I noticed while profiling calls to TimeUntilNextProcess.
It's benign, but the implementation seems to be done to accomodate the
test rather than the other way around, so I'm making the test behave
more like a ProcessThread would.
BUG=None
Review-Url: https://codereview.webrtc.org/2715133002
Cr-Commit-Position: refs/heads/master@{#16867}
Fixes a bug where a missing boolean-not causes AppRTCMobile to
always crash on an assert when built for Debug.
BUG=None
NOTRY=True
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2721583002
Cr-Commit-Position: refs/heads/master@{#16865}
Reason for revert:
Reverting while fixing build issue in Chromium.
Original issue's description:
> Use TaskQueue in IncomingVideoStream instead of the PlatformThread + event timer approach.
>
> BUG=webrtc:7219, webrtc:7253
>
> Review-Url: https://codereview.webrtc.org/2716473002
> Cr-Commit-Position: refs/heads/master@{#16860}
> Committed: e2d1d64295TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7219, webrtc:7253
Review-Url: https://codereview.webrtc.org/2714393003
Cr-Commit-Position: refs/heads/master@{#16863}
In this CL:
- Add message BweProbeCluster and BweProbeResult to rtc_event_log.proto.
- Add corresponding log functions to RtcEventLog.
- Add optional field |probe_cluster_id| to RtpPacket message and added
an overload function to log with this information.
- Propagate the probe_cluster_id to where RTP packets are logged.
BUG=webrtc:6984
Review-Url: https://codereview.webrtc.org/2666533002
Cr-Commit-Position: refs/heads/master@{#16857}
Also reducing locking in NackModule (and by extension RtpStreamReceiver)
BUG=webrtc:7246
Review-Url: https://codereview.webrtc.org/2720603002
Cr-Commit-Position: refs/heads/master@{#16856}
Connection::nominated() is updated to mean
(remote_nomination_ || acked_nomination_), which means both a
controlling and controlled agent can be said to be "nominated".
Previously this was (remote_nomination_ > 0) which only applies to the
controlling agent.
PortTest.TestNomination added to test nomination values and nomination
stat.
This value is surfaced through cricket::ConnectionInfo::nominated.
RTCStatsCollector uses this value in its collection of
RTCIceCandidatePairStats.
RTCStatsCollectorTest.CollectRTCIceCandidatePairStats updated to test
that ConnectionInfo::nominated is surfaced using mocks.
rtcstats_integrationtest.cc updated to expect nomination set without
using mocks.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-nominated
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2709293004
Cr-Commit-Position: refs/heads/master@{#16855}
This CL adds correction of the echo suppressor gain
It contains 2 changes:
-Bounds the upper value for the echo suppression gain
for bin 1 to avoid that the high-pass filter causes the
gain to be high for bin 1, which in turn may impact the
realization of any lower gains for the neighboring bins.
-Bounds the upper values for the echo suppression gains
for the higher bins to avoid any impact of the
external anti-aliasing filters.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2718993002
Cr-Commit-Position: refs/heads/master@{#16854}
Reason for revert:
Reland after fixing broken perf tests.
Original issue's description:
> Revert of Set scaling limit at 320 * 180 for all implementations. (patchset #2 id:20001 of https://codereview.webrtc.org/2709153002/ )
>
> Reason for revert:
> Looks like webrtc_perf_test started failing on linux, mac and windows after this cl landed.
>
> Example failure:
>
> https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1386/steps/webrtc_perf_tests/logs/stdio
>
> [ RUN ] CallPerfTest.ReceivesCpuOveruseAndUnderuse
> ../../webrtc/call/call_perf_tests.cc:522: Failure
> Value of: Wait()
> Actual: false
> Expected: true
> Timed out before receiving an overuse callback.
> [ FAILED ] CallPerfTest.ReceivesCpuOveruseAndUnderuse (120056 ms)
>
>
> Original issue's description:
> > Set scaling limit at 320 * 180 for all implementations.
> >
> > The MediaCodec decoder on android has trouble decoding video at
> > so low resolutions. We set the limit a bit higher for all implementations
> > pending a robust software fallback implementation for MediaCodec.
> >
> > BUG=webrtc:7206
> >
> > Review-Url: https://codereview.webrtc.org/2709153002
> > Cr-Commit-Position: refs/heads/master@{#16798}
> > Committed: 560ddb7321
>
> TBR=magjed@webrtc.org,sprang@webrtc.org,kthelgason@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7206
>
> Review-Url: https://codereview.webrtc.org/2711913007
> Cr-Commit-Position: refs/heads/master@{#16839}
> Committed: 37510bf094TBR=magjed@webrtc.org,sprang@webrtc.org,tommi@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7206
Review-Url: https://codereview.webrtc.org/2718013002
Cr-Commit-Position: refs/heads/master@{#16853}
Reason for revert:
I unfortunately have to revert this since trybots are still trying to run these tests and now they fail because the tests aren't there :-/ Before relanding, can you work with engprod to make sure we don't run them?
