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 This test is no
longer valid on android, as the failure is expected behaviour.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}
For example, zero rtt may be reported to:
BitrateControllerImpl::OnReceivedRtcpReceiverReport:
- SendSideBandwidthEstimation::UpdateReceiverBlock
- SendSideBandwidthEstimation::UpdateUmaStats
BitrateAllocator::OnNetworkChanged:
- ProtectionBitrateCalculator::SetTargetRates
Re-add check that was removed in https://codereview.webrtc.org/2422063002.
BUG=webrtc:6692
Review-Url: https://codereview.webrtc.org/2552883010
Cr-Commit-Position: refs/heads/master@{#15486}
Reason for revert:
There was a bug in the implementation where the adapter could get stuck at really low resolutions. That has now been fixed.
Original issue's description:
> Revert of Add ability to scale to arbitrary factors (patchset #7 id:120001 of https://codereview.webrtc.org/2555483005/ )
>
> Reason for revert:
> Issue discovered with scaling back up.
>
> Original issue's description:
> > Add ability to scale to arbitrary factors
> >
> > This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
> >
> > BUG=webrtc:6837
> >
> > Committed: https://crrev.com/710c335d785b104bda4a912bd7909e4d27f9b04f
> > Cr-Commit-Position: refs/heads/master@{#15469}
>
> 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/7722a4cc8d31e5e924e9e6c5c97412ce8bbbe59d
> Cr-Commit-Position: refs/heads/master@{#15470}
R=magjed@webrtc.org
BUG=webrtc:6837,webrtc:6848
Review-Url: https://codereview.webrtc.org/2558243003
Cr-Commit-Position: refs/heads/master@{#15485}
Also deletes one call to CongestionController::pacer.
BUG=None
Review-Url: https://codereview.webrtc.org/2542113003
Cr-Commit-Position: refs/heads/master@{#15479}
This moves some GN check configurations out of .gn to individual targets.
The now checked target is:
"//webrtc/modules/audio_processing/*",
BUG=webrtc:6828
NOTRY=True
R=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2558813003
Cr-Commit-Position: refs/heads/master@{#15475}
This CL breaks out the output sample rate calculation from
webrtc::AudioMixerImpl. A new OutputRateCalculator interface is added
to make the sample rate configurable. There are at least three reasons
for this change:
1. The mixer will be used for an internal project, in which no
resampling is done after the mixing. There the sample rate should
be static. Currently, it can differ across mix iterations and
depends on the number of audio sources. If there are no sources,
the WebRTC mixer behavior is to produce silence at 48 kHz.
2. A planned change to WebRTC will make audio processing steps
happen at constant sample rates. A configurable sample rate
calculator will make the transition simpler for the mixer.
3. The current mixer design is a single large file. Behavior is not
always simple to change (e.g. as in this case to mix at a
constant rate), unrelated behavior can be broken, reusing the
mixer in internal projects is tricky. Using DI for the sample
rate calculation solves parts of these issues.
Changes:
The protected mixer c-tor now takes
unique_ptr<OutputRateCalculator>. The current output rate calculation
is moved to DefaultOutputRateCalculator. A new factory method
AudioMixerImpl::CreateWithOutputRateCalculator is added. The old
factory method passes the default rate calculator.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2557713006
Cr-Commit-Position: refs/heads/master@{#15472}
Reason for revert:
Issue discovered with scaling back up.
Original issue's description:
> Add ability to scale to arbitrary factors
>
> This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
>
> BUG=webrtc:6837
>
> Committed: https://crrev.com/710c335d785b104bda4a912bd7909e4d27f9b04f
> Cr-Commit-Position: refs/heads/master@{#15469}
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/2557323002
Cr-Commit-Position: refs/heads/master@{#15470}
This CL adds a fallback for the case when no optimized scale factor produces a low enough resolution for what was requested. It also ensures that all resolutions provided by the video adapter are divisible by four. This is required by some hardware implementations.
BUG=webrtc:6837
Review-Url: https://codereview.webrtc.org/2555483005
Cr-Commit-Position: refs/heads/master@{#15469}
This makes it obvious that cricket::Codec should not be
instantiated; only subclasses should be instantiated.
BUG=none
Review-Url: https://codereview.webrtc.org/2546363002
Cr-Commit-Position: refs/heads/master@{#15468}
It should be 'JNIEXPORT rettype JNICALL' not 'rettype JNIEXPORT JNICALL'.
BUG=webrtc:6660
Review-Url: https://codereview.webrtc.org/2557793003
Cr-Commit-Position: refs/heads/master@{#15467}
BitBlt returns a BOOL value, which should be taken care in ScreenCapturerWinGdi.
Meanwhile, this change also replaces assert() / abort() with RTC_DCHECK() /
RTC_CHECK() / RTC_NOTREACHED().
This change cannot fix the bug, the reason of the issue is still unknown, but it
is still the right thing to do.
In ScreenCapturerIntegrationTest, each frame will be captured at most 600 times.
Since the test case fails, which means the ScreenCapturerWinGdi consistently
returns a white frame for 600 times under a certain state. With this change,
instead of returning white frame, ScreenCapturerWinGdi will return a temporary
error. But I do not think a ScreenCapturerWinGdi can automatically recover by
retrying.
BUG=webrtc:6843
Review-Url: https://codereview.webrtc.org/2553353002
Cr-Commit-Position: refs/heads/master@{#15465}
Create the RtpReceiver.Observer which is a Java wrapper over the webrtc::RtpReceiverObserverInterface.
The callback function onFirstPacketReceived will be called whenever the first audio or video packet it received.
BUG=webrtc:6742
Review-Url: https://codereview.webrtc.org/2531333003
Cr-Commit-Position: refs/heads/master@{#15464}
