RTCStatsMemberInterface::Type's kBool and kSequenceBool.
This means that RTCStats-derived classes ("RTCStats-derived
dictionaries"[1]) can contain boolean and sequence of boolean members.
[1] https://w3c.github.io/webrtc-stats/
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2387343002
Cr-Commit-Position: refs/heads/master@{#14509}
hbos and hta are already OWNERS of webrtc/stats/ and of rtcstats* files
(per-file rtcstats*=) in webrtc/api/. When the webrtc/api/stats/ folder
was created we forgot to add this OWNERS file (per-file OWNERS does not
apply to subfolders apparently).
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2392633002
Cr-Commit-Position: refs/heads/master@{#14482}
Changing this constant has been empirically proven to solve the issue.
BUG=webrtc:6455,b/31827852
Review-Url: https://codereview.webrtc.org/2382733006
Cr-Commit-Position: refs/heads/master@{#14479}
The former is always defined (by webrtc/base/checks.h) to either 0 or
1, whereas the latter isn't necessarily defined.
NOTRY=true
BUG=webrtc:6451
Review-Url: https://codereview.webrtc.org/2384693002
Cr-Commit-Position: refs/heads/master@{#14474}
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.
Original cl description:
Let ViEEncoder handle resolution changes.
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.
With this change, many variables in WebRtcVideoSendStream no longer need to be locked.
BUG=webrtc:5687, webrtc:6371, webrtc:5332
Review-Url: https://codereview.webrtc.org/2351633002
Cr-Commit-Position: refs/heads/master@{#14445}
Remove onOutputFormatRequest from the VideoCapturer interface and from
all implementations of that interface. Apps should now use
VideoSource.adaptOutputFormat() instead.
BUG=webrtc:6391
Review-Url: https://codereview.webrtc.org/2373353002
Cr-Commit-Position: refs/heads/master@{#14428}
This added an SCTP codec, which is later re-interpreted as a video
codec. We shouldn't be adding codecs that don't match the type of the
media description.
BUG=chromium:648062
Review-Url: https://codereview.webrtc.org/2354723002
Cr-Commit-Position: refs/heads/master@{#14421}
Also provide a new set of thresholds for the VideoToolbox encoder. The new thresholds were experimentally determined to work well on the iPhone 6S, and also adequately on the iPhone 5S.
BUG=webrtc:5678
Review-Url: https://codereview.webrtc.org/2309743002
Cr-Commit-Position: refs/heads/master@{#14420}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
The old method is marked deprecated and will be removed soon. The new
method checks if all cameras on the device support better than legacy
implementation of the camera2 API.
BUG=webrtc:6390
Review-Url: https://codereview.webrtc.org/2354883002
Cr-Commit-Position: refs/heads/master@{#14350}
The Java VideoSource class wraps the C++ AndroidVideoTrackSource.
AndroidVideoTrackSource is the object actually owning the VideoAdapter.
We currently control the VideoAdapter through the Java VideoCapturer,
but it is more natural and direct to control it through the Java
VideoSource class. This CL adds the necessary function to do this, and
the function in VideoCapturer is deprecated.
BUG=webrtc:6391
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2350933006 .
Cr-Commit-Position: refs/heads/master@{#14332}
Reason for revert:
Fix bugs causing the new code to fail.
Original issue's description:
> Revert of Release camera statistics after switching camera on CameraCapturer. (patchset #1 id:1 of https://codereview.webrtc.org/2353263002/ )
>
> Reason for revert:
> Breaks bots.
>
> Original issue's description:
> > Release camera statistics after switching camera on CameraCapturer.
> >
> > BUG=webrtc:6397
> > TBR=magjed@webrtc.org
> > NOTRY=True
> >
> > Committed: https://crrev.com/d5e9237b303e5fe253dc6530fbcf939921f4eaed
> > Cr-Commit-Position: refs/heads/master@{#14323}
>
> TBR=magjed@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6397
>
> Committed: https://crrev.com/ad5d65845f5c859d0564811a4eea68e1efdb8450
> Cr-Commit-Position: refs/heads/master@{#14324}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6397
Review-Url: https://codereview.webrtc.org/2357893002
Cr-Commit-Position: refs/heads/master@{#14330}
Reason for revert:
Breaks bots.
Original issue's description:
> Release camera statistics after switching camera on CameraCapturer.
>
> BUG=webrtc:6397
> TBR=magjed@webrtc.org
> NOTRY=True
>
> Committed: https://crrev.com/d5e9237b303e5fe253dc6530fbcf939921f4eaed
> Cr-Commit-Position: refs/heads/master@{#14323}
TBR=magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6397
Review-Url: https://codereview.webrtc.org/2353163003
Cr-Commit-Position: refs/heads/master@{#14324}
These are no longer used in Chromium, so deleting them will not break any
third party project.
BUG=chromium:627816
NOTRY=True
Review-Url: https://codereview.webrtc.org/2352993002
Cr-Commit-Position: refs/heads/master@{#14309}
This change adds a new statistic for logging how many calls to
NetEq::GetAudio resulted in a "muted output". A muted output happens
if the packet stream has been dead for some time (and the last decoded
packet was not comfort noise).
BUG=webrtc:5606
BUG=b/31256483
Review-Url: https://codereview.webrtc.org/2341293002
Cr-Commit-Position: refs/heads/master@{#14302}
In the new implementation session reports all events through
CameraSession.Events interface. CameraCapturer passes these events
forward.
BUG=webrtc:6325
Review-Url: https://codereview.webrtc.org/2331343010
Cr-Commit-Position: refs/heads/master@{#14286}
Reason for revert:
Import breakage has been fixed.
Original issue's description:
> Revert of Optimize Android NV12 capture (patchset #2 id:20001 of https://codereview.webrtc.org/2317443003/ )
>
> Reason for revert:
> Import breakage in g3.
>
> Original issue's description:
> > Optimize Android NV12 capture
> >
> > 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
> >
> > Committed: https://crrev.com/36d38cbb153e19bdc3c62a750aba6889da40aac2
> > Cr-Commit-Position: refs/heads/master@{#14167}
>
> TBR=sakal@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6319
TBR=sakal@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6319
Review-Url: https://codereview.webrtc.org/2332213011
Cr-Commit-Position: refs/heads/master@{#14273}
This name is only used for a DCHECK. Having it as a parameter leads to unnecessary string copying even on release builds.
This CL instead adds a GetHistogramName function that is only called on debug builds.
Checking if pointer is null is also moved outside HistogramAdd function. Having it there is confusing since Chromium implementation doesn't have it there.
BUG=webrtc:6329
Review-Url: https://codereview.webrtc.org/2337883003
Cr-Commit-Position: refs/heads/master@{#14263}
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}
