Added a test that verifies that waiting for a condition variable
actually waits for a non-zero time.
This used to fail due to a TSAN / CLANG bug, but this failure
is supposed to have been fixed.
This was originally https://webrtc-codereview.appspot.com/2145004
BUG=2259
Review URL: https://codereview.webrtc.org/1416873002
Cr-Commit-Position: refs/heads/master@{#10341}
Some toolchains (in this case referring to a g++ 4.9, or "arm-linux-
androideabi-g++ (GCC) 4.9 20140827 (prerelease)" according to my
--version, from the Android NDK r10e-rc4 and potentially with custom
patches; others may be affected as well) fail to prove that myVec in
WebRtcIsac_CorrelateInterVec is never used uninitialized. This is likely
due to the compiler thinking the assignment in line 468 might not
happen. Changing the loop condition in line 466 to rowCntr <
SOME_CONSTANT also helps, suggesting that the compiler can't infer that
there are only 2 values interVecDim can have at that point, and neither
of them are 0. Of course, this is not an acceptable fix, as it changes
behaviour.
This seems to be a compiler bug, or at least an issue with its
heuristics. However, we can't really change toolchains at the moment,
and ultimately this change improves support for certain older compilers.
BUG=
Review URL: https://codereview.webrtc.org/1406423004
Cr-Commit-Position: refs/heads/master@{#10337}
Two concurrently running decoders will trigger data races on cpu_info_
which is lazily initialized on reading TestCpuFlag without proper
atomics.
BUG=libyuv:508
R=kjellander@webrtc.org
TEST=Running EndToEndTest.SendsAndReceivesMultipleStreams under TSan.
Review URL: https://codereview.webrtc.org/1414093003 .
Cr-Commit-Position: refs/heads/master@{#10335}
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"
If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"
BUG=
Review URL: https://codereview.webrtc.org/1311533012
Cr-Commit-Position: refs/heads/master@{#10333}
Reason for revert:
Reverting to see of this resolves build bot failures (Nexus 7.2, especially debug) which now seems to sometimes fail to start tests altogether.
Original issue's description:
> Add screenshare perf tests with lossy links
>
> BUG=
>
> Committed: https://crrev.com/4af0f1a098bc908cfe825981b825687673d5134a
> Cr-Commit-Position: refs/heads/master@{#10290}
TBR=pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1415603002
Cr-Commit-Position: refs/heads/master@{#10322}
- "WebRTC.Video.QualityLimitedResolutionInPercent"
and if a frame is downscaled, the average number of times the frame is downscaled:
- "WebRTC.Video.QualityLimitedResolutionDownscales"
BUG=
Review URL: https://codereview.webrtc.org/1325153009
Cr-Commit-Position: refs/heads/master@{#10319}
This CL changes as little as possible and I'll follow up later with
ownership of the other members in ChannelGroup.
The next step is to remove the id used for channels.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1411723002
Cr-Commit-Position: refs/heads/master@{#10318}
External consumers may have a dependency on the old name, so this will give them the opportunity to switch over.
BUG=
Review URL: https://codereview.webrtc.org/1414543002
Cr-Commit-Position: refs/heads/master@{#10310}
Sounds better according to a MUSHRA listening test.
The computational complexity is unaffected.
An empirically estimated gain was added to compensate for the attenuation introduced by the algorithm.
There are some TODOs, which I will address in follow up CLs.
It was tested in Hangouts without headphones and highest volume, to make sure it doesn't affect the AEC.
Review URL: https://codereview.webrtc.org/1378973003
Cr-Commit-Position: refs/heads/master@{#10308}
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397123003
Cr-Commit-Position: refs/heads/master@{#10307}
Add events to track when camera is requested to open,
when first camera frame is available and when camera is
closed.
BUG=b/24271359
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1398793005 .
Cr-Commit-Position: refs/heads/master@{#10306}
This patch removes set_fail_redirect()/fail_redirect() method accessors
from HttpClient class and converts their usage to
set_redirection_action/redirection_action where appropriate.
BUG=None
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1396683005
Cr-Commit-Position: refs/heads/master@{#10304}
Time to keep old events in buffer is now changeable at runtime.
Added unit test for removing old events from buffer.
Review URL: https://codereview.webrtc.org/1303713002
Cr-Commit-Position: refs/heads/master@{#10302}
This reverts commit 90754174d98d6b71fd4aaed897bd54980f7e59c4.
Revert "Fix use of scaler in MediaCodecVideoEncoder"
This reverts commit ec93628e75fdb81f23635b39b5f3da846bcefd21.
R=magjed@webrtc.orgTBR=glaznev@webrtc.org
BUG=webrtc:4993 b/24984012
Review URL: https://codereview.webrtc.org/1407263002 .
Cr-Commit-Position: refs/heads/master@{#10300}
This removes the TRFC rate control which does not introduce any help in the
computation of the sending rate.
BUG=5083
Review URL: https://codereview.webrtc.org/1383813003
Cr-Commit-Position: refs/heads/master@{#10299}
This is the first CL to get ready for adapting audio bitrate based on
BWE. I've kept this CL as small as possible and had to add a few getters
to ChannelManager. The next CL will do the same for receive ViEChannels.
The getters are a bit uggly, but is an in-between-state. Let's discuss
future ownership of the different modules and what do do with
ChannelGroup.
BUG=5079
Review URL: https://codereview.webrtc.org/1394243006
Cr-Commit-Position: refs/heads/master@{#10298}
Merging the PeerConnectionTestClientBase and JsepTestClient classes,
since there's no real logical distinction. This should make it slightly
less painful to write new PeerConnection tests.
Review URL: https://codereview.webrtc.org/1393223005
Cr-Commit-Position: refs/heads/master@{#10292}
