Move the following targets into webrtc/video/BUILD.gn:
* screenshare_loopback
* video_quality_test
* video_loopback
Add new target 'run_tests' in webrtc/test/BUILD.gn, being used by two
of the above and make then depend on that instead.
BUG=webrtc:6440
NOTRY=True
Review-Url: https://codereview.webrtc.org/2438973002
Cr-Commit-Position: refs/heads/master@{#14735}
Specifically set max_len to 2000, to simulate multi-packet insertions.
BUG=webrtc:5654
NOTRY=true
Review-Url: https://codereview.webrtc.org/2391263002
Cr-Commit-Position: refs/heads/master@{#14656}
Reason for revert:
Speculative revert.
Intermittent memory access errors suspected to be caused by this cl.
See for instance https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/8018
UNADDRESSABLE ACCESS of freed memory: reading 0x0331d330-0x0331d334 4 byte(s)
# 0 webrtc::voe::RtcpRttStatsProxy::LastProcessedRtt
# 1 webrtc::ModuleRtpRtcpImpl::Process
Original issue's description:
> Add RtcpRttStats to AudioStream
>
> BUG=webrtc:6508
>
> Committed: https://crrev.com/e0729c56d35acfaf9738fdb32c6508cd78eaf089
> Cr-Commit-Position: refs/heads/master@{#14595}
TBR=stefan@webrtc.org,minyue@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6508
Review-Url: https://codereview.webrtc.org/2415943002
Cr-Commit-Position: refs/heads/master@{#14631}
This CL is a pure refactoring which should not result in any functinal
changes. It moves ownership of the RtcEventLog from webrtc::Call to the
webrtc::PeerConnection object.
This is done so that we can add RtcEventLog support for ICE events -
which will require the TransportController to have a pointer to the
RtcEventLog. PeerConnection is the closest common owner of both Call and
TransportController (through WebRtcSession).
BUG=webrtc:6393
Review-Url: https://codereview.webrtc.org/2353033005
Cr-Commit-Position: refs/heads/master@{#14578}
The original CL (https://codereview.webrtc.org/2315633002) was
reverted since the fuzzer depended on gflags and files in the
resources folder; neither of this is allowed for a fuzzer test in
Chromium. This new version streamlines the dependencies, and changes
the test to generate a sinusoid input audio signal instead of reading
from a file.
Original commit message:
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.
BUG=webrtc:5447
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device
Review-Url: https://codereview.webrtc.org/2384423002
Cr-Commit-Position: refs/heads/master@{#14523}
This CL introduces changes that clearly demarcate
where we disable Unequal Protection in the FEC.
No functional changes are expected.
BUG=webrtc:5654
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2314743002 .
Cr-Commit-Position: refs/heads/master@{#14496}
- Change some member functions to be private. These were only
called by other private member functions.
- Replace DeleteMediaPackets() with direct calls to
media_packets_.clear()
- Rename GetFecPacketsAsRed to GetUlpfecPacketsAsRed.
No functional changes are intended by this CL.
BUG=webrtc:5654
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2305793003 .
Cr-Commit-Position: refs/heads/master@{#14491}
The RtcEventLog headers need to be accessible from any place which needs
logging, and the implementation needs access to data structures that are
logged.
After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future).
The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/).
This change allows using RtcEventLog in the p2p/ directory, so that we
can log STUN pings and ICE state transitions.
BUG=webrtc:6393
R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2380683005 .
Cr-Commit-Position: refs/heads/master@{#14485}
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 change reduces the number of places where we first fread a I420
frame into a uint8_t buffer, followed by a copy into a frame buffer
object.
BUG=None
Review-Url: https://codereview.webrtc.org/2362683002
Cr-Commit-Position: refs/heads/master@{#14456}
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}
I think this will make a rtc::Thread object exist for the lifetime of
the environment, which will remove some uninteresting crashes.
BUG=chrome:648075
Review-Url: https://codereview.webrtc.org/2365373002
Cr-Commit-Position: refs/heads/master@{#14438}
This is done to ensure GN targets are placed in the same directory as of the source files.
