Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.
Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1917043005
Cr-Commit-Position: refs/heads/master@{#12509}
Framerate-reduction code is disabled on all platforms, and this code
adds complexity. It's necessary to react fast, especially on mobile
platforms or other bad network conditions and framerate reduction adds
another step between HD and QVGA.
BUG=webrtc:5678, webrtc:5830
R=jackychen@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1885893002 .
Cr-Commit-Position: refs/heads/master@{#12503}
This is a first CL in a row of CLs to move everything related to
receiving RTP from ViEChanel to ViEReceiver, rename the classes and
move ViEReceiver ownership to VideoReceiveStream.
Review URL: https://codereview.webrtc.org/1912133002
Cr-Commit-Position: refs/heads/master@{#12486}
They should match the parameters used in the FullStack quality tests, so
that it is easier to visualize what we are actually testing.
BUG=
Review URL: https://codereview.webrtc.org/1871543002
Cr-Commit-Position: refs/heads/master@{#12466}
ViEEncoder doesn't need a full VideoCodingModule since it only uses the
sender side either way.
BUG=webrtc:3608,webrtc:5687
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1904983002 .
Cr-Commit-Position: refs/heads/master@{#12456}
Reason for revert:
RTCVideoEncoder has been updated to not make assumptions on calling threads/post back to a worker thread. This should now be landable again.
Original issue's description:
> Revert of Initialize/configure video encoders asychronously. (patchset #4 id:60001 of https://codereview.webrtc.org/1757313002/ )
>
> Reason for revert:
> Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.
>
> Original issue's description:
> > Initialize/configure video encoders asychronously.
> >
> > Greatly speeds up setRemoteDescription() by moving encoder initialization
> > off the main worker thread, which is free to move onto gathering ICE
> > candidates and other tasks while InitEncode() is performed. It also
> > un-blocks PeerConnection GetStats() which is no longer blocked on
> > encoder initialization.
> >
> > BUG=webrtc:5410
> > R=stefan@webrtc.org
> >
> > Committed: fb647a67be
>
> R=stefan@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:595274, chromium:595308, webrtc:5410
>
> Committed: https://crrev.com/81cbd924447d507559dbd6e6d1f9fe439fcf2716
> Cr-Commit-Position: refs/heads/master@{#12086}
TBR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410
Review URL: https://codereview.webrtc.org/1896413002
Cr-Commit-Position: refs/heads/master@{#12446}
Reason for revert:
A fix is being prepared downstream so this can now go in.
Original issue's description:
> Revert of Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead. (patchset #5 id:80001 of https://codereview.webrtc.org/1897233002/ )
>
> Reason for revert:
> API changes broke downstream.
>
> Original issue's description:
> > Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> > EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> > EncodedImageCallback can of course be cleaned up in the future.
> >
> > This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
> >
> > BUG=webrtc::5687
> >
> > Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> > Cr-Commit-Position: refs/heads/master@{#12436}
>
> TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5687
>
> Committed: https://crrev.com/a261e6136655af33f283eda8e60a6dd93dd746a4
> Cr-Commit-Position: refs/heads/master@{#12441}
TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687
Review URL: https://codereview.webrtc.org/1905583002
Cr-Commit-Position: refs/heads/master@{#12442}
Reason for revert:
API changes broke downstream.
Original issue's description:
> Deprecate VCMPacketizationCallback::SendData and use EncodedImageCallback instead.
> EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
> EncodedImageCallback can of course be cleaned up in the future.
>
> This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
>
> BUG=webrtc::5687
>
> Committed: https://crrev.com/f5d55aaecdc39e9cc66eb6e87614f04afe28f6eb
> Cr-Commit-Position: refs/heads/master@{#12436}
TBR=stefan@webrtc.org,pbos@webrtc.org,perkj@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc::5687
Review URL: https://codereview.webrtc.org/1903193002
Cr-Commit-Position: refs/heads/master@{#12441}
EncodedImageCallback is used by all encoder implementations and seems to be what we should try to use in the transport.
EncodedImageCallback can of course be cleaned up in the future.
This moves creation of RTPVideoHeader from the GenericEncoder to the PayLoadRouter.
BUG=webrtc::5687
Review URL: https://codereview.webrtc.org/1897233002
Cr-Commit-Position: refs/heads/master@{#12436}
This fix a potential race where the rotation information of a sent frame does not match the encoded frame.
BUG=webrtc:5783
TEST= Run ApprtcDemo on IOs and Android with and without capture to texture and both VP8 and H264.
R=magjed@webrtc.org, pbos@webrtc.org, tkchin@webrtc.org
TBR=tkchin_webrtc // For IOS changes.
Review URL: https://codereview.webrtc.org/1886113003 .
