This is the first CL in a series of major cleanup and dependency
corrections needed in order to satisfy 'gn check'.
BUG=webrtc:4243, webrtc:5589
NOTRY=True
Review-Url: https://codereview.webrtc.org/1990593002
Cr-Commit-Position: refs/heads/master@{#12790}
This allows connecting between clients without using external servers, which is useful to OEMs if they are working in a network without internet connection. Implementation uses custom AppRTCClient that replaces WebSocketRTCClient if roomId looks like an IP. Instead of a web socket, this class uses direct TCP connection between peers as a signaling channel.
Review-Url: https://codereview.webrtc.org/1963053002
Cr-Commit-Position: refs/heads/master@{#12789}
With this change, when max-bundle and rtcp-mux are both enabled, we no
longer create and destroy a temporary transport channel when a media
channel gets added. Instead, the media channel uses the correct bundled
transport channel from the start.
This fixes a bug where adding a media type would cause the ICE state to
briefly become Disconnected and then immediately recover. The temporary
channel was created in a non-writable state, which caused the
TransportController to declare the ICE state to be Disconnected (as not
all transport channels were writable). Right after creation, the
temporary channel was then destroyed and the ICE state went back to the
correct one.
BUG=webrtc:5856
Review-Url: https://codereview.webrtc.org/1972493002
Cr-Commit-Position: refs/heads/master@{#12781}
Renaming the ParticipantFramePair to ParticipantFrameStruct. The muted
variable is not yet used.
BUG=webrtc:5609
Review-Url: https://codereview.webrtc.org/1981243002
Cr-Commit-Position: refs/heads/master@{#12780}
Deleted the temporary ACM method without the muted parameter, and had
to modify several tests for this. The muted parameter is not yet propagated to the AudioConferenceMixer; this is the next step.
BUG=webrtc:5609
TBR=perkj@webrtc.org
Review-Url: https://codereview.webrtc.org/1985743002
Cr-Commit-Position: refs/heads/master@{#12779}
Chrome does not detect NEON instruction set at runtime in WebRTC code starting
with M50, which is now in Stable. Remove support for runtime detection for
simplicity.
The only remaining piece of Chrome that will continue to depend on runtime
detection is /net, where devices with _broken_ neon support are also detected,
and it is not configurable via GYP/GN.
BUG=522035
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/1955413003
Cr-Commit-Position: refs/heads/master@{#12778}
MJPEG capture is not used on Android. Therefore, disable jpeg support to
reduce libjingle_peerconnection_so file size by removing dependency to
libjpeg_turbo.
Also, remove unused build_libjpeg and rtc_build_libjpeg variables.
Review-Url: https://codereview.webrtc.org/1978243002
Cr-Commit-Position: refs/heads/master@{#12777}
We're now ready https://codereview.webrtc.org/1984503002/ downstream,
so make sure we can enable libevent but still choose which libevent
implementation to use. This follows the common pattern where an enable_
flag controls whether we should use the feature at all, whereas build_
controls if we should use the dependency from our DEPS file or
something else.
NOTRY=True
Review-Url: https://codereview.webrtc.org/1980003002
Cr-Commit-Position: refs/heads/master@{#12772}
This is a follow-up to https://codereview.webrtc.org/1965313002/ which
was TBR-landed.
Minor code clean-up/corrections:
Property nativeConfiguration -> - method createNativeConfiguration.
RTCLogWarning -> RTCLogError.
setConfiguration returning NO instead of false.
initWithFactory returning nil instead of nullptr.
Braces around ifs.
Review-Url: https://codereview.webrtc.org/1978233002
Cr-Commit-Position: refs/heads/master@{#12770}
This allows creating tests for AppRTC Android demo that will be run on
the host machine instead of a device. These tests can mock Android APIs
through Robolectric. Because the tests are run on the host machine,
they run much faster.
BUG=webrtc:5896
NOTRY=True
Review-Url: https://codereview.webrtc.org/1985663002
Cr-Commit-Position: refs/heads/master@{#12769}
Needed to avoid DrMemory warnings, if the frame is passed to libyuv
AVX assembly functions.
