New file structure and targets:
rtc_stats_api
webrtc/api/stats/rtcstats.h
webrtc/api/stats/rtcstats_objects.h
webrtc/api/stats/rtcstatsreport.h
rtc_stats (dep on rtc_stats_api)
webrtc/stats/rtcstats.cc
webrtc/stats/rtcstats_objects.cc
webrtc/stats/rtcstatsreport.cc
libjingle_peerconnection (dep on rtc_stats)
webrtc/api/rtcstatscollector.cc
webrtc/api/rtcstatscollector.h
Placing rtc_stats_api headers in this separate target instead of
libjingle_peerconnection avoids a circular dependency
libjingle_peerconnection -> rtc_stats -> libjingle_peerconnection
Code changes:
PeerConnectionInterface::GetStats(RTCStatsCollectorCallback*) added for
the new stats collection API. Implemented by PeerConnection.
BUG=chromium:627816
Review-Url: https://codereview.webrtc.org/2331373004
Cr-Commit-Position: refs/heads/master@{#14246}
std::string is all we need. const char* is an annoying special case
because they can't be compared with ==. Having two different string
types was a premature optimization.
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2340303002
Cr-Commit-Position: refs/heads/master@{#14235}
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
During GN vs GYP auditing it was discovered that some
GN targets that had public_configs were not exposing them
to dependents where the dependent depended on a group, which
in turn included that target as a dependency. Instead of
changing those public_configs to all_dependent_configs
(which would be a change from GYP), it's better to just change
those group targets to use public_deps instead.
BUG=webrtc:6323
NOTRY=True
TESTED=Generated GYP and GN project files on Mac and ran the
tools/gyp_flag_compare.py script before and after this patch was
applied. The file in question used for inspection was the
webrtc/api/webrtcsessiondescriptionfactory.cc
which is a part of the libjingle_peerconnection target.
Review-Url: https://codereview.webrtc.org/2344623002
Cr-Commit-Position: refs/heads/master@{#14222}
Camera id doesn't really exist for Camera2. Changing onCameraOpening to
take a string instead removes ugly code.
BUG=webrtc:6325
R=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2331013002
Cr-Commit-Position: refs/heads/master@{#14212}
This CL appends the EGL error code in exceptions after an EGL function
fails. This information is helpful when debugging.
BUG=webrtc:6350
Review-Url: https://codereview.webrtc.org/2338033002
Cr-Commit-Position: refs/heads/master@{#14208}
Also enforce a minimum inter-frame interval of 1 ms,
fix a bug in the clipping logic, and improve comments.
BUG=webrtc:5740
Review-Url: https://codereview.webrtc.org/2325563002
Cr-Commit-Position: refs/heads/master@{#14206}
The method can be used to print the stack trace of the camera thread in
error conditions.
BUG=webrtc:6148
Review-Url: https://codereview.webrtc.org/2332693002
Cr-Commit-Position: refs/heads/master@{#14187}
This CL optimizes the Android capture NV12 -> I420 + scaling code. For
example, when the input is 1280x720 and we adapt to 640x360, this CL:
- Reduces conversion time from 3.37 ms to 1.46 ms.
- Reduces memory footprint by 1 MB.
BUG=webrtc:6319
Review-Url: https://codereview.webrtc.org/2317443003
Cr-Commit-Position: refs/heads/master@{#14167}
I have written a large part of the code in these files and I feel I
should be an OWNER of them.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2322983002
Cr-Commit-Position: refs/heads/master@{#14148}
Only has counts stats right now but enumeration stats can easily be added in the future if needed.
BUG=webrtc:6313
Review-Url: https://codereview.webrtc.org/2320473002
Cr-Commit-Position: refs/heads/master@{#14146}
libjingle_peerconnection_so is not including common_config, which is
causing some differences is the defines.
We'd like to prevent that happening in the future.
NOTRY=True
BUG=webrtc:5949
Review-Url: https://codereview.webrtc.org/2325603002
Cr-Commit-Position: refs/heads/master@{#14127}
Exynos VP8 HW encoder may generate a bitrate which deviates too
much from target value causing frame drops and fps reduction or
BWE stuck at low values.
Add one more option to bitrate adjustment in HW encoder wrapper
which allows to track and dynamicaly scale bitrate used for
codec configuration.
BUG=b/30951236
R=wzh@webrtc.org
Review URL: https://codereview.webrtc.org/2308843002 .
Cr-Commit-Position: refs/heads/master@{#14095}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
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}
