Matches the include order in webrtc/base/criticalsection.h and makes use
of winsock2.h instead of winsock.h for consistency.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1407053008
Cr-Commit-Position: refs/heads/master@{#10389}
Before this change, UpdateEstimate would repeatedly decrease bitrate
even though there's no fresh corresponding RTCP loss report, triggering
multiple reactions to a single indication of high packet loss.
BUG=webrtc:5101
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1417723005
Cr-Commit-Position: refs/heads/master@{#10374}
Xvfb is needed for the screen capture tests in modules_unittests,
which also brings in xdisplaycheck used by testing/xvfb.py.
libjingle_media_unittest was missing a resource video in the .isolate
file.
BUG=chromium:497757
R=stip@chromium.org
Review URL: https://codereview.webrtc.org/1415603005 .
Cr-Commit-Position: refs/heads/master@{#10365}
We don't allow more than one retransmission within one RTT, but the RTT
estimate might be off. Reasonably, the remote end will not send a NACK
until the packet after has been received - so always resend on first
request.
Review URL: https://codereview.webrtc.org/1414563003
Cr-Commit-Position: refs/heads/master@{#10362}
Reports show that we see full echo from the OnePlus 2 device.
Disabling hardware effects and revert to WebRTC-based
components instead as a test to see if it helps.
R=tommi@webrtc.org
TBR=tommi
BUG=b/25096456
Review URL: https://codereview.webrtc.org/1417093002 .
Cr-Commit-Position: refs/heads/master@{#10357}
This patch also also ensures that audio is restored after an incoming
GSM call.
BUG=webrtc:5058, webrtc:5012
TEST=Manual tests using modified AppRTCDemo and three different BT headsets
Review URL: https://codereview.webrtc.org/1401963002
Cr-Commit-Position: refs/heads/master@{#10354}
Some toolchains (in this case referring to a g++ 4.9, or "arm-linux-
androideabi-g++ (GCC) 4.9 20140827 (prerelease)" according to my
--version, from the Android NDK r10e-rc4 and potentially with custom
patches; others may be affected as well) fail to prove that myVec in
WebRtcIsac_CorrelateInterVec is never used uninitialized. This is likely
due to the compiler thinking the assignment in line 468 might not
happen. Changing the loop condition in line 466 to rowCntr <
SOME_CONSTANT also helps, suggesting that the compiler can't infer that
there are only 2 values interVecDim can have at that point, and neither
of them are 0. Of course, this is not an acceptable fix, as it changes
behaviour.
This seems to be a compiler bug, or at least an issue with its
heuristics. However, we can't really change toolchains at the moment,
and ultimately this change improves support for certain older compilers.
BUG=
Review URL: https://codereview.webrtc.org/1406423004
Cr-Commit-Position: refs/heads/master@{#10337}
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"
If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"
BUG=
Review URL: https://codereview.webrtc.org/1311533012
Cr-Commit-Position: refs/heads/master@{#10333}
- "WebRTC.Video.QualityLimitedResolutionInPercent"
and if a frame is downscaled, the average number of times the frame is downscaled:
- "WebRTC.Video.QualityLimitedResolutionDownscales"
BUG=
Review URL: https://codereview.webrtc.org/1325153009
Cr-Commit-Position: refs/heads/master@{#10319}
External consumers may have a dependency on the old name, so this will give them the opportunity to switch over.
BUG=
Review URL: https://codereview.webrtc.org/1414543002
Cr-Commit-Position: refs/heads/master@{#10310}
Sounds better according to a MUSHRA listening test.
The computational complexity is unaffected.
An empirically estimated gain was added to compensate for the attenuation introduced by the algorithm.
There are some TODOs, which I will address in follow up CLs.
It was tested in Hangouts without headphones and highest volume, to make sure it doesn't affect the AEC.
Review URL: https://codereview.webrtc.org/1378973003
Cr-Commit-Position: refs/heads/master@{#10308}
This removes the TRFC rate control which does not introduce any help in the
computation of the sending rate.
BUG=5083
Review URL: https://codereview.webrtc.org/1383813003
Cr-Commit-Position: refs/heads/master@{#10299}
Implements SupportsNativeHandle() in SimulcastEncoderAdapter which works
when there's only a single encoder.
BUG=webrtc:5060
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1397653004
Cr-Commit-Position: refs/heads/master@{#10291}
Encoders need to be externally provided. To use software encoders they
need to be created and registered from the outside.
BUG=webrtc:1695
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1394823002 .
Cr-Commit-Position: refs/heads/master@{#10283}
Also updated the RTPPayloadRegistry::RestoreOriginalPacket signature to not take the first arg as a **, since it isn't modified.
Review URL: https://codereview.webrtc.org/1394573004
Cr-Commit-Position: refs/heads/master@{#10276}
This is no longer used. Related code in the iSAC codec itself will be
deleted a follow-up CL.
