On Windows8/10, we prefer cropping desired window out from a whole screen
capture due to some reasons. The problem is Win10 has an invisible border
around the window. If we leave the border, it will expose background a bit.
This cl is about to always remove the border of desired window on Win8/10.
This will help a lot to capturing still windows during window sharing.
This cl still can't handle the background exposure issue when you move the
target window around during capturing. More investigation is needed.
BUG=chromium:737278
Review-Url: https://codereview.webrtc.org/2973853002
Cr-Commit-Position: refs/heads/master@{#18921}
In practice, this change will make AudioFrame::muted_ replicate the
explicit muted variable, passed as a pointer to NetEq::GetAudio.
BUG=webrtc:7944
Review-Url: https://codereview.webrtc.org/2965203002
Cr-Commit-Position: refs/heads/master@{#18914}
Lower then bitrate so that the delta between the highest layer of the
lower stream and lowest layer of the higher stream is not too large.
BUG=webrtc:4172
This is a reland of the following CL:
Review-Url: https://codereview.webrtc.org/2791273002
Cr-Commit-Position: refs/heads/master@{#18232}
Committed: dceb42da3e
https: //codereview.webrtc.org/2883963002
Review-Url: https://codereview.webrtc.org/2966833002
Cr-Commit-Position: refs/heads/master@{#18913}
Some frames are already marked as 'timing frames' via video-timing RTP header extension. Timestamps along full WebRTC pipeline are gathered for these frames. This CL implements reporting of these timestamps for a single
timing frame since the last GetStats(). The frame with the longest end-to-end delay between two consecutive GetStats calls is reported.
The purpose of this timing information is not to provide a realtime statistics but to provide debugging information as it will help identify problematic places in video pipeline for outliers (frames which took longest to process).
BUG=webrtc:7594
Review-Url: https://codereview.webrtc.org/2946413002
Cr-Commit-Position: refs/heads/master@{#18909}
* Remove the ReleaseDecoder and Release methods that were used in combination with deleting the decoder object. Now simply deleting the object does the right thing.
* Remove 'friend' relationship between the two classes since they don't need to touch each other's state directly anymore.
* Use std::unique_ptr for holding pointers and transferring ownership.
These changes were previously reviewed here:
https://codereview.webrtc.org/2764573002/
BUG=webrtc:7361, 695438
Review-Url: https://codereview.webrtc.org/2966823002
Cr-Commit-Position: refs/heads/master@{#18908}
The implementation of this method did not follow the description in
the method comment. It was supposed to delete all packets in a range
[A, B], but if at least one packet in the buffer had a timestamp lower
than A, then no packets at all were discarded. This is now fixed.
BUG=webrtc:7937
Review-Url: https://codereview.webrtc.org/2969123003
Cr-Commit-Position: refs/heads/master@{#18903}
1) Function responsible for receiving feedback, digesting data and deciding switch scenarios.
2) Function which enters Startup mode.
3) Function which exits Startup mode.
4) Function which calculates, whether or not full bandwidth is reached.
BUG=webrtc:7713
Review-Url: https://codereview.webrtc.org/2924603002
Cr-Commit-Position: refs/heads/master@{#18901}
This change extends the definition of wired headset to also include USB
devices. The effect is that audio will now be routed to USB audio devices
when used in combination with AppRTCMobile.
BUG=webrtc:7931
Review-Url: https://codereview.webrtc.org/2971613003
Cr-Commit-Position: refs/heads/master@{#18889}
Original issue:
https://codereview.webrtc.org/2957073002/
Reason for reland:
Failed Android unit tests and failed Windows compile.
The tests seemed related at the time, but not after more consideration.
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2970793003
Cr-Commit-Position: refs/heads/master@{#18886}
This CL addresses the issue of echo leakage of low level
echoes by making the echo canceller more restrictive for
that scenario.
BUG=webrtc:7930
Review-Url: https://codereview.webrtc.org/2969943002
Cr-Commit-Position: refs/heads/master@{#18884}
Reason for revert:
The following, seemingly related, unit tests crash on Android32 (M Nexus5X).
org.webrtc.PeerConnectionTest#testCompleteSession
org.webrtc.PeerConnectionTest#testDataChannelOnlySession
A Windows build fails with a mysterious compile error.
Original issue's description:
> Add received audio/video call duration metrics based on packets.
>
> Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
>
> BUG=webrtc:7882
>
> Review-Url: https://codereview.webrtc.org/2957073002
> Cr-Commit-Position: refs/heads/master@{#18881}
> Committed: 746749237aTBR=stefan@webrtc.org,aleloi@webrtc.org,asapersson@webrtc.org,holmer@google.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2972613002
Cr-Commit-Position: refs/heads/master@{#18882}
Tracks time between first and last audio and packets to successfully pass through Call object's DeliverRtp method, timed with packet timestamps.
BUG=webrtc:7882
Review-Url: https://codereview.webrtc.org/2957073002
Cr-Commit-Position: refs/heads/master@{#18881}
The complexity test for the audio processing module have long proven
to give false alarms of complexity regressions for which no related
changes can be identified. Attempts to address that has improved the
that, but the tests do still give false alarms.
This CL deactivates the complexity tests until a better way of
testing this is available.
BUG=chromium:713507, webrtc:5846,webrtc:6685,webrtc:7712
Review-Url: https://codereview.webrtc.org/2897403006
Cr-Commit-Position: refs/heads/master@{#18879}
