outputs various elements of the RTP header when there is a delivery error.
output example:
Packet len=984 pt=100 seq=47914 ts=1532364329 ssrc=0xdeadbef0
BUG=webrtc:6991
Review-Url: https://codereview.webrtc.org/2621163006
Cr-Commit-Position: refs/heads/master@{#16294}
Reason for revert:
Bugfixes related to the new jitter buffer has landed.
Original issue's description:
> Revert of Make the new jitter buffer the default jitter buffer. (patchset #2 id:230001 of https://codereview.webrtc.org/2642753002/ )
>
> Reason for revert:
> Breaks tests downstream.
>
> Original issue's description:
> > Reland of Make the new jitter buffer the default jitter buffer. (patchset #1 id:1 of https://codereview.chromium.org/2632123005/ )
> >
> > Reason for revert:
> > Fix in this CL: https://codereview.chromium.org/2640793003/
> >
> > Original issue's description:
> > > Revert of Make the new jitter buffer the default jitter buffer. (patchset #7 id:120001 of https://codereview.chromium.org/2627463004/ )
> > >
> > > Reason for revert:
> > > Breaks android bots.
> > >
> > > Original issue's description:
> > > > Make the new jitter buffer the default jitter buffer.
> > > >
> > > > This CL contains only the changes necessary to make the switch to the new jitter
> > > > buffer, clean up will be done in follow up CLs.
> > > >
> > > > In this CL:
> > > > - Removed the WebRTC-NewVideoJitterBuffer experiment and made the
> > > > new video jitter buffer the default one.
> > > > - Moved WebRTC.Video.KeyFramesReceivedInPermille and
> > > > WebRTC.Video.JitterBufferDelayInMs to the ReceiveStatisticsProxy.
> > > >
> > > > BUG=webrtc:5514
> > > >
> > > > Review-Url: https://codereview.webrtc.org/2627463004
> > > > Cr-Commit-Position: refs/heads/master@{#16114}
> > > > Committed: 0f0763d86d
> > >
> > > TBR=stefan@webrtc.org,terelius@webrtc.org
> > > # Skipping CQ checks because original CL landed less than 1 days ago.
> > > NOPRESUBMIT=true
> > > NOTREECHECKS=true
> > > NOTRY=true
> > > BUG=webrtc:5514
> > >
> > > Review-Url: https://codereview.webrtc.org/2632123005
> > > Cr-Commit-Position: refs/heads/master@{#16117}
> > > Committed: c08c191f7d
> >
> > TBR=stefan@webrtc.org,terelius@webrtc.org
> > # Not skipping CQ checks because original CL landed more than 1 days ago.
> > BUG=webrtc:5514
> >
> > Review-Url: https://codereview.webrtc.org/2642753002
> > Cr-Commit-Position: refs/heads/master@{#16149}
> > Committed: f20dd0014d
>
> TBR=stefan@webrtc.org,terelius@webrtc.org,philipel@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5514
>
> Review-Url: https://codereview.webrtc.org/2638423003
> Cr-Commit-Position: refs/heads/master@{#16159}
> Committed: 04926b8264TBR=stefan@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org,kjellander@google.com
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2652043005
Cr-Commit-Position: refs/heads/master@{#16293}
Reason for revert:
This was false alarm. The next Chromium roll fixed the problem and the bot has been green for 3 builds.
Original issue's description:
> Disabled two iOS tests due to bot breakage. Affected tests are
> AudioDeviceTest.RunPlayoutWithFileAsSource and
> AudioDeviceTest.StartStopRecording
>
> NOTRY=True
> TBR=henrika@webrtc.org
> BUG=7056
>
> Review-Url: https://codereview.webrtc.org/2652423002
> Cr-Commit-Position: refs/heads/master@{#16286}
> Committed: 8e775e16ebTBR=henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=7056
Review-Url: https://codereview.webrtc.org/2656533005
Cr-Commit-Position: refs/heads/master@{#16289}
Removed conditional disabling of
ReceivesFlexfecAndSendsCorrespondingRtcp on Asan, since failure occurs
at other platforms as well.
BUG=webrtc:7050
TBR=holmer@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2651673011
Cr-Commit-Position: refs/heads/master@{#16288}
WebRTC standalone build may depend on "video_capture_internal_impl"
instead of "video_capture". Including "video_capture" in public_deps
leads to duplicated definition in this case.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2657783002
Cr-Commit-Position: refs/heads/master@{#16279}
Order as given by inheritance in class definition.
No functional changes are intended with this CL.
BUG=None
Review-Url: https://codereview.webrtc.org/2646343005
Cr-Commit-Position: refs/heads/master@{#16272}
We are moving the whole content of "webrtc/build" to new
locations (see crbug.com/611808 for further information).
The new location for the "webrtc/build/android" stuff is
"tools-webrtc/android".
BUG=webrtc:7030
Review-Url: https://codereview.webrtc.org/2647343006
Cr-Commit-Position: refs/heads/master@{#16269}
due to timeout-caused build failure (see bugs.webrtc.org/7047). The
timeout is governed by CallTest::kDefaultTimeoutMs, which is set to 30
seconds. This can be too low for Asan.
TBR=brandtr@webrtc.org
BUG=webrtc:7047
Review-Url: https://codereview.webrtc.org/2657823003
Cr-Commit-Position: refs/heads/master@{#16267}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
We have to move the content of the 'webrtc/build/' directory up of one level (see crbug.com/611808 for further details).
To avoid a collision with the DEPSed build/ at the top level we are moving all the content of 'webrtc/build/' to different locations (see webrtc:7030 for the suggested locations).
BUG=webrtc:7030
NOTRY=True
Review-Url: https://codereview.webrtc.org/2651913002
Cr-Commit-Position: refs/heads/master@{#16259}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
This CL introduces a dedicated unit test for webrtc::RtpStreamReceiver.
Focus of this CL is testing RtpStreamReciver::OnReceivedPayloadData().
Dependencies with virtual interfaces are (g)mocked, non-virtual
dependencies are instantiated.
This CL is chained to https://codereview.webrtc.org/2638933002/ .
BUG=webrtc:5948
Review-Url: https://codereview.webrtc.org/2641463002
Cr-Commit-Position: refs/heads/master@{#16240}
It was only assigned at construction, and this improves consistency
with remote_estimator_proxy_.
The declaration of the private WrappingBitrateEstimator had to be
moved to the header file, and it was also converted from
system_wrappers' CriticalSectionWrapper to rtc::CriticalSection.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2642363003
Cr-Commit-Position: refs/heads/master@{#16236}
They were passed on via RtpRtcp::Configuration, but unused for a
receive only RtpRtcp module.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2639423007
Cr-Commit-Position: refs/heads/master@{#16234}
I've tested both the old and the new delay estimators and they all work ok at 15 kbps. The new estimators are a bit slower to adapt down so the maximum delay is a bit higher, especially at lower bitrates.
None of the estimators work at 10 kbps, but that is likely because the bitrate controller is configured to never go below 10 kbps. This means that it's impossible to empty the queues after a capacity drop to 10kbps (regardless of what the estimators do), so the delay stays high until the capacity increases.
BUG=webrtc:7022
Review-Url: https://codereview.webrtc.org/2644463009
Cr-Commit-Position: refs/heads/master@{#16233}
