a new round of gathering is started that
uses the new configuration.
This CL also makes some minor unrelated changes: changing the reference SDP in
the PC tests to more match what we generate, and relaxing the network thread
requirement for JsepTransport (since there's no reason the "needs-ice-restart"
flag can't be accessed from the signaling thread).
BUG=webrtc:6714
Review-Url: https://codereview.webrtc.org/2563153002
Cr-Commit-Position: refs/heads/master@{#15527}
The enum is at about the same level of detail as DOMExceptions, and I
looked through the spec making sure that chromium will be able to perform
the DOMException mapping for each one.
The new enum is called RtcError and is outside the PeerConnectionInterface
scope, because we may want to use this for things not associated with a
PeerConnection in the future.
This CL doesn't yet use the error enum anywhere; that will probably happen
in follow-up CLs for the individual methods.
BUG=webrtc:6855
Review-Url: https://codereview.webrtc.org/2564683002
Cr-Commit-Position: refs/heads/master@{#15526}
ScreenCapturerWinGdi randomly returns black frames in test environment. The root
cause is still unknown, so change ScreenCapturerWinGdi tests into MANUAL mode to
execute in test environment, but unblock other developers. We can eventually get
a failure ratio and more samples for debugging.
BUG=webrtc:6666, webrtc:6843
Review-Url: https://codereview.webrtc.org/2564173002
Cr-Commit-Position: refs/heads/master@{#15518}
This should remove the test flakiness, as before this change there
could be collisions from sequence numbers coming from two sequence
number spaces (the media SSRC and the FlexFEC SSRC). The probability
of collisions was low, and hence the flakes were far between.
This change also reduces the packet loss to 5% (down from ~50%), in
order for the BWE to have an easier time to ramp up.
BUG=webrtc:6825
R=philipel@webrtc.org, mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2554403003
Cr-Commit-Position: refs/heads/master@{#15512}
A decision was recently made to limit downscaling to 320x180 on
Android. This causes the perf tests to fail. This test is no
longer valid on android, as the failure is expected behaviour.
BUG=None
NOTRY=true
TBR=phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/2563913003
Cr-Commit-Position: refs/heads/master@{#15510}
Current implementation of JavaToStdString applies additional encoding that modifies ISO-8859-1 encoded strings (e.g. byte array). This CL is to fix this.
A planned use of this is to pass a protobuf serialized string as a MediaConstraint to WebRTC to configure audio network adaptor.
BUG=webrtc:6815
Review-Url: https://codereview.webrtc.org/2549783002
Cr-Commit-Position: refs/heads/master@{#15509}
Theil and Sen's estimator essentially looks at the line through every pair of points and selects the median slope. This is robust to corruption of up to 29% of the data points.
Wire up new estimator to field trial experiment. Add unit and integration tests. Results are promising.
BUG=webrtc:6728
Review-Url: https://codereview.webrtc.org/2512693002
Cr-Commit-Position: refs/heads/master@{#15508}
Reason for revert:
Fixed perf tests.
Original issue's description:
> Revert of Bump up scaling limit for MediaCodec. (patchset #3 id:40001 of https://codereview.webrtc.org/2566533002/ )
>
> Reason for revert:
> Failed on the perf tests.
>
> Original issue's description:
> > Bump up scaling limit for MediaCodec.
> >
> > Wait until MediaCodec is better tested at these low
> > resolutions, and until some fallback mechanism is in place
> > before lowering this threshold.
> >
> > BUG=webrtc:6837
> >
> > Committed: https://crrev.com/3e9b1330467edf6b5af609b375c15efb9e6b4933
> > Cr-Commit-Position: refs/heads/master@{#15498}
>
> TBR=magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6837
>
> Committed: https://crrev.com/1cd0a0ab43846072b1e2f37c953ecd770feb5963
> Cr-Commit-Position: refs/heads/master@{#15500}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6837
Review-Url: https://codereview.webrtc.org/2567613002
Cr-Commit-Position: refs/heads/master@{#15505}
orders of the ProcessStream and ProcessReverseStream API calls.
This CL adds the ability to specify that call order in a file.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2561843003
Cr-Commit-Position: refs/heads/master@{#15503}
is the one stored in the aecdump.
During AEC development, it is handy to be able to simulate different
doubletalk scenarios. This CL adds the ability to add an
artificial nearend on top of that present in the aecdump, which
allows for the developer to artificially customize the scenario
being tested
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2562593003
Cr-Commit-Position: refs/heads/master@{#15502}
Reason for revert:
Increase in encode time larger than expected.
Original issue's description:
> Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabled.
>
> Error resilience is currently always enabled for VP9 which reduces quality.
>
> BUG=webrtc:6783
>
> Committed: https://crrev.com/4eb03c76fa2320534d669fda2aabf800e7a6f579
> Cr-Commit-Position: refs/heads/master@{#15390}
TBR=stefan@webrtc.org,marpan@webrtc.org,mflodman@webrtc.org,marpan@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6783
Review-Url: https://codereview.webrtc.org/2554403006
Cr-Commit-Position: refs/heads/master@{#15501}
Reason for revert:
Failed on the perf tests.
Original issue's description:
> Bump up scaling limit for MediaCodec.
>
> Wait until MediaCodec is better tested at these low
> resolutions, and until some fallback mechanism is in place
> before lowering this threshold.
>
> BUG=webrtc:6837
>
> Committed: https://crrev.com/3e9b1330467edf6b5af609b375c15efb9e6b4933
> Cr-Commit-Position: refs/heads/master@{#15498}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6837
Review-Url: https://codereview.webrtc.org/2562963002
Cr-Commit-Position: refs/heads/master@{#15500}
Wait until MediaCodec is better tested at these low
resolutions, and until some fallback mechanism is in place
before lowering this threshold.
BUG=webrtc:6837
Review-Url: https://codereview.webrtc.org/2566533002
Cr-Commit-Position: refs/heads/master@{#15498}
GDI capturer may randomly return a blank frame. So this change enables tests for
Linux / DirectX capturer / magnifier capturer.
BUG=webrtc:6666
Review-Url: https://codereview.webrtc.org/2559583002
Cr-Commit-Position: refs/heads/master@{#15489}
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/video_capture/*",
"//webrtc/modules/video_coding/*",
BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2555333004
Cr-Commit-Position: refs/heads/master@{#15488}
