The packet size was only used to control how often to output DTMF
packets. However, it likely did not work as intended, since that
interval was only set during initialization. No changes to the packet
size, like what AudioEncoder::Num10MsFramesInNextPacket could
indicate, were picked up. The value was instead taken from an entry in
ACMCodecDB.
Since it was not-fully-functional, its exact value didn't seem to
matter and it was getting in the way of making it possible to supply
an external audio encoder factory, I've decided to remove it
altogether. The DTMF code now uses an interval of 50 ms regardless,
which is a value recommended by the RFC.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2545753002
Cr-Commit-Position: refs/heads/master@{#15380}
QualityScaler may scale down the resolution, so our tests shouldn't
expect the input resolution to exactly match the resolution received on
the other side. Instead, we now just check that the aspect ratio
matches.
BUG=webrtc:5907
Review-Url: https://codereview.webrtc.org/2547673002
Cr-Commit-Position: refs/heads/master@{#15373}
This isn't a performance test, so it may be running in a slow
environment, and shouldn't be subject to strict timeouts.
BUG=webrtc:6801
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2539183005
Cr-Commit-Position: refs/heads/master@{#15370}
We still DCHECK for RTP, but not RTCP. RTCP packets can be sent before
offer/answer negotiation is complete, due to this bug:
https://bugs.chromium.org/p/webrtc/issues/detail?id=6809
This bug can only occur if the RTCP mux policy is "require", which is
why we started hitting it recently (the default in unit tests was
recently changed to "require").
BUG=webrtc:6776
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2542233002
Cr-Commit-Position: refs/heads/master@{#15369}
This change adds HistogramFactoryGetCountsLinear and
RTC_HISTOGRAM_COUNTS_LINEAR. Note that the default implementation of
HistogramFactoryGetCounts in metrics_default.cc also provides a
linearly spaced histogram, while the Chrome UMA implementation
provides exponentially spaced buckets.
BUG=none
Review-Url: https://codereview.webrtc.org/2548463002
Cr-Commit-Position: refs/heads/master@{#15367}
AdaptiveVideoSource is used in testing/simulations of the bandwidth estimator.
Nada's reaction to delay depends on the current bitrate and the configured max rate in a non-intuituve way. Increase the starting bitrate to compensate for the increased max bitrate. This is only used in unit tests.
BUG=webrtc:6807
# Presubmit warns about a lint error in bwe.h that's unrelated to my change. Fixing it is beyond the scope of this CL.
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2542843003
Cr-Commit-Position: refs/heads/master@{#15364}
When starting from 720p this is necessary to achieve acceptable
quality at low bitrates.
BUG=webrtc:6495
Review-Url: https://codereview.webrtc.org/2538913003
Cr-Commit-Position: refs/heads/master@{#15356}
These tests have been a little flaky on the bots.
Hopefully increasing the timeout by 200% will help.
BUG=webrtc:6799
Review-Url: https://codereview.webrtc.org/2541743006
Cr-Commit-Position: refs/heads/master@{#15355}
This CL makes a method available on the AVFoundationVideoCapturer
that adapts the output format of captured video to the specified
width and height.
BUG=webrtc:6753
Review-Url: https://codereview.webrtc.org/2528493004
Cr-Commit-Position: refs/heads/master@{#15351}
The goal with this CL is to move implementation details out from the
webrtc root (webrtc/video_decoder.h) to simplify the dependency graph.
Another goal is to streamline the creation of VideoDecoders in
webrtcvideoengine2.cc; it will now have two factories of the same
WebRtcVideoDecoderFactory type, one internal and one external.
Specifically, this CL:
* Removes webrtc::VideoDecoder::DecoderType and use webrtc::VideoCodecType
instead.
* Removes 'static VideoDecoder* Create(DecoderType codec_type)' and
moves the create function to the internal decoder factory instead.