BUG=webrtc:6412
NOTRY=True
Review-Url: https://codereview.webrtc.org/2365383004
Cr-Commit-Position: refs/heads/master@{#14411}
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}
This is to fix an issue introduced with iOS 10 where all applications that access the microphone have to include a string in the Info.plist file explaining why they need it.
BUG=webrtc:6403
Review-Url: https://codereview.webrtc.org/2359863003
Cr-Commit-Position: refs/heads/master@{#14354}
- Rename GenerateFec -> EncodeFec in ForwardErrorCorrection. This naming
is more consistent with DecodeFec.
- Add appropriate using directives, to reduce clutter in tests.
- Move ConstructMediaPackets to fec_test_helper.{h,cc}. This will help
future tests of ULPFEC/FlexFEC header formatters.
- Generalize tests in rtp_fec_unittest.cc to typed tests. This will help
testing ForwardErrorCorrection with both ULPFEC and FlexFEC.
This CL should not impact functionality or performance.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2267393002
Cr-Commit-Position: refs/heads/master@{#14314}
Reason for revert:
Will fix android build failure.
Original issue's description:
> Revert of Update test code to use I420Buffer when writing pixel data. (patchset #2 id:140001 of https://codereview.webrtc.org/2342783003/ )
>
> Reason for revert:
> I was too impatient; this made android builds fail instead. See https://build.chromium.org/p/client.webrtc/builders/Linux32%20ARM/builds/585/steps/compile/logs/stdio
>
> Original issue's description:
> > Reland of Update test code to use I420Buffer when writing pixel data. (patchset #1 id:1 of https://codereview.webrtc.org/2342123003/ )
> >
> > Reason for revert:
> > Intending to fix problem and reland.
> >
> > Original issue's description:
> > > Revert of Update test code to use I420Buffer when writing pixel data. (patchset #5 id:80001 of https://codereview.webrtc.org/2333373007/ )
> > >
> > > Reason for revert:
> > > Fails 64-bit windows builds, it turns out I missed some of the needed int/size_t casts. Example https://build.chromium.org/p/client.webrtc/waterfall?builder=Win64%20Release
> > >
> > > Hope our windows try bots get back in working shape soon.
> > >
> > > Original issue's description:
> > > > Update test code to use I420Buffer when writing pixel data.
> > > >
> > > > VideoFrameBuffer and VideoFrame will become immutable.
> > > >
> > > > BUG=webrtc:5921
> > > > R=magjed@webrtc.org, phoglund@webrtc.org
> > > >
> > > > Committed: https://crrev.com/280ad1514e44bf6717e5871526dd4632f759eb3d
> > > > Cr-Commit-Position: refs/heads/master@{#14249}
> > >
> > > TBR=phoglund@webrtc.org,palmkvist@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:5921
> > >
> > > Committed: https://crrev.com/fbf14607267adf03d235273283ca452a1e564861
> > > Cr-Commit-Position: refs/heads/master@{#14251}
> >
> > TBR=phoglund@webrtc.org,palmkvist@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:5921
> >
> > Committed: https://crrev.com/d21534a8cfe636bbcf3d7bb151945590abc92b2a
> > Cr-Commit-Position: refs/heads/master@{#14258}
>
> TBR=phoglund@webrtc.org,palmkvist@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:5921
>
> Committed: https://crrev.com/3011627142bccdd73fce9fec854abb1f6b02b5c1
> Cr-Commit-Position: refs/heads/master@{#14259}
TBR=phoglund@webrtc.org,palmkvist@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
BUG=webrtc:5921
Review-Url: https://codereview.webrtc.org/2347863002
Cr-Commit-Position: refs/heads/master@{#14283}
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values
This cl
Revert "Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ )"
This reverts commit 9fdbda6aa3f66ea872344c22e79b23361047cbab.
and fix the problem in the original cl in video_quality_test.cc
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2348533002
Cr-Commit-Position: refs/heads/master@{#14265}
Reason for revert:
Intending to fix problem and reland.
