Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
These methods are not used by the new AndroidVideoTrackSource API.
Review-Url: https://codereview.webrtc.org/2280873002
Cr-Commit-Position: refs/heads/master@{#14036}
Using a timestamp based on a timer that is monotonically increasing for
the cache, so that cache's freshness can be checked regardless of if
system clock is modified.
Using a system clock for the stats' timestamp, which needs to be
relative to UNIX epoch (Jan 1, 1970, UTC).
This CL removes the dependency on faketiming.h.
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2299643002
Cr-Commit-Position: refs/heads/master@{#13997}
Reason for revert:
Downstream apps should now be fixed.
Original issue's description:
> Revert of Remove the old AndroidVideoCapturer stack code. (patchset #2 id:20001 of https://codereview.webrtc.org/2235893003/ )
>
> Reason for revert:
> Breaks downstream.
>
> Original issue's description:
> > Remove the old AndroidVideoCapturer stack code.
> >
> > This code is no longer needed. Apps should be using the new API introduced here: https://codereview.webrtc.org/2127893002/
> >
> > Committed: https://crrev.com/1b365a8db070f9cdcbf35ec871f758dcd909e51d
> > Cr-Commit-Position: refs/heads/master@{#13950}
>
> TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/e39f251dacf66e50153bcda615f06b7c59e5856b
> Cr-Commit-Position: refs/heads/master@{#13958}
TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
Review-Url: https://codereview.webrtc.org/2298063003
Cr-Commit-Position: refs/heads/master@{#13988}
hbos and hta are webrtc/stats/ OWNERS. Public api headers relating to
rtcstats are placed in webrtc/api/ and implementations are placed in
webrtc/stats/. This ownership allows the rtcstats owners to own both .cc
and .h files.
For example, rtcstats.[h/cc] and rtcstatsreport.[h/cc].
(Soon there will also be rtcstats_objects.[h/cc] and more.)
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2294693002
Cr-Commit-Position: refs/heads/master@{#13981}
This is the stats collector for the new stats types, RTCStats[1] and
RTCStatsReport[2]. It so far only produces RTCPeerConnectionStats[3] as
an example of how it would collect stats. Each RTCStats subclass will
get a corresponding RTCStatsCollector::ProduceFooStats().
Stats reports are cached and returned as const references (ref
counting). This allows stats to be inspected by multiple observers and
across multiple threads. No copies will have to be made when surfacing
this to Blink or other places.
The current implementation of ProducePeerConnectionStats() only look at
existing DataChannels. This might be incorret if data channels can be
removed? Will investigate in a follow-up, crbug.com/636818.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#idl-def-rtcstats
[2] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
[3] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html#pcstats-dict*
BUG=chromium:627816, chromium:636818
Review-Url: https://codereview.webrtc.org/2242043002
Cr-Commit-Position: refs/heads/master@{#13979}
Reason for revert:
Breaks downstream.
Original issue's description:
> Remove the old AndroidVideoCapturer stack code.
>
> This code is no longer needed. Apps should be using the new API introduced here: https://codereview.webrtc.org/2127893002/
>
> Committed: https://crrev.com/1b365a8db070f9cdcbf35ec871f758dcd909e51d
> Cr-Commit-Position: refs/heads/master@{#13950}
TBR=magjed@webrtc.org,glaznev@webrtc.org,kjellander@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/2291583002
Cr-Commit-Position: refs/heads/master@{#13958}
This would make it possible to gather stats on multiple threads, store
the results in multiple reports and to merge the results.
Added rtcstatsreport_unittest.cc, moving a RTCStatsReport-related test
from rtcstats_unittest.cc. Added more unittests covering the order of
stats and TakeMembersFrom.
Also changed RTCStatsReport[] to RTCStatsReport::Get to avoid
confusion with other usages of the [] operator.
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2278433003
Cr-Commit-Position: refs/heads/master@{#13957}
Log the DTLS handshake error code in OpenSSLStreamAdapter.
Forward the error code to WebRTCSession with the Signals.
This part is only for the WebRTC native code.
To make it work, need another CL for Chromium.
BUG=webrtc:5959
Review-Url: https://codereview.webrtc.org/2167363002
Cr-Commit-Position: refs/heads/master@{#13940}
The code that extracts certificate stats from an SSLCertificate and its
certificate chain is moved into SSLCertificate::GetStats. The stats
collector code loops through the resulting SSLCertificateStats and
creates the StatsReports for those stats.
This will allow the new stats collector to reuse GetStats in a future
CL.
BUG=chromium:627816, chromium:629436
Review-Url: https://codereview.webrtc.org/2259283002
Cr-Commit-Position: refs/heads/master@{#13917}
I found that, depending on when it's called, ClearGettingPorts may or
may not signal CandidatesAllocationDone, and may or may not continue
to gather more ports/candidates.
I'm fixing this inconsistency by having it always signal
CandidatesAllocationDone (if needed), and always stop gathering until
the next network change event. This makes it equivalent to
StopGettingPorts, except that it allows gathering to be restarted if
a network change occurs.
