This moves some GN check configurations out of .gn to individual
targets.
The now checked target is:
"//webrtc/modules/audio_conference_mixer/*"
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2593003002
Cr-Commit-Position: refs/heads/master@{#15759}
Wait until first frame is decoded to avoid include zeros in stats.
BUG=b/32659204
Review-Url: https://codereview.webrtc.org/2582313002
Cr-Commit-Position: refs/heads/master@{#15752}
Reason for revert:
Breaks Chromium WebRTC FYI bots:
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Builder/builds/12337
The error was masked by another breaking change that was committer earlier. This is the first build showing the error.
Original issue's description:
> Make P2PTransportChannel inherit from IceTransportInternal.
>
> Make P2PTransportChannel inherit from IceTransportInternal instead of
> TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
> be separated from P2PTransportChannel.
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2590063002
> Cr-Commit-Position: refs/heads/master@{#15743}
> Committed: 12749d89d9TBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2594343002
Cr-Commit-Position: refs/heads/master@{#15751}
Created a java 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/2513723002
Cr-Commit-Position: refs/heads/master@{#15745}
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.
BUG=none
Review-Url: https://codereview.webrtc.org/2590063002
Cr-Commit-Position: refs/heads/master@{#15743}
Problem fixed: RTP header extensions were not properly set in tests.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2593963003
Cr-Commit-Position: refs/heads/master@{#15741}
Modify ExpectReportContainsCertificateInfo to use EXPECT_EQ checks of
RTCCertificateStats objects.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2594553003
Cr-Commit-Position: refs/heads/master@{#15738}
Remove ExpectReportContainsCandidate in favor of EXPECT_EQ checks of
RTC[Local/Remote]IceCandidateStats objects.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2594753002
Cr-Commit-Position: refs/heads/master@{#15737}
Prior to this change, the receiver_time metric had huge outliers
whenever FlexFEC was enabled. This was due to a measurement problem,
where the time of the incoming packet was incorrectly set to zero.
This happened for packets that were lost in transit, but recovered
through FEC.
This CL fixes this problem by simply not recording samples where the
incoming packet time is undefined. The CL also removes the possibility
of timestamp collisions in the data structures.
TESTED=Ran './webrtc_perf_tests --gtest_filter="*ForemanCifPlr5H264Flexfec*" | grep receiver_time' locally 10 times, without experiencing any outliers.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2596793002
Cr-Commit-Position: refs/heads/master@{#15735}
Reason for revert:
This CL broke some buildbots. I will investigate it later.
Original issue's description:
> Refactor webrtc/modules/video_processing for GN check
>
> This moves some GN check configurations out of .gn to individual
> targets.
>
> The now checked target is:
> "//webrtc/modules/video_processing/*"
>
> BUG=webrtc:6828
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2595543002
> Cr-Commit-Position: refs/heads/master@{#15732}
> Committed: 00a810b844TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6828
Review-Url: https://codereview.webrtc.org/2594973002
Cr-Commit-Position: refs/heads/master@{#15733}
This moves some GN check configurations out of .gn to individual
targets.
The now checked target is:
"//webrtc/modules/video_processing/*"
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2595543002
Cr-Commit-Position: refs/heads/master@{#15732}
Remove ExpectReportContainsDataChannel in favor of EXPECT_EQ checks of
RTCDataChannelStats objects.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2597433002
Cr-Commit-Position: refs/heads/master@{#15731}
Previously it was allowed to call AddStats with stats of the same ID
multiple times.
This revealed a few things:
- Local and remote streams can have the same label.
RTCMediaStreamStats's ID is updated to include "local"/"remote".
- The same certificate can show up multiple times (e.g. for local and
remote in a loopback), so we skip creating RTCCertificateStats for the
same certificate multiple times
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2593503003
Cr-Commit-Position: refs/heads/master@{#15730}
This moves some GN check configurations out of .gn to individual
targets.
The now checked target is:
"//webrtc/modules/pacing/*"
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2594523003
Cr-Commit-Position: refs/heads/master@{#15729}
This moves some GN check configurations out of .gn to individual
targets.
The now checked target is:
"//webrtc/modules/media_file/*"
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2593693002
Cr-Commit-Position: refs/heads/master@{#15726}
This moves some GN check configurations out of .gn to individual
targets.
