CalculateFrequency() results in zero frequency (floating point) if the RTP timestamps in the RTCP list are equal.
Added check in UpdateRtcpList to not insert RTCP SR with the same RTP timestamp.
BUG=webrtc:5780
Review URL: https://codereview.webrtc.org/1891703002
Cr-Commit-Position: refs/heads/master@{#12429}
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}
on an IPv6 network that contains the actual default local address. This is for preventing potential IP leaking.
BUG=webrtc:5376
Review URL: https://codereview.webrtc.org/1837823005
Cr-Commit-Position: refs/heads/master@{#12417}
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}
For historical reasons gyp_webrtc.py was launcher script
for gyp_webrtc and the python logic lived in the
gyp_webrtc. This change moves python code into the
.py file makes the extension-free gyp_webrtc a launcher
for gyp_webrtc.py.
Other changes:
* Move the code into a main() function.
* Add call to disable GC to save some processing time.
* Set executable permission on gyp_webrtc.py and remove it from
gyp_webrtc.
Similar Chromium CL: https://codereview.chromium.org/1216863010
Motivation for this change:
* Gets checked with PyLint
* Easy to add unit tests if we add our own functionality.
R=phoglund@webrtc.orgTBR=tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1895713002 .
Cr-Commit-Position: refs/heads/master@{#12410}
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}
Eliminate most uses of the old methods.
To continue on this path, once we agree the new methods make sense,
the next step is to rename cricket::VideoFrame::GetVideoFrameBuffer
--> video_frame_buffer, to match the name in webrtc::VideoFrame (if we
think that name is ok?). And then start updating all code to access
planes via the VideoFrameBuffer, and delete corresponding methods in
both cricket::VideoFrame and webrtc::VideoFrame.
BUG=webrtc:5682
Review URL: https://codereview.webrtc.org/1878623002
Cr-Commit-Position: refs/heads/master@{#12407}
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}
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}
Reason for revert:
I think the bug has been making enough progress for us to turn this back on (Chrome turned on their trybot).
I fired of builds and now have 6 green ones in a row.
Original issue's description:
> CQ: Disable win_x64_clang_dbg trybot
>
> Needed to unblock rolling chromium_revision in WebRTC DEPS.
>
> BUG=chromium:595702
> TBR=phoglund@webrtc.org
>
> Committed: 053fe8c6b1TBR=phoglund@webrtc.org
NOTRY=True
BUG=chromium:595702
Review URL: https://codereview.webrtc.org/1897743002
Cr-Commit-Position: refs/heads/master@{#12399}
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}