I also found that P2PTransportChannel was signaling "gathering
complete" even when continual gathering was enabled. This wasn't caught
by the unit tests due to the inconsistency of ClearGettingPorts as
described above.
Review-Url: https://codereview.webrtc.org/2124283003
Cr-Commit-Position: refs/heads/master@{#13908}
Normally, when creating a data channel with an out-of-range ID,
createDataChannel returns nullptr. But due to an off-by-one
error, creating a data channel with ID 1023 returns a data channel
that silently fails later.
This probably occurred because it wasn't clear whether "kMaxSctpSid" was an
inclusive or exclusive maximum, so I changed the value to
"kMaxSctpStreams". This wasn't caught by unit tests because the
off-by-one error persisted to the unit tests as well.
Also getting rid of some dead code. We were adding SCTP streams to the
ContentDescription object but they weren't being used.
BUG=619849
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2254003002 .
Cr-Commit-Position: refs/heads/master@{#13906}
The old and new getStats are very different. This CL proposes rewriting
the new getStats from scratch with a bottom-up approach, starting with
the fundamental stats classes. This will allow cleaner and more
efficient code that is more aligned with the spec.
RTCStats and subclasses are the equivalent to RTCStats and RTCStats-
-derived dictionaries from the specs[1][2]. The dictionary members are
public member variables of type RTCStatsMember<T>, where T is one of the
supported types. All members derive from RTCStatsMemberInterface and
iteration of members is possible with RTCStats::Members().
The members are not stored in a map for performance and readability.
Type checking is supported with static class variables, kType.
Only the supported member types T are specialized and may be
instantiated, and sequences are supported with std::vector<...>. Type
checking is again supported with static class variables, kType.
RTCStatsReport is the equivalent from the spec[3], and maps RTCStats::id
to RTCStats-objects. RTCStatsReport is reference counted. It and its
contained stats may be destroyed on any thread. When the
RTCStatsCollector is added in a follow-up CL, it will return const
references to the RTCStatsReports. This means copies don't have to be
made for multiple stats observers or when jumping threads. In fact, no
copies of any stats will have to be made in surfacing stats to Blink.
[1] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstats-dictionary
[2] https://w3c.github.io/webrtc-stats/archives/20160526/webrtc-stats.html
[3] https://www.w3.org/TR/2016/WD-webrtc-20160531/#rtcstatsreport-object
This adds the new folder webrtc/stats/, with target rtc_stats and binary
rtc_stats_unittests. Public api headers are placed in webrtc/api/ and
.cc files are placed in webrtc/stats/.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2241093002
Cr-Commit-Position: refs/heads/master@{#13879}
Reason for revert:
Breaks chromium.
Original issue's description:
> Add field_trial_default dependency to libjingle_peerconnection
>
> This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
>
> NOTRY=True
>
> Committed: https://crrev.com/a7a01df2aebe7108afad208ccd0341c2f0bc7b3b
> Cr-Commit-Position: refs/heads/master@{#13836}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org,kjellander@webrtc.org,arlolra@gmail.com
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2263063002
Cr-Commit-Position: refs/heads/master@{#13837}
This is needed for webrtc::field_trial::FindFullName in peerconnection.cc
NOTRY=True
Review-Url: https://codereview.webrtc.org/2120673004
Cr-Commit-Position: refs/heads/master@{#13836}
This allows to at least get a camera error back if the camera thread freezes. Application can use this as a signal to restart the program.
R=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2257123002
Cr-Commit-Position: refs/heads/master@{#13830}
New files, classes moved from statscollector_unittest.cc:
+webrtc/api/test/mock_peerconnection.h
for MockPeerConnectionFactory and MockPeerConnection
+webrtc/api/test/mock_webrtcsession.h
for MockWebRtcSession
+webrtc/media/base/test/mock_mediachannel.h
for MockVideoMediaChannel and MockVoiceMediaChannel
The webrtc/media/base/test folder is new.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2238933002
Cr-Commit-Position: refs/heads/master@{#13769}
This is in preparation for adding a gn target for audio_device_tests.
BUG=webrtc:6170,webrtc:163
NOTRY=True
Review-Url: https://codereview.webrtc.org/2222563002
Cr-Commit-Position: refs/heads/master@{#13768}
when building with default warnings.
This is in preparation for making a gn target for audio_device_tests.
BUG=webrtc:6170, webrtc:163
NOTRY=True
Review-Url: https://codereview.webrtc.org/2219653004
Cr-Commit-Position: refs/heads/master@{#13759}
If the data transport is destroyed while data is buffered (due to
the PC being closed, or a description set with data rejected), the
data channel was getting stuck in a "closing" state, waiting to
finish sending its buffered data. But since there's no more transport,
it will never get another chance to send buffered data.
It just needs to terminate non-gracefully and discard the buffered data
in this situation.
R=skvlad@webrtc.org, zhihuang@webrtc.org
Review URL: https://codereview.webrtc.org/2235843003 .
Cr-Commit-Position: refs/heads/master@{#13737}