The now checked target is:
"//webrtc/modules/desktop_capture/*"
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2593713002
Cr-Commit-Position: refs/heads/master@{#15725}
The CGDisplayStream API returns rects in physical pixel coordinates, not
Density-Independent Pixel coordinates. The code was incorrectly re-applying the
dip_to_pixel scaling.
BUG=chromium:675490
Review-Url: https://codereview.webrtc.org/2588973002
Cr-Commit-Position: refs/heads/master@{#15720}
that the level of the output in the audio processing
module is monitored. This CL adds that.
BUG=webrtc:6181, webrtc:6183, webrtc:6220
Review-Url: https://codereview.webrtc.org/2549143004
Cr-Commit-Position: refs/heads/master@{#15718}
Chromium has now been updated, so we can remove the base headers from
rtc_media.
BUG=None
Review-Url: https://codereview.webrtc.org/2590813002
Cr-Commit-Position: refs/heads/master@{#15712}
The new name, OnCompleteReports rather than OnComplete, is needed
because in C++ method lookup, overriding a method hides all otherwise
inherited methods with the same name, even if they have a different
signature. And here, the intention is that each subclass should
override one or the other of the two methods, and inherit the method it
doesn't override.
This cl is a prerequisite for
https://codereview.webrtc.org/2567143003/, because the Chrome glue
code needs to retain the stats report after the OnComplete method has
returned.
Currently, Chrome makes a copy of the stats mapping (which breaks when
changing ValuePtr from an rtc::linked_ptr to an std::unique_ptr). After
this cl, Chrome can be fixed to take ownership and no longer needs to
copy anything, unblocking cl 2567143003.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2584553002
Cr-Commit-Position: refs/heads/master@{#15708}
Documentation was also unclear, it seems it returned the RTP packet
size including RTP headers.
BUG=None.
Review-Url: https://codereview.webrtc.org/2588343002
Cr-Commit-Position: refs/heads/master@{#15707}
Rename variables and private functions to follow style,
replace remaining asserts with DCHECKs.
add 'ms' suffix to time variables derived from clock_
add 'ntp' suffix to time variables derived from ntp time.
No functional changes expected.
BUG=None
Review-Url: https://codereview.webrtc.org/2588753002
Cr-Commit-Position: refs/heads/master@{#15706}
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.
Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 1) The policy is always explicitly initialized to secure.
> 2) API users have to explicitly integrate with the feature to
> use it, and will otherwise get no change in behavior.
> 3) The feature is not immediately exposed in non-native
> contexts. For example, disabling of certificate validation
> is not implemented via URI parsing since this would
> immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9eTBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
The test verifies that the hysteresis window in the configuration works
as intended.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2594563002
Cr-Commit-Position: refs/heads/master@{#15700}
No need to pass a whole struct around, when only one member is used.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2589833002
Cr-Commit-Position: refs/heads/master@{#15687}
Reason for revert:
breaks downstream project.
Can you make this change in a compatible way using anonymous union:
union {
bool is_first_packet_in_frame;
RTC_DEPRECATED bool isFirstPacket;
};
(unfortunetly this this treak breaks braced initialization in rtp_rtcp_impl_unittest.cc,
so that should be rewritting in a more classic way)
Original issue's description:
> Rename RTPVideoHeader.isFirstPacket to .is_first_packet_in_frame.
>
> Name should represent the actual meaning.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2574943003
> Cr-Commit-Position: refs/heads/master@{#15684}
> Committed: efde908380TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None
Review-Url: https://codereview.webrtc.org/2589783003
Cr-Commit-Position: refs/heads/master@{#15686}
"WebRTC.Call.NumberOfPauseEvents" -> "WebRTC.Video.NumberOfPauseEvents"
Recorded if a certain time has passed (10 sec) since the first media packet was sent.
Moved to per stream to know when media has started and to prevent logging stats for calls that was never in use.
Add histogram for percentage of paused video time for sent video streams:
"WebRTC.Video.PausedTimeInPercent"
BUG=b/32659204
Review-Url: https://codereview.webrtc.org/2530393003
Cr-Commit-Position: refs/heads/master@{#15681}
Reason for revert:
Bug affecting perf tests has been fixed. The issue was that I had accidentally disabled cpu overuse adaptation based on the encoders ScalingSettings, not just quality-based scaling.
