- org.appspot.apprtc.test.PeerConnectionClientTest#testCaptureFormatChange
- org.appspot.apprtc.test.PeerConnectionClientTest#testVideoSourceRestart
The tests were previously flaky but significant changes to the test
architecture have been made since. Opportunistically try to re-enable
them.
BUG=webrtc:6478
Review-Url: https://codereview.webrtc.org/2748863003
Cr-Commit-Position: refs/heads/master@{#17245}
Bad changes are from CL 2745523002.
These changes were originally done by Sprang@. Sometimes, when encoder is failed to be configured on release build it causes a crash at vie_encoder.cc:451. That changes look like they are not important to other changes. This CL is simply reverting them.
BUG=chromium:701526
Review-Url: https://codereview.webrtc.org/2747403002
Cr-Commit-Position: refs/heads/master@{#17241}
It appears that for some H264 streams that the width/height is not set for
the first packet of the IDR but in the packet containing the SPS/PPS.
BUG=chromium:698088, webrtc:7139
Review-Url: https://codereview.webrtc.org/2750633003
Cr-Commit-Position: refs/heads/master@{#17239}
I cannot even imagine this change is useful. But it consistently reduces average
capture time by 0.375% (4.07 -> 4.055), and average encode time by 0.313%
(8.042 -> 8.016) without other impacts. Considering this is a one-line change,
it's worthy to be added.
BUG=679523
Review-Url: https://codereview.webrtc.org/2743233002
Cr-Commit-Position: refs/heads/master@{#17235}
The getters are not used and the implementation cannot be guaranteed
to return a correct value except when called synchronously from
the decoding thread while decoding.
The methods as is imply that the implementation needs to offer some
sort of synchronization, and that's not desirable.
BUG=webrtc:7328
R=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2741853008 .
Cr-Commit-Position: refs/heads/master@{#17233}
different sample rate frequency.
BUG=webrtc:7327
Problems before the fix:
1. NetEqImpl::timestamp_ is inconsistent. Initially it is set to
the original RTP timestamp, but later gets updated with the
scaled timestamp.
2. NetEqImpl::InsertPacketInternal::main_timestamp is set with
the original RTP timestamp, but later gets compared with the
NetEqImpl::timestamp_ which may or may not be with the same
sample rate frequency and this results in major problems.
3. IncreaseEndTimestamp(main_timestamp - timestamp_) will be
incorrect when SSRC is changed and not the first packet.
4. delay_manager_->Update() may not be always invoked, since
the (main_timestamp - timestamp_) >= 0 will not be true when
the previous scaled timestamp_ is bigger than the main_timestamp
(current RTP timestamp) even if the current RTP timestamp is
bigger than the previous RTP timestamp.
5. delay_manager_->Update() parameters are main_timestamp
which increments with the RTP sample rate frequency and the
fs_hz_ which is the decoder sample rate frequency. When these
two frequencies are different as is the case with g.722, the
DelayManager::Update() will misfire and calculate incorrect
packet_len_ms and inter-arrival time (IAT) as a result. This
in effect will cause neteq to enter kPreemptiveExpand operation
and will keep expanding the jitter buffer even if the RTP packets
arrive with no jitter at all.
The fix corrects all these problems by making sure the
main_timestamp and the timestamp_ are always set with the scaled
timestamp and increment with the decoder sample rate frequency.
Review-Url: https://codereview.webrtc.org/2743063005
Cr-Commit-Position: refs/heads/master@{#17232}
* The _receiveCallback member of VCMDecodedFrameCallback does actually not require locking now that the threading model is slightly clearer. Documentation and checks have been added.
* UserReceiveCallback() never returns null and must always be called on the decoder thread. Checks have been added and the two test suites that were failing to set this callback, have been fixed and a new mock class added. (looks like sakal@ may have hit some issues with flaky tests there).
* Changed VcmPayloadSink to use move semantics which I suspect was the intention at the time the code was written (when we didn't have move semantics).
* Added thread checker to a couple of classes and started adding thread checks for known behavior. There's more to be done there.
* Remove the |_decoder| member variable in VideoReceiver. It is not needed and as it could be used, left us open to a race.
* TODOs added for places where we can reduce locking. I suspect that we can get away with not needing a lock around _codecDataBase in VideoReceiver once we've got a clear picture of the threading model and ensured that all adhere to it.
BUG=webrtc:7328
Review-Url: https://codereview.webrtc.org/2744013002
Cr-Commit-Position: refs/heads/master@{#17226}
The --purify flag can now be passed to remove the temporary
files and directories created while building the iOS Framework or static
library. That way, only the final result(s) are taking up space in the
output folder.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2740923003
Cr-Commit-Position: refs/heads/master@{#17224}
The only non-const operation was a one-time initialization of a member only used in this function. I've moved it to the ctor.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2741733002
Cr-Commit-Position: refs/heads/master@{#17223}
By default the return value will be 0, which if we hit, could cause busy loops.
BUG=webrtc:7187
Review-Url: https://codereview.webrtc.org/2750503002
Cr-Commit-Position: refs/heads/master@{#17213}
Reason to go back is that we may end up with a bunch of streams that are never cleaned up and consume resources.
BUG=webrtc:7175, b/35863246
Review-Url: https://codereview.webrtc.org/2746763002
Cr-Commit-Position: refs/heads/master@{#17210}
A DCHECK added in a recent bugfix, which asserted that a signed 64->32
bit cast did not overflow, has been found to not always pass. We fix
this by saturating.
BUG=chromium:693868
Review-Url: https://codereview.webrtc.org/2746903002
Cr-Commit-Position: refs/heads/master@{#17209}
Modification affects EglRenderer on Android. Moves frame dropping to the
renderer thread. Frame listeners are triggered even when FPS reduction is
active unless applyFpsReduction is set to true.
BUG=webrtc:7149
Review-Url: https://codereview.webrtc.org/2688843002
Cr-Commit-Position: refs/heads/master@{#17206}
This was a trivial delegation wrapper, with only a single use.
BUG=None
Review-Url: https://codereview.webrtc.org/2741413003
Cr-Commit-Position: refs/heads/master@{#17205}
Turns out temporal_layer_thresholds_bps doesn't work quite as expected.
It's for instance not honored at all for normal VP8 video. We need to
take a pass over this in general.
BUG=chromium:700297
Review-Url: https://codereview.webrtc.org/2744823002
Cr-Commit-Position: refs/heads/master@{#17199}
Reason for revert:
CallPerfTest.ReceivesCpuOveruseAndUnderuse perf test fails due to this CL. It requires very accurate frame rate, which may not be so accurate now.
Original issue's description:
> Reland of rewrite frame generator capturer to use TaskQueue instead of EventTimeWrapper (patchset #1 id:1 of https://codereview.webrtc.org/2743993002/ )
>
> And enable large full-stack test depending on that change (Reland of https://codereview.webrtc.org/2741823003/)
> TBR=stefan@webrtc.org,tommi@webrtc.org
> BUG=webrtc:7301,webrtc:7325
>
> Review-Url: https://codereview.webrtc.org/2744003002
> Cr-Commit-Position: refs/heads/master@{#17196}
> Committed: 8c0a5896d1TBR=stefan@webrtc.org,tommi@webrtc.org,sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7301,webrtc:7325
Review-Url: https://codereview.webrtc.org/2748643002
Cr-Commit-Position: refs/heads/master@{#17198}