Original issue's description:
> Removes a test suite that is no longer used or maintained.
>
> BUG=NONE
>
> Review-Url: https://codereview.webrtc.org/2716843002
> Cr-Commit-Position: refs/heads/master@{#16825}
> Committed: bb5136f940TBR=solenberg@webrtc.org,nisse@webrtc.org,henrika@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2721433002
Cr-Commit-Position: refs/heads/master@{#16845}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
The issue was that VideoCapturerTrackSource was adding a reference
to itself, causing it to not be deleted even after no external objects
reference it. The objects underneath it (threads for instance) may
then be destroyed before the object dereferences them.
BUG=webrtc:6487
Review-Url: https://codereview.webrtc.org/2717023002
Cr-Commit-Position: refs/heads/master@{#16841}
In some rare circumstances, capturing a reference may be undesired. For
example, when creating an asynchronous task owned by the object itself,
the object may not need or want this task to keep itself alive.
BUG=None
Review-Url: https://codereview.webrtc.org/2711113008
Cr-Commit-Position: refs/heads/master@{#16840}
Reason for revert:
Looks like webrtc_perf_test started failing on linux, mac and windows after this cl landed.
Example failure:
https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1386/steps/webrtc_perf_tests/logs/stdio
[ RUN ] CallPerfTest.ReceivesCpuOveruseAndUnderuse
../../webrtc/call/call_perf_tests.cc:522: Failure
Value of: Wait()
Actual: false
Expected: true
Timed out before receiving an overuse callback.
[ FAILED ] CallPerfTest.ReceivesCpuOveruseAndUnderuse (120056 ms)
Original issue's description:
> Set scaling limit at 320 * 180 for all implementations.
>
> The MediaCodec decoder on android has trouble decoding video at
> so low resolutions. We set the limit a bit higher for all implementations
> pending a robust software fallback implementation for MediaCodec.
>
> BUG=webrtc:7206
>
> Review-Url: https://codereview.webrtc.org/2709153002
> Cr-Commit-Position: refs/heads/master@{#16798}
> Committed: 560ddb7321TBR=magjed@webrtc.org,sprang@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7206
Review-Url: https://codereview.webrtc.org/2711913007
Cr-Commit-Position: refs/heads/master@{#16839}
I'd like to make a change to rtc::Bind in another CL, and that will
be easier if there are fewer lines of code to modify.
BUG=None
Review-Url: https://codereview.webrtc.org/2719683002
Cr-Commit-Position: refs/heads/master@{#16838}
Created an Obj-C wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2513063003
Cr-Commit-Position: refs/heads/master@{#16835}
Reason for revert:
Reverting this change since it didn't affect the perf regressions we were seeing and actually seems to have caused more regressions as per comment in the bug.
Original issue's description:
> Use sched_yield on all POSIX platforms in PlatformThread.
> (not only MacOS)
>
> This is a test to see if perf regressions we're seeing may be related to the use of nanosleep().
>
> BUG=695438
> TBR=solenberg@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2716683002 .
> Cr-Commit-Position: refs/heads/master@{#16807}
> Committed: 384498abb5TBR=solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=695438
Review-Url: https://codereview.webrtc.org/2712133003
Cr-Commit-Position: refs/heads/master@{#16833}
Reason for revert:
Buildbot issues have been fixed.
Original issue's description:
> Revert of Remove some warning suppressions from SocketRocket. (patchset #1 id:1 of https://codereview.webrtc.org/2704383004/ )
>
> Reason for revert:
> Breaks buildbot
>
> Original issue's description:
> > Remove some warning suppressions from SocketRocket.
> >
> > These warnings started appearing on a clang update. This CL patches the
> > vendored library and removes the supression. We assert on the return as
> > we're not equipped to deal with failures there anyway.
> >
> > BUG=webrtc:6396
> > NOTRY=true
> >
> > Review-Url: https://codereview.webrtc.org/2704383004
> > Cr-Commit-Position: refs/heads/master@{#16820}
> > Committed: 49990e88fb
>
> TBR=kjellander@webrtc.org,magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6396
>
> Review-Url: https://codereview.webrtc.org/2714123002
> Cr-Commit-Position: refs/heads/master@{#16822}
> Committed: e47de1a69cTBR=kjellander@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6396
Review-Url: https://codereview.webrtc.org/2718653002
Cr-Commit-Position: refs/heads/master@{#16823}
Reason for revert:
Breaks buildbot
Original issue's description:
> Remove some warning suppressions from SocketRocket.
