Passing transport_frame_id() to VideoSink will allow to identify incoming video
frames, which will make it possible to correlate video frames on the
sender and on the receiver.
BUG=chromium:621691
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2088953002 .
Cr-Commit-Position: refs/heads/master@{#13596}
A reset of Bwe and rates on network route change triggers
switching of the RemoteBitrateEstimator from DelayBasedBwe
to RemoteBitrateEstimatorAbsSendTime. This is unexpected,
fix this.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2193043002 .
Cr-Commit-Position: refs/heads/master@{#13594}
Also makes the properties protected, as previously requested by Stefan.
Review-Url: https://codereview.webrtc.org/2179223003
Cr-Commit-Position: refs/heads/master@{#13593}
The intended behavior of the pacer is that it should send audio (i.e high priority) packets immediatelly even if the pacer is paused and even if the available media budget has been used. The existing code will not send audio if the pacing budget has been used. It will normally send audio when the pacer is paused, but if the media budget was already used up when the pacer was paused, it would neither send audio nor update the budget.
Updated the implementatation and the unit test. Added logging for when the pacer is paused, and when it receives a zero bitrate estimate.
BUG=webrtc:6155
Review-Url: https://codereview.webrtc.org/2193183003
Cr-Commit-Position: refs/heads/master@{#13592}
The ProbingCalculator class calculates and validates the results from
probing attempts.
BUG=webrtc:5859
Review-Url: https://codereview.webrtc.org/2121183002
Cr-Commit-Position: refs/heads/master@{#13589}
double check rtp_sender in sending mode when altering sequence_number
adjust test to skip validating timestamp on rtx streams
fix test by waiting for all 3 media streams instead of 3 out 6 media and rtx streams.
BUG=webrtc:4332
Review-Url: https://codereview.webrtc.org/2177523002
Cr-Commit-Position: refs/heads/master@{#13587}
The LOG_END time is incorrect if the event log is stopped
by the file size limit instead of a duration limit or a
manual stop. This makes the call appear to be very long
(10^13 seconds) in the analysis tool. This CL is a workaround
for that problem.
BUG=webrtc:6138
Review-Url: https://codereview.webrtc.org/2176663002
Cr-Commit-Position: refs/heads/master@{#13585}
This happens if we stop logging because we have reached the file size limit. The large timestamp causes problems in the analysis tool.
BUG=webrtc:6138
Review-Url: https://codereview.webrtc.org/2175713002
Cr-Commit-Position: refs/heads/master@{#13581}
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.
R=phoglund@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2193763002 .
Cr-Commit-Position: refs/heads/master@{#13574}
The new version is much shorter than the old one, and hopefully easier
to read. This is part of the effort to rewrite the old mixer.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2132563002
Cr-Commit-Position: refs/heads/master@{#13570}
Reason for revert:
Multiple definitions of webrtc::MockMixerParticipant::MockMixerParticipant() during linking of modules_unittests. Please investigate and resubmit.
Original issue's description:
> Rewrote UpdateToMix in the audio mixer.
>
> The new version is much shorter than the old one, and hopefully easier
> to read. This is part of the effort to rewrite the old mixer.
>
> Committed: https://crrev.com/2942e240f4a985752714dac18c141064c97696d4
> Cr-Commit-Position: refs/heads/master@{#13568}
TBR=ossu@webrtc.org,ivoc@webrtc.org,aleloi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2195633002
Cr-Commit-Position: refs/heads/master@{#13569}
The new version is much shorter than the old one, and hopefully easier
to read. This is part of the effort to rewrite the old mixer.
Review-Url: https://codereview.webrtc.org/2132563002
Cr-Commit-Position: refs/heads/master@{#13568}
If a port is not used by any channel and if it has no connection for 30
seconds, it will be removed.
Note, as long as a port is used by a transport channel, it will be kept
even if it does not have any connection. This will be beneficial to
continual gathering because new connections can be created in the future
when network changes.
BUG=
R=pthatcher@webrtc.org, zhihuang@webrtc.org
Review URL: https://codereview.webrtc.org/2171183002 .
Cr-Commit-Position: refs/heads/master@{#13567}
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.
We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.
I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2192963002 .
Cr-Commit-Position: refs/heads/master@{#13566}
- Renamed variables and some function to comply with style guide.
- Removed default argument values.
- Removed some dead code.
- Cleaned up comments formatting in rtp_rtcp.h
R=danilchap@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2067673004 .
Cr-Commit-Position: refs/heads/master@{#13565}
If all connections on a port is destroyed, it will schedule an event
to check if it is dead after a timeout. Previously if a new connection
is created but destroyed before the event is fired, it will destroy the
port. With this change, we will not destoy it until it times out again
after the last created connection is destroyed.
BUG=
R=pthatcher@webrtc.org, zhihuang@webrtc.org
Review URL: https://codereview.webrtc.org/2184013003 .
Cr-Commit-Position: refs/heads/master@{#13563}
Due to a recent interface change for svc_params in vp9 svc, which
allows speed setting per layer, svc_params should be inited to 0
for safety.
Review-Url: https://codereview.webrtc.org/2179753003
Cr-Commit-Position: refs/heads/master@{#13561}
It was generating a random ID using the test case's "this" pointer
and the current time. However, the current time may be imprecise. And
the "this" pointer may have repeatable values.
BUG=webrtc:5898
Review-Url: https://codereview.webrtc.org/2190533004
Cr-Commit-Position: refs/heads/master@{#13560}
Reason for revert:
Breaks downstream targets.
Original issue's description:
> Add BWE plot to event log analyzer.
>
> The plot is constructed by actually running the congestion controller with
> the logged rtp headers and rtcp feedback messages to reproduce the same behavior
> as in the real call.
>
> R=phoglund@webrtc.org, terelius@webrtc.org
>
> Committed: https://crrev.com/2beea2a8c920000ef19eea20cce397507fc3d5e7
> Cr-Commit-Position: refs/heads/master@{#13558}
TBR=phoglund@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2190013002
Cr-Commit-Position: refs/heads/master@{#13559}
The plot is constructed by actually running the congestion controller with
the logged rtp headers and rtcp feedback messages to reproduce the same behavior
as in the real call.
R=phoglund@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2188033004 .
Cr-Commit-Position: refs/heads/master@{#13558}
This is an issue if the sequence numbers are to be used to compute packet loss statistics since it introduces gaps which are not related to loss.
Also making sure that the header extensions are properly guarded by the send crit sect.
Review-Url: https://codereview.webrtc.org/2190913002
Cr-Commit-Position: refs/heads/master@{#13557}
The goal of this change is to log the volume level for the
current audio stream so we can keep track of what volume the
user selects during a call.
BUG=b/30376577
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/2182043005 .
Cr-Commit-Position: refs/heads/master@{#13555}
Memory frames are now expected to be owned by the mixing participants.
Review-Url: https://codereview.webrtc.org/2127763002
Cr-Commit-Position: refs/heads/master@{#13554}
In Android AppRTC Demo, there was a bug where toggling disable hardware
AGC/NS would not be reflected into the summary of those settings. This
change fixes this issue.
Review-Url: https://codereview.webrtc.org/2184223003
Cr-Commit-Position: refs/heads/master@{#13551}
OutputMixer and AudioConferenceMixer communicated via a callback. OutputMixer implemented an AudioMixerOutputReceiver interface, which defines the callback function NewMixedAudio. This has been removed and replaced by a simple function in the new mixer. The audio frame with mixed audio is now copied one time less. I have also removed one forward declaration.
Review-Url: https://codereview.webrtc.org/2111293003
Cr-Commit-Position: refs/heads/master@{#13550}