* Removes video_decoder.cc. webrtc::VideoDecoder is now just an
interface without any static functions.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2521203002
Cr-Commit-Position: refs/heads/master@{#15350}
This change makes the Java classes and constructors for
SurfaceTextureHelper and I420Frame public. This allows applications to
use the WebRTC CameraVideoCapturer to obtain raw frames, and to render
frames on the WebRTC VideoRenderer without having to pass them through a
VideoTrack - such as when using Quartc.
BUG=None.
Review-Url: https://codereview.webrtc.org/2544563002
Cr-Commit-Position: refs/heads/master@{#15344}
This is to assess the performance penalty of the (current)
lack of integration with FlexFEC and BWE.
This CL also enables send-side BWE for the following tests:
- foreman_cif_net_delay_0_0_plr_0_VP9
- foreman_cif_net_delay_0_0_plr_0_H264
- foreman_cif_delay_50_0_plr_5_VP9
- foreman_cif_delay_50_0_plr_5_H264
- foreman_cif_delay_50_0_plr_5_H264_flexfec
- foreman_cif_delay_50_0_plr_5_H264_ulpfec
Perf alerts on these tests are therefore expected.
R=stefan@webrtc.org
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2534203004
Cr-Commit-Position: refs/heads/master@{#15339}
It is unused since cl https://codereview.webrtc.org/2386573002.
The new denoiser implementation and its tests are kept for now. This
code is also unused, but there are some plans to take this code into
use in the not too distant future.
BUG=None
Review-Url: https://codereview.webrtc.org/2496153002
Cr-Commit-Position: refs/heads/master@{#15338}
We already have an implementation in h264_common. We should have
as few of these as possible as they are subtly hard to get right
and it creates work to maintain N implementations.
BUG=webrtc:6546
Review-Url: https://codereview.webrtc.org/2538133002
Cr-Commit-Position: refs/heads/master@{#15336}
These tests have been causing a large number of false alerts, which is
tons of work for the perf sheriff. It's infeasible to test every single
permutation, so we must choose carefully. This CL reduces the number
of RESULT lines from the test from ~450 to ~50. I attempted to choose
interesting permutations, but you probably know better what's
interesting...
https://chromeperf.appspot.com/report?
sid=a7193c96f708018848ca07ad9c78ac657cadab3c70b3939c42bd7d70a092d61a
also suggests most of the metrics have enormous standard deviations,
so maybe you could look into how stable the metrics really are
and remove/stabilize the ones that aren't?
BUG=chromium:666725
Review-Url: https://codereview.webrtc.org/2529393006
Cr-Commit-Position: refs/heads/master@{#15329}
And delete the method CongestionController::packet_router.
BUG=None
Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
RTCIceCandidatePairStats.transport_id is set to the related
RTCTransportStats' id.
Unittest for RTCIceCandidatePairStats is updated to do EXPECT_EQ
between actual and an expected hardcoded dictionary. The previous way of
testing, ExpectReportContainsCandidatePair, is removed.
(ExpectReportContainsCandidate still exist, we might want to replace
this by EXPECT_EQ testing in a follow up.)
Unittest for RTCTransportStats is similarly updated and
ExpectReportContainsTransportStats is removed. A bug was uncovered where
the "rtcp_connection_info.best_connection = true" case was not tested
(a copy of rtcp_connection_info was used in the test, modifying that had
no affect on the test) - fixed.
rtcstats_integrationtest.cc updated to take transport_id into account.
In order to reuse an updated version of expected_rt[c]p_transport in the
unittest, timestamps are ignored by RTCStats::operator==.
BUG=chromium:627816, chromium:653873, chromium:653873, webrtc:6755
Review-Url: https://codereview.webrtc.org/2527113002
Cr-Commit-Position: refs/heads/master@{#15316}
JitterBufferDelayInMs is used for the WebRTC-NewVideoJitterBuffer finch
experiment, and therefore needs to be calculated.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2534093003
Cr-Commit-Position: refs/heads/master@{#15313}