Cr-Commit-Position: refs/heads/master@{#12426}
Used only by tests. Deleted the EndToEndTest.UsesFrameCallbacks, which
modified pixel data. Change callback from in EndToEndTest.GetStats to call SleepMs, rath than
modifying the timestamp.
BUG=
Review URL: https://codereview.webrtc.org/1891733002
Cr-Commit-Position: refs/heads/master@{#12406}
Reason for revert:
The delay stats are high.
Original issue's description:
> Update histogram "WebRTC.Video.OnewayDelayInMs" to use the estimated one-way delay.
> Previous logged delay was: network delay (rtt/2) + jitter delay + decode time + render delay.
>
> Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).
>
> BUG=
>
> Committed: https://crrev.com/5249599a9b69ad9c2d513210d694719f1011f977
> Cr-Commit-Position: refs/heads/master@{#11901}
TBR=stefan@webrtc.org,pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:603838
Review URL: https://codereview.webrtc.org/1893543003
Cr-Commit-Position: refs/heads/master@{#12400}
Instead, use the corresponding method on VideoFrameBuffer. In the process,
reduce code duplication in frame comparison functions used in
the test code.
Make FramesEqual use FrameBufsEqual. Make the latter support texture frames.
The cl also refactors VideoFrame::CopyFrame to use I420Buffer::Copy. This
has possibly undesired side effects of never reusing the frame buffer of
the destination frame, and producing a frame buffer which may use different
stride than the source frame.
BUG=webrtc:5682
Review URL: https://codereview.webrtc.org/1881953002
Cr-Commit-Position: refs/heads/master@{#12373}
Also refactor GenericEncoder to use these file writers, and remove use
of preprocessor to enable file writing.
BUG=
Review URL: https://codereview.webrtc.org/1853813002
Cr-Commit-Position: refs/heads/master@{#12372}
We can (and should) use std::vector<std::unique_ptr<T>> instead.
Because it's standard, and because it's safer since callers have to
manually wrap elements in std::unique_ptr before inserting them and
manually unwrap them after inserting them.
Review URL: https://codereview.webrtc.org/1839603002
Cr-Commit-Position: refs/heads/master@{#12182}
Adds logging of:
- video stats that are recorded when a stream is removed
- bitrate stats that are recorded at the end of a call
- initial bwe rampup stats
BUG=
Review URL: https://codereview.webrtc.org/1788783002
Cr-Commit-Position: refs/heads/master@{#12133}
To replace the SmoothsRenderedFrames method, added a corresponding
flag to VideoReceiveStream::Config instead.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1818023002
Cr-Commit-Position: refs/heads/master@{#12102}
This change enables voice-only calls to keep track of the network state.
This is only a partial fix - the last modality to change state controls
the state for the entire call, so a call with a failed video transport
will also stop sending audio packets. Handling this condition correctly
would require the call to keep track of network state for each media
type separately, and take care of conditions such as a failed video
channel getting removed, while a functioning audio channel remains.
BUG=webrtc:5307
Review URL: https://codereview.webrtc.org/1757683002
Cr-Commit-Position: refs/heads/master@{#12093}
Reason for revert:
Breaks RTCVideoEncoder which has incorrect assumptions on where InitEncode etc. is called from. Temporarily reverting until RTCVideoEncoder has been updated.
Original issue's description:
> Initialize/configure video encoders asychronously.
>
> Greatly speeds up setRemoteDescription() by moving encoder initialization
> off the main worker thread, which is free to move onto gathering ICE
> candidates and other tasks while InitEncode() is performed. It also
> un-blocks PeerConnection GetStats() which is no longer blocked on
> encoder initialization.
>
> BUG=webrtc:5410
> R=stefan@webrtc.org
>
> Committed: fb647a67beR=stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:595274, chromium:595308, webrtc:5410
Review URL: https://codereview.webrtc.org/1821983002 .
Cr-Commit-Position: refs/heads/master@{#12086}
Removes code duplication and use of the dangerous public destructor in
RefCountImpl.
Also making wider use of scoped_refptr and fixing various leaks in the
process.
BUG=webrtc:5229
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1477013005 .
Cr-Commit-Position: refs/heads/master@{#12075}
webrtc::VideoRenderer class, replacing it by rtc::VideoSinkInterface.
The next step is to convert all places where a renderer is attached to
rtc::VideoSourceInterface, and at that point, the
SmoothsRenderedFrames method can be replaced by a flag
rtc::VideoSinkWants::smoothed_frames.
Delete unused method IsTextureSupported.
Delete unused time argument to RenderFrame.
Let webrtc::VideoRenderer inherit rtc::VideoSinkInterface. Rename RenderFrame --> OnFrame.
TBR=kjellander@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1814763002
Cr-Commit-Position: refs/heads/master@{#12070}
It was possible that even after a VideoSendStream was destroyed,
it remained registered as a BitrateAllocator observer, causing a
segfault later.