Original issue's description:
> Revert of Properly report number of quality downscales in stats. (patchset #11 id:220001 of https://codereview.webrtc.org/2564373002/ )
>
> Reason for revert:
> Breaks perf tests
>
> Original issue's description:
> > Properly report number of quality downscales in stats.
> >
> > A regression was introduced in 876222f that caused these stats to
> > be reported incorrectly. This used to be only implemented for VP8
> > but should now be available for all codecs.
> >
> > BUG=webrtc:6860
> >
> > Review-Url: https://codereview.webrtc.org/2564373002
> > Cr-Commit-Position: refs/heads/master@{#15673}
> > Committed: 0c8c538835
>
> TBR=asapersson@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2586783003
> Cr-Commit-Position: refs/heads/master@{#15678}
> Committed: fe04bd43ccTBR=asapersson@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6860
Review-Url: https://codereview.webrtc.org/2588743002
Cr-Commit-Position: refs/heads/master@{#15680}
This effectively reverts commit c3e1cabc696240e4b5a128653264785292878205
(https://codereview.webrtc.org/2589703002/).
The reason the test was failing before was missing resource
dependencies in the GN file. This is now fixed.
Furthermore, the test did not trigger the complexity adaptation that
it was supposed to test, since the hysteresis window of the bitrate
was not taken into account. This is also fixed.
Finally, a percent label was added to a printout, to match the same
printout in the other test.
BUG=webrtc:6708
Review-Url: https://codereview.webrtc.org/2580383002
Cr-Commit-Position: refs/heads/master@{#15679}
Reason for revert:
Breaks perf tests
Original issue's description:
> Properly report number of quality downscales in stats.
>
> A regression was introduced in 876222f that caused these stats to
> be reported incorrectly. This used to be only implemented for VP8
> but should now be available for all codecs.
>
> BUG=webrtc:6860
>
> Review-Url: https://codereview.webrtc.org/2564373002
> Cr-Commit-Position: refs/heads/master@{#15673}
> Committed: 0c8c538835TBR=asapersson@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6860
Review-Url: https://codereview.webrtc.org/2586783003
Cr-Commit-Position: refs/heads/master@{#15678}
I always find myself re-writing this function for debug purposes. It'd
save so much time if it already existed...
BUG=none
Review-Url: https://codereview.webrtc.org/2589773002
Cr-Commit-Position: refs/heads/master@{#15677}
Reason for revert:
Still hitting NOTREACHED.
Original issue's description:
> Revert of Disabling NOTREACHED which we're hitting flakily in browser tests. (patchset #1 id:1 of https://codereview.webrtc.org/2477663002/ )
>
> Reason for revert:
> To see if the NOTREACHED is still hit.
>
> Original issue's description:
> > Disabling NOTREACHED which we're hitting flakily in browser tests.
> >
> > I have no idea how bad it is that we're hitting this limit; I'm just
> > doing this to stop the tests from flaking.
> >
> > BUG=webrtc:6484
> >
> > Committed: https://crrev.com/6eaa55867b449df992752c1df540ec42f9d9b057
> > Cr-Commit-Position: refs/heads/master@{#14974}
>
> TBR=stefan@webrtc.org,phoglund@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6484
>
> Review-Url: https://codereview.webrtc.org/2585183002
> Cr-Commit-Position: refs/heads/master@{#15665}
> Committed: 9d7ea0920cTBR=stefan@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6484
Review-Url: https://codereview.webrtc.org/2585273002
Cr-Commit-Position: refs/heads/master@{#15676}
RTCStatsCollector relies on PeerConnection and its WebRtcSession. If the
PeerConnection is destroyed, reference counting keeps the
RTCStatsCollector alive until the request has completed. But the request
is using PeerConnection/WebRtcSession resources that are destroyed in
~PeerConnection().
To get around this problem, RTCStatsCollector::WaitForPendingRequest()
is added, which is invoked at ~PeerConnection().
Integration test added, it caused a segmentation fault before this
change / EXPECT failure.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2583613003
Cr-Commit-Position: refs/heads/master@{#15674}