Original issue's description:
> Revert of Update test code to use I420Buffer when writing pixel data. (patchset #5 id:80001 of https://codereview.webrtc.org/2333373007/ )
>
> Reason for revert:
> Fails 64-bit windows builds, it turns out I missed some of the needed int/size_t casts. Example https://build.chromium.org/p/client.webrtc/waterfall?builder=Win64%20Release
>
> Hope our windows try bots get back in working shape soon.
>
> Original issue's description:
> > Update test code to use I420Buffer when writing pixel data.
> >
> > VideoFrameBuffer and VideoFrame will become immutable.
> >
> > BUG=webrtc:5921
> > R=magjed@webrtc.org, phoglund@webrtc.org
> >
> > Committed: https://crrev.com/280ad1514e44bf6717e5871526dd4632f759eb3d
> > Cr-Commit-Position: refs/heads/master@{#14249}
>
> TBR=phoglund@webrtc.org,palmkvist@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:5921
>
> Committed: https://crrev.com/fbf14607267adf03d235273283ca452a1e564861
> Cr-Commit-Position: refs/heads/master@{#14251}
TBR=phoglund@webrtc.org,palmkvist@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:5921
Review-Url: https://codereview.webrtc.org/2342783003
Cr-Commit-Position: refs/heads/master@{#14258}
These were blocked on GN work; should be good to go
now. If they break the WebRTC FYI bots, there's more
work to be done. Verified they build locally at least.
Setting no-try because all bots went green except broken android bots.
BUG=6368
NOTRY=true
Review-Url: https://codereview.webrtc.org/2342843002
Cr-Commit-Position: refs/heads/master@{#14256}
Reason for revert:
Fails on Mac and Linux webrtc_perf_tests
Original issue's description:
> Replace VideoCapturerInput with VideoSinkInterface.
> Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
>
> This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
>
> BUG=webrtc:5687
> // Android CQ seems broken.
> NOTRY=true
>
> Committed: https://crrev.com/95a226f55ae7e32b83a6ba96232fb105a014dc6c
> Cr-Commit-Position: refs/heads/master@{#14238}
TBR=nisse@webrtc.org,sprang@webrtc.org,mflodman@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:5687
Review-Url: https://codereview.webrtc.org/2344923002
Cr-Commit-Position: refs/heads/master@{#14239}
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
BUG=webrtc:5687
// Android CQ seems broken.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2257413002
Cr-Commit-Position: refs/heads/master@{#14238}
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}
This way we don't have to rely on the existence of DEPS, and the tests
can be run in swarming bots (which don't have a checkout and therefore
don't have a DEPS file).
This seems to be where Chromium is assumming the project root path to
be.
NOTRY=True
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2340773002
Cr-Commit-Position: refs/heads/master@{#14230}
Declare resources for GN targets so that they can be isolated
NOTRY=True
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2340753002
Cr-Commit-Position: refs/heads/master@{#14210}
Only used in VoE tests so I'm moving it in under voice_engine/ until we've removed its usage and can deprecate.
Note: submitting this with PRESUBMIT=false because the files do not adhere to style guide conventions and I'd rather change that in a separate CL.
BUG=
NOPRESUBMIT=true
Committed: https://crrev.com/ade2a038a9290ee0c85d8c682eba5447aca943cd
Review-Url: https://codereview.webrtc.org/2319583005
Cr-Original-Commit-Position: refs/heads/master@{#14191}
Cr-Commit-Position: refs/heads/master@{#14198}
- Rename GetNumberOfFecPackets -> NumFecPackets and
PacketOverhead -> MaxPacketOverhead in ForwardErrorCorrection.
- Rename FECPacketOverhead -> FecPacketOverhead in ProducerFec.
- Move ownership of ForwardErrorCorrection from RTPSenderVideo
to ProducerFec.
- Make MaxPacketOverhead a member function of ForwardErrorCorrection.
This will allow for changing it, based on FEC header types, later on.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2275443002
Cr-Commit-Position: refs/heads/master@{#14194}