Review URL: https://codereview.webrtc.org/1815733002
Cr-Commit-Position: refs/heads/master@{#12067}
The fundamental issue is that RTCP packet timestamps were accidentally
being fed into wrap_handler_, causing it to think the 32-bit timestamp
had wrapped around when it actually hadn't.
Was also using a 32-bit timestamp instead of a 64-bit timestamp in one
place, meaning that if wrapping actually DID occur, the test would still
fail due to a 64-bit value being cast to a 32-bit value.
BUG=webrtc:5668
R=pbos@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1814023003 .
Cr-Commit-Position: refs/heads/master@{#12055}
Reason for revert:
The openmax_dl include change breaks downstream projects.
Original issue's description:
> Add check_deps rules in DEPS files.
>
> Add fine-grained check_deps rules for all of WebRTC.
> This will help both maintaining sane dependencies and provides a way
> to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
>
> Example:
> buildtools/checkdeps/graphdeps.py --root=. --format=png \
> --out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
> --excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
>
> will produce a neat webrtc.png image showcasing the dependencies
> (according to the DEPS file) for the bitrate_controller module.
> Some dependencies are filtered out for readability.
>
> BUG=webrtc:5623
> TESTED=Passing runs using:
> buildtools/checkdeps/checkdeps.py --root=. talk
> buildtools/checkdeps/checkdeps.py --root=. webrtc
>
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/086f851b7b9b4bcbd4fe507c3bf83b760bd7f4d9
> Cr-Commit-Position: refs/heads/master@{#12008}
TBR=tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5623
Review URL: https://codereview.webrtc.org/1808573002
Cr-Commit-Position: refs/heads/master@{#12009}
Add fine-grained check_deps rules for all of WebRTC.
This will help both maintaining sane dependencies and provides a way
to visualize dependency graphs using the buildtools/checkdeps/graphdeps.py script.
Example:
buildtools/checkdeps/graphdeps.py --root=. --format=png \
--out=./webrtc.png --incl='^webrtc/modules/bitrate_controller->' \
--excl='chromium|base|external|testing|webrtc/test|\.h$|\.cc$'
will produce a neat webrtc.png image showcasing the dependencies
(according to the DEPS file) for the bitrate_controller module.
Some dependencies are filtered out for readability.
BUG=webrtc:5623
TESTED=Passing runs using:
buildtools/checkdeps/checkdeps.py --root=. talk
buildtools/checkdeps/checkdeps.py --root=. webrtc
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1796413002 .
Cr-Commit-Position: refs/heads/master@{#12008}
"WebRTC.Video.AVSyncOffsetInMs"
The absolute value of the sync offset between a rendered video frame and the latest played audio frame is measured per video frame. The average offset per received video stream is recorded when a stream is removed.
Updated sync tests in call_perf_tests.cc to use this implementation.
BUG=webrtc:5493
Review URL: https://codereview.webrtc.org/1756193005
Cr-Commit-Position: refs/heads/master@{#11993}
Greatly speeds up setRemoteDescription() by moving encoder initialization
off the main worker thread, which is free to move onto gathering ICE
candidates and other tasks while InitEncode() is performed. It also
un-blocks PeerConnection GetStats() which is no longer blocked on
encoder initialization.
BUG=webrtc:5410
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1757313002 .
Cr-Commit-Position: refs/heads/master@{#11983}
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
This CL will be followed up with a CL adding AudioSendStream to
BitrateAllocator, so this is a small CL to have the video connection to
BitrateAllocator "at the same level" as for audio.
BUG=webrtc:5079
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1785283002 .
Cr-Commit-Position: refs/heads/master@{#11955}
Also fix a timestamp issue in video analyzer test.
BUG=webrtc:5637, webrtc:5537
Review URL: https://codereview.webrtc.org/1779773002
Cr-Commit-Position: refs/heads/master@{#11938}
* Both timestamps must be unwrapped before comparing
* rtp timestamp delta must be subtracted before unwrapping
BUG=webrtc:5637, webrtc:5537
Review URL: https://codereview.webrtc.org/1774123003
Cr-Commit-Position: refs/heads/master@{#11926}
Previous logged delay was: network delay (rtt/2) + jitter delay + decode time + render delay.
Make capture time in local timebase available for decoded VP9 video frames (propagate ntp_time_ms from EncodedImage to decoded VideoFrame).
BUG=
Review URL: https://codereview.webrtc.org/1688143003
Cr-Commit-Position: refs/heads/master@{#11901}
Makes VideoCaptureInput easier to test and enables running more things
outside VideoCaptureInput on the encoder thread in the future
(initializing encoders and reconfiguring them, for instance).
BUG=webrtc:5410, webrtc:5494
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1763693002 .
Cr-Commit-Position: refs/heads/master@{#11860}