BUG=libyuv:377
Review-Url: https://codereview.webrtc.org/1985693002
Cr-Commit-Position: refs/heads/master@{#12765}
Reason for revert:
This will take longer time for the RTT to converge.
Need to update the RTT calculation algorithm if doing this.
Original issue's description:
> Increase the stun ping interval.
>
> Writable connections are pinged at a slower rate.
> The function IsPingable will filter out the writable connections.
> The interval for slower ping rate by default is WRITABLE_CONNECTION_PING_INTERVAL(2500ms) and can be set with the configuration.
>
> BUG=webrtc:1161
>
> Committed: https://crrev.com/8f7a5aad55a64f0d81b6436a22ffbdfcdcde91e0
> Cr-Commit-Position: refs/heads/master@{#12736}
TBR=honghaiz@webrtc.org,pthatcher@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:1161
Review URL: https://codereview.webrtc.org/1982713002 .
Cr-Commit-Position: refs/heads/master@{#12762}
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.
R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1917193008 .
Cr-Commit-Position: refs/heads/master@{#12761}
New task queueing primitive for async tasks: TaskQueue.
TaskQueue is a new way to asynchronously execute tasks sequentially
in a thread safe manner with minimal locking. The implementation
uses OS supported APIs to do this that are compatible with async IO
notifications from things like sockets and files.
This class is a part of rtc_base_approved, so can be used by both
the webrtc and libjingle parts of the WebRTC library. Moving forward,
we can replace rtc::Thread and webrtc::ProcessThread with this implementation.
NOTE: It should not be assumed that all tasks that execute on a TaskQueue,
run on the same thread. E.g. on Mac and iOS, we use GCD dispatch queues
which means that tasks might execute on different threads depending on
what's the most efficient thing to do.
TBR=perkj@webrtc.org,phoglund@webrtc.org
Review-Url: https://codereview.webrtc.org/1984503002
Cr-Commit-Position: refs/heads/master@{#12749}
Reason for revert:
Speculative revert to see if failures on the DrMemory bot are related to this cl. See e.g. here:
https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/4243
UNINITIALIZED READ: reading 0x04980040-0x04980060 32 byte(s) within 0x04980040-0x04980060
# 0 CopyRow_AVX
# 1 CopyPlane
# 2 I420Copy
# 3 webrtc::ExtractBuffer
# 4 cricket::WebRtcVideoCapturer::SignalFrameCapturedOnStartThread
# 5 cricket::WebRtcVideoCapturer::OnIncomingCapturedFrame
# 6 FakeWebRtcVideoCaptureModule::SendFrame
# 7 WebRtcVideoCapturerTest_TestCaptureVcm_Test::TestBody
# 8 testing::internal::HandleSehExceptionsInMethodIfSupported<>
Original issue's description:
> Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1935443002/ )
>
> Reason for revert:
> I plan to reland this change in a week or two, after downstream users are updated.
>
> Original issue's description:
> > Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
> >
> > Reason for revert:
> > Breaks chrome FYI bots.
> >
> > Original issue's description:
> > > Delete webrtc::VideoFrame methods buffer and stride.
> > >
> > > To make the HasOneRef/IsMutable hack work, also had to change the
> > > video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> > > to not imply an AddRef.
> > >
> > > BUG=webrtc:5682
> >
> > TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/5b3c443d301f2c2f18dac5b02652c08b91ea3828
> > Cr-Commit-Position: refs/heads/master@{#12558}
>
> TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5682
>
> Committed: https://crrev.com/d0dc66e0ea30c8614001e425a4ae0aa7dd56c2a7
> Cr-Commit-Position: refs/heads/master@{#12721}
TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,stefan@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/1983583002
Cr-Commit-Position: refs/heads/master@{#12745}
Remove ViEEncoder::SetNetworkStatus.
Original cl description:
This cl removed ViEEncoder::SetNetworkStatus. Instead the PacedSender will report that frames can not be sent when the network is down and the BitrateController will report an estimated available bandwidth of 0 bps.
Patchset #1 is a pure reland.
Patchset #2 change the bitrate allocator to always return an initial bitrate >0 regardless of the estimates. The observer will be notified though if the network is down.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/1972183004
Cr-Commit-Position: refs/heads/master@{#12743}
