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}
Add IsClosed check when excuting some functions so that they can return early if the PeerConnection is closed.
The observer will not be called after the PeerConnection is closed.
BUG=webrtc:5861
Review-Url: https://codereview.webrtc.org/1975453002
Cr-Commit-Position: refs/heads/master@{#13544}
Apparently, a class will fail with VerifyError if it contains catch
statements with an Exception from a newer API, even if the code is never
executed. This happens only on Android versions before 4.4.2 and is a
bug. See https://code.google.com/p/android/issues/detail?id=209129 for
more info.
BUG=b/30376736
Review-Url: https://codereview.webrtc.org/2185833003
Cr-Commit-Position: refs/heads/master@{#13542}
Updated the sources in audio_processing:audioproc_test_utils to match the configuration on
"webrtc/modules/audio_processing/audio_processing_tests.gypi"
Removed audio_buffer_tools from modules_unittests to match the gyp file.
BUG=webrtc:6041
Review-Url: https://codereview.webrtc.org/2178963002
Cr-Commit-Position: refs/heads/master@{#13541}
There is a bug on pre LMR1 devices that only support legacy
implementation of camera2 API where aspect ratio of the camera is
incorrect if the output format doesn't match the aspect ratio of the
sensor array. On these devices, we want to disable the output formats that
have different aspect ratio.
Review-Url: https://codereview.webrtc.org/2181803003
Cr-Commit-Position: refs/heads/master@{#13538}
It can fail in some real circumstances, such as when IDs are exhausted
or you explicitly try to create one with an already-used ID.
Review-Url: https://codereview.webrtc.org/2181933002
Cr-Commit-Position: refs/heads/master@{#13535}
These tests will be reenabled and fixed after Opus 1.1.3 has landed in
Chromium and is rolled into WebRTC.
BUG=
Review-Url: https://codereview.webrtc.org/2185673002
Cr-Commit-Position: refs/heads/master@{#13534}
maximum allowed sized raised from limited by physical udp packet size to
limited by theoritical maximum rtcp packet size.
BUG=webrtc:5260
R=åsapersson
Review-Url: https://codereview.webrtc.org/1998633002
Cr-Commit-Position: refs/heads/master@{#13532}
This is a first CL wiring up AudioSendStream to BitrateAllocator. This
is still experimental and there is a test added for the audio only case,
combined audio video variable bitrate test cases will be added as a
follow up.
BUG=5079
Review-Url: https://codereview.webrtc.org/2165743003
Cr-Commit-Position: refs/heads/master@{#13527}
The iOS H264 video toolbox encoder is currently undershooting the
intended bitrate. This seems to be caused by the data rate limit
property. This CL increases the data rate limit to a set
percentage above the intended bitrate to avoid undershooting. The
AverageBitRate property is still set to the intended bitrate, which
keeps it from overshooting the intended bitrate.
BUG=b/28713684
Review-Url: https://codereview.webrtc.org/2177873003
Cr-Commit-Position: refs/heads/master@{#13526}
to be aware about rare situation where packet resend before sent:
Expectations reduced by validating frame was rendered after or before last
packet for that frame was dropped.
BUG=webrtc:5540
Review-Url: https://codereview.webrtc.org/2180903002
Cr-Commit-Position: refs/heads/master@{#13523}
Test was expecting no rtx packet before dropped packet.
Because of prober there might be some non-padding rtx packets before nack.
Those checks removed, test primary expectations are unaffected.
BUG=webrtc:5540
R=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2180843002
Cr-Commit-Position: refs/heads/master@{#13522}
This CL fixes a crash that could happen when JSON event tracing is
shutting down. The cause of the crash was the fact that the logger
thread function was returning 'true', causing the platform thread to run
it repeatedly even though that wasn't the intention.
Usually the EventLogger::Stop() function would set the event requesting
the logging thread to clean up and close the file, and then immediately
call PlatformThread::Stop() which would stop the outer loop. The Log()
function would only run once and everything behaves as expected.
However, if a context switch happens between the shutdown_event_.Set()
and logging_thread_.Stop() calls in EventLogger::Stop(), the logger
thread function would close the file and exit the Log() method, while
PlatformThread will rerun it again. So the Log() function runs twice,
and the second time output_file_ is NULL which either causes the DCHECK
to fail (in debug builds) or the fprintf() to crash with SIGSEGV (in
release builds).
The fix simply changes the return value of the thread function to false
so it never runs twice.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2168283002 .
Cr-Commit-Position: refs/heads/master@{#13510}