BUG=4210
Review URL: https://codereview.webrtc.org/1404463003
Cr-Commit-Position: refs/heads/master@{#10272}
Due to https://codereview.chromium.org/1397493004 we're now adding
a build_overrides directory in WebRTC. Thanks to this, we no longer
need to pass --args="build_with_chromium=false" when running GN in
standalone WebRTC.
Change log: c089d37..159828f
Full diff: c089d37..159828f
No dependencies changed.
No update to Clang.
BUG=webrtc:5070,chromium:541791
TBR=tommi@webrtc.org
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal
Review URL: https://codereview.webrtc.org/1403453003 .
Cr-Commit-Position: refs/heads/master@{#10270}
This is no longer used. Related code in NetEq and the iSAC codec itself
will be deleted in follow-up CLs.
BUG=4210
Review URL: https://codereview.webrtc.org/1404623002
Cr-Commit-Position: refs/heads/master@{#10264}
This CL makes AddRef() and Release() const member methods and the refcount integer mutable. This is reasonable, because they only manage the lifetime of the object, and this is also how it's done in Chromium.
The purpose is to be able to capture a const pointer in a scoped_refptr, which is currenty impossible. The practial problem this CL solves is this:
void Foo::Bar() const {}
rtc::Callback0<void> Foo::MakeClosure() const {
return rtc::Bind(&Foo::Bar, this);
}
We currently capture |this| as const Foo*. With this CL, |this| will be captured as scoped_refptr<const Foo>.
A test is also added in bind_unittest to check this behaviour.
BUG=webrtc:5065
R=perkj@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1403683004 .
Cr-Commit-Position: refs/heads/master@{#10253}
Reason for revert:
Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works.
Original issue's description:
> Adding support for simulcast and spatial layers into VideoQualityTest
>
> The CL includes several changes:
> - Adding flags describing the streams and spatial layers.
> - Reorganizing the order of the flags, to make them easier to maintain.
> - Adding a member .params_ to VideoQualityAnalyzer.
> (instead of passing it to every member function manually)
> - Updating VideoAnalyzer to support simulcast.
> (select appropriate ssrc and fix timestamps which are sometimes increased by 1)
> - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
> Changing to first read bitrates and resolution ratios from the flags, if specified.
> If not specified, reverting to the old code are setting the values automatically.
> - Changing the parameters in LayerFilteringTransport, replacing
> xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
>
> Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e
> Cr-Commit-Position: refs/heads/master@{#10215}
TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1397363002
Cr-Commit-Position: refs/heads/master@{#10252}
This CL restructures the RtcEventLog protobuf format, by removing the DebugEvent message. This is done by moving the LOG_START and LOG_END events to the EventType enum and making a seperate message for audio playout events. In addition to these changes, some fields were added to the AudioReceiveConfig and AudioSendConfig messages, but these are for future use and are not currently logged yet.
This is a follow-up to CL 1340283002 which adds a SSRC to AudioPlayout events in the RtcEventLog.
BUG=webrtc:4741
R=henrik.lundin@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/1348113003 .
Cr-Commit-Position: refs/heads/master@{#10221}
The CL includes several changes:
- Adding flags describing the streams and spatial layers.
- Reorganizing the order of the flags, to make them easier to maintain.
- Adding a member .params_ to VideoQualityAnalyzer.
(instead of passing it to every member function manually)
- Updating VideoAnalyzer to support simulcast.
(select appropriate ssrc and fix timestamps which are sometimes increased by 1)
- VP9EncoderImpl already had code for automatic calculation of bitrate for each layer.
Changing to first read bitrates and resolution ratios from the flags, if specified.
If not specified, reverting to the old code are setting the values automatically.
- Changing the parameters in LayerFilteringTransport, replacing
xx_discard_thresholds with selected_xx, to make it easier to use for the end user.
Review URL: https://codereview.webrtc.org/1353263005
Cr-Commit-Position: refs/heads/master@{#10215}
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.
BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1392513002 .
Cr-Commit-Position: refs/heads/master@{#10211}
In video_sender.cc, properly read the number of temporal layers for VP9 too.
Also, some cleanup in video_loopback.cc and video_quality_test.h.
Review URL: https://codereview.webrtc.org/1351693005
Cr-Commit-Position: refs/heads/master@{#10201}
When fetching a packet from the rtp packet history, cuased by a
retransmission, the transport seq extension header is enabled but the
sequence number is set to 0. A new transport seq should be assigned in
this case.
BUG=
Review URL: https://codereview.webrtc.org/1385563005
Cr-Commit-Position: refs/heads/master@{#10183}
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1369923005 .
Cr-Commit-Position: refs/heads/master@{#10181}
