The new class is intended to be used as a central time-keeping object
inside NetEq. The actual use of the class will come in subsequent
changes.
BUG=webrtc:5608
Review URL: https://codereview.webrtc.org/1910523003
Cr-Commit-Position: refs/heads/master@{#12477}
This is done by changing the RateStatistics so that it resets its window when the accumulator is empty. It also keeps a dynamic window, so that the rates computed before a full window worth of data has been received will be computed over a smaller window. This means that the rate will be closer to the true rate, but with a higher variance.
BUG=webrtc:5773
R=perkj@webrtc.org, sprang@webrtc.org
Review URL: https://codereview.webrtc.org/1908893003 .
Cr-Commit-Position: refs/heads/master@{#12470}
They can all benefit from moving, since they contain std::string and
std::vector. We intended to add these in
https://codereview.webrtc.org/1896953004/, but got compiler errors we
couldn't make sense of, so we skipped them. It turns out that what the
compiler was complaining about was that when we said we'd have a
user-defined move constructor, it stopped generating a copy assignment
operator for us. This CL solves the problem by outfitting the types
with defaulted copy and move assignment operators too.
Review URL: https://codereview.webrtc.org/1899173002
Cr-Commit-Position: refs/heads/master@{#12469}
This will allow us to fix the sample rate of each AudioDecoder at
instantiation time.
This change results in different checksums for the following tests:
AcmReceiverBitExactnessOldApi.8kHzOutput
AcmReceiverBitExactnessOldApi.16kHzOutput
AcmReceiverBitExactnessOldApi.32kHzOutput
AcmReceiverBitExactnessOldApi.48kHzOutputExternalDecoder
AcmReceiverBitExactnessOldApi.48kHzOutput
Because they make an ACM and then ask it to decode both 16 kHz and 32
kHz iSAC. (The arm32 and arm64 checksums didn't change, because the
tests skip 32 kHz iSAC on arm.)
BUG=webrtc:5801
Review URL: https://codereview.webrtc.org/1908923002
Cr-Commit-Position: refs/heads/master@{#12463}
The fs_hz member variable is going away too, being replaced by a
method in the AudioDecoder interface. If we ever end up needing the
RTP sample rate here, a method ought to be the right solution for that
too.
BUG=webrtc:5801
Review URL: https://codereview.webrtc.org/1907183002
Cr-Commit-Position: refs/heads/master@{#12462}
Removed deprecated files, types and methods in modules/webrtc that were
kept there to avoid breaking chromium, and which are no longer needed.
BUG=172183
Review URL: https://codereview.webrtc.org/1909593002
Cr-Commit-Position: refs/heads/master@{#12457}
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}
Current implementation of AEC metrics does not read nicely. It messes up between a noise-removed calculation and a raw calculation.
I tried to clean it up, in which, I stick to the raw calculation since the noise-removed version can show some problem when the noise is overestimated.
BUG=
Review URL: https://codereview.webrtc.org/1581183005
Cr-Commit-Position: refs/heads/master@{#12455}
The first approach landed here: https://codereview.webrtc.org/1773173002
But it was partially reverted, because it affected the AEC performance, here: https://codereview.webrtc.org/1867483003/
The main difference of this approach is that it doesn't use the 3-band splitting filter in the reverse stream, which seems to be the culprit of the AEC regression.
Also, the 2-band splitting filter has been used for the 32kHz case for a long time without any problem, and this is expanded in the CL to cover the 48kHz case as well.
BUG=webrtc:5725
TBR=tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1865633005
Cr-Commit-Position: refs/heads/master@{#12451}
Integer wraparound when casting from int32 to int16 can cause invalid array indices to be accessed.
Fix for wraparound issue.
BUG=webrtc:5781
Review URL: https://codereview.webrtc.org/1894483002
Cr-Commit-Position: refs/heads/master@{#12449}
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}
Declared in webrtc::VideoRender, implemented in IncomingVideoStream.
This cl also eliminates some of the few uses of
webrtc::VideoFrame::CopyFrame.
BUG=webrtc:5682
Review URL: https://codereview.webrtc.org/1885323002
Cr-Commit-Position: refs/heads/master@{#12427}
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}
By eliminating one of the two constructors, handling decoder ownership
with a unique_ptr instead of a raw pointer, and making all member
variables const (except one, which is made private instead).
BUG=webrtc:5801
Review URL: https://codereview.webrtc.org/1899733002
Cr-Commit-Position: refs/heads/master@{#12425}
processing module experiment description that was present
when AEC3 was not activated and when RefinedAdaptiveFilter
was activated.
BUG=webrtc:5778, webrtc:5777
Review URL: https://codereview.webrtc.org/1899123002
Cr-Commit-Position: refs/heads/master@{#12424}
Makes QualityScaler start at QVGA for <250k initial bitrates. Useful in
combination with overriding max bitrates to a max lower than that for
connections where we know that the max bitrate is capped below where VGA
is useful.
BUG=webrtc:5678
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1900483004 .
Cr-Commit-Position: refs/heads/master@{#12416}
Remove the deprecated EncodeInternal interface from AudioEncoder
Also hid MaxEncodedBytes by making it private. It will get removed as soon as subclasses have had time to remove their overrides.
BUG=webrtc:5591
Review URL: https://codereview.webrtc.org/1881003003
Cr-Commit-Position: refs/heads/master@{#12409}
Increases measure time for downscale back to 5 seconds, this is required
to not over-react on hand-waving or quick device rotations.
Also increase max thresholds for QP a bit to not overreact when quality
still looks somewhat OK. Min thresholds for H264 seemed very low and are
increased to be sure that we can go back up again. The window is still
quite big with the increased max QP.
Also changes libvpx thresholds to use the same thresholds as the
encoder, they were excessively low before and wouldn't adapt on bad QPs
at all before (but rely on >60% framedropping based on bitrates to go
down).
BUG=webrtc:5678
R=stefan@webrtc.orgTBR=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1894083002 .
Cr-Commit-Position: refs/heads/master@{#12403}
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}
does not have to use the aec state as an input.
Furthermore, the debug dump output of e_fft was removed as
it is not really used in any analysis scripts.
BUG=webrtc:5298
Review URL: https://codereview.webrtc.org/1883293003
Cr-Commit-Position: refs/heads/master@{#12387}
The following algorithmic functionality was added:
-Add support for an exact regressor power to be computed
which avoids the issue with the updating of the filter
sometimes being unstable.
-Lowered the fixed step size of the adaptive filter to 0.05
which significantly reduces the sensitivity of the
adaptive filter to near-end noise, nonlinearities,
doubletalk and the unmodelled echo path tail. It also
reduces the tracking speed of the adaptive filter but the
chosen value proved to give a sufficient tradeoff for the
requirements on the adaptive filter.
To allow the new functionality to be selectively applied the following was done:
-A new Config was added for selectively activating the functionality.
-Functionality was added in the audioprocessing and echocancellationimpl classes
for passing the activation of the functionality down to the AEC algorithms.
To make the code for the introduction of the functionality clean,
the following refactoring was done:
-The selection of the step size was moved to a single place.
-The constant for the step size of the adaptive filter in extended filter mode was
made local.
-The state variable storing the step-size was renamed to a more describing name.
When the new functionality is not activated, the changes
have been tested for bitexactness on Linux.
TBR=minyue@webrtc.org
BUG=webrtc:5778, webrtc:5777
Review URL: https://codereview.webrtc.org/1887003002
Cr-Commit-Position: refs/heads/master@{#12384}
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}