>
> These warnings started appearing on a clang update. This CL patches the
> vendored library and removes the supression. We assert on the return as
> we're not equipped to deal with failures there anyway.
>
> BUG=webrtc:6396
> NOTRY=true
>
> Review-Url: https://codereview.webrtc.org/2704383004
> Cr-Commit-Position: refs/heads/master@{#16820}
> Committed: 49990e88fbTBR=kjellander@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6396
Review-Url: https://codereview.webrtc.org/2714123002
Cr-Commit-Position: refs/heads/master@{#16822}
These warnings started appearing on a clang update. This CL patches the
vendored library and removes the supression. We assert on the return as
we're not equipped to deal with failures there anyway.
BUG=webrtc:6396
NOTRY=true
Review-Url: https://codereview.webrtc.org/2704383004
Cr-Commit-Position: refs/heads/master@{#16820}
This avoids adding an additional test target. Plus, everything in
rtc_api_unittests is (and likely will be) simple utility classes akin to
what's already being tested in rtc_unittests.
BUG=None
TBR=kjellander@webrtc.org
Review-Url: https://codereview.webrtc.org/2709573003
Cr-Commit-Position: refs/heads/master@{#16819}
A bug has been reported to complaint the ScreenCapturerWinDirectx cannot capture
the first frame, which is used in "Report an issue" page.
A simple solution here is to skip the first frame.
This change also removes the friend relationship between
DxgiDuplicatorController / DxgiAdapterDuplicator / DxgiOutputDuplicator, which
is not really necessary.
BUG=682112
Review-Url: https://codereview.webrtc.org/2703123002
Cr-Commit-Position: refs/heads/master@{#16815}
The documentation for AsyncInvoker states that it owns the lifetime of
calls, and when its destructor is called, all in-flight calls are
cancelled or finish executing. The "cancelled" part is working, but if
a call is in the middle of executing, the destructor does *not* wait.
This is fixed by keeping a count of pending invocations, which is
decremented when a call is either cleared from a message queue or
finishes executing.
BUG=webrtc:3914, webrtc:3911
Review-Url: https://codereview.webrtc.org/2694723004
Cr-Commit-Position: refs/heads/master@{#16811}
This is a short term solution to change the volume of a RTCAudioTrack (which contains an RTCAudioSource property) until applyConstraints for RTCMediaStreamTracks has been implemented.
This CL adds one new Objective-C method to AudioSourceInterface's wrapper: -(void)setVolume:(double)volume
BUG=webrtc:6533, webrtc:6805
This is my first CL for Chromium/WebRTC, so please let me know if I did something wrong.
Review-Url: https://codereview.webrtc.org/2534843002
Cr-Commit-Position: refs/heads/master@{#16809}
When textures are not enabled and we are using byte buffer outputs, the
decoder is currently crashing for odd heights because of an RTC_CHECK.
This CL removes the check and handles the pointer offset to the chroma
planes for the odd height case instead.
This has been verified to work correctly on a Pixel device.
BUG=webrtc:6651
Review-Url: https://codereview.webrtc.org/2709923005
Cr-Commit-Position: refs/heads/master@{#16805}
Explicit initialization of const member of new EchoCanceller 3
submodule.
NOTRY=True
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2715573003
Cr-Commit-Position: refs/heads/master@{#16802}
calculating frame sizes. Added actual_bitrate metric which also accounts
for TL and SL info. Metric encoded_frame_size calculation is cleaned up. Perf alerts should be ignored.
BUG=webrtc:7095
Review-Url: https://codereview.webrtc.org/2709483009
Cr-Commit-Position: refs/heads/master@{#16800}
The MediaCodec decoder on android has trouble decoding video at
so low resolutions. We set the limit a bit higher for all implementations
pending a robust software fallback implementation for MediaCodec.
BUG=webrtc:7206
Review-Url: https://codereview.webrtc.org/2709153002
Cr-Commit-Position: refs/heads/master@{#16798}
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/
webrtc::PacedSender::Process <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- this CL end here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
Also adds a basic unit test for VP9 implementation.
BUG=webrtc:6541
Review-Url: https://codereview.webrtc.org/2654813002
Cr-Commit-Position: refs/heads/master@{#16795}
This CL harmonizes and improves the naming of the config structs
in the VideoProcessor tests. It should have no functional implications.
CodecConfigPars -> CodecParams:
This struct mainly contains codec settings.
QualityMetrics -> QualityThresholds:
This struct contains thresholds against which the calculated
PSNR and SSIM metrics are compared to.
RateControlMetrics -> RateControlThresholds:
This struct contains thresholds against which the calculated
rate control metrics are compared to.
BUG=webrtc:6634
Review-Url: https://codereview.webrtc.org/2703333004
Cr-Commit-Position: refs/heads/master@{#16794}
