Seems like there are still issues when uploading.
TBR=phoglund@webrtc.org
No-Try: true
Bug: webrtc:7156
Change-Id: I4ed1149afa1dc4f38ad7d48926f5b624743d1caa
Reviewed-on: https://webrtc-review.googlesource.com/37960
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21520}
This will require a manual roll to downstream projects, since
the //test:perf_test target was introduced.
This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8
Original change's description:
> iOS: Save perf results under Documents/perf_result.json
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:7156
> Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5
> Reviewed-on: https://webrtc-review.googlesource.com/29202
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21244}
R=henrika@webrtc.org, phoglund@webrtc.org
Bug: webrtc:7156
Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424
Reviewed-on: https://webrtc-review.googlesource.com/37643
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21518}
This creates a new target for pure defines and interfaces. I think
that makes sense (though include/ makes it harder to see when .cc and
.h files should live together).
Bug: webrtc:7620
Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e
Reviewed-on: https://webrtc-review.googlesource.com/34657
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21516}
This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
Bug: webrtc:7925
Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454
Reviewed-on: https://webrtc-review.googlesource.com/37000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21513}
The removal of these headers has been announced in November with
https://groups.google.com/forum/#!topic/discuss-webrtc/0vWBzJs0yDU.
Bug: webrtc:5883
Change-Id: I6ead2e3bd619472db1a78c0ded5dc57bdb66b76c
Reviewed-on: https://webrtc-review.googlesource.com/34648
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21512}
Starting from Chromium Roll [1], WebRTC should start to use NDK r16
for Android builds. The roll cannot be completed because of three
compilation errors:
../../sdk/android/src/jni/pc/androidnetworkmonitor.cc:15:9: error: 'RTLD_NOLOAD' macro redefined [-Werror,-Wmacro-redefined]
^
../../third_party/android_tools/ndk/sysroot/usr/include/dlfcn.h:62:9: note: previous definition is here
../../modules/audio_device/android/audio_record_jni.cc:251:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
../../modules/audio_device/android/audio_track_jni.cc:229:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
This CL forward fixes these errors in order to fix the Chromium Roll
into WebRTC.
[1] - https://webrtc-review.googlesource.com/c/src/+/37540
Bug: webrtc:8710
Change-Id: I5bc64e73919eee7c9e965a442a386b5e1897b56a
Reviewed-on: https://webrtc-review.googlesource.com/37640
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21510}
AddTrack is just a legacy wrapper for the new AddTrack method, so
calling the new AddTrack method should do everything that the old one
does.
Bug: None
Change-Id: I272a9e9584c470d54243377c1307b786f41c660d
Reviewed-on: https://webrtc-review.googlesource.com/37546
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21508}
Bug: webrtc:8714
Change-Id: I107b7d5908cf6efe7f114522959da4503450d7fc
Reviewed-on: https://webrtc-review.googlesource.com/37724
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21506}
Expose RTCDtmfSender API for ObcC SDK via exising RTCRtpSender
to provide ability to use DTMF tones in ObjC apps which uses WebRTC.
Android SDK has already exposed DTMF API via Java's DtmfSender
object, there changes provide similar functionaly to ObjC SDK.
Bug: webrtc:8713
Change-Id: Id68fddbbc362211dc8032fa31b38812d1cff8ed9
Reviewed-on: https://webrtc-review.googlesource.com/35800
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21505}
Exposed setOptions API for iOS SDK via RTCPeerConnectionFactory method
to provide ability to disable encryption and control which network
adapters are ignored.
Only subset of webrtc::PeerConnectionFactoryInterface::Options options
are exposed via iOS SDK, additional options can be exposed as requested.
Android SDK has already exposed setOption API via Java's PeerConnection
constructor, there changes provide similar functionaly to iOS SDK.
Bug: webrtc:8712
Change-Id: Ia2de38cf382afc1bad9bbec6c6eac21ad29aee89
Reviewed-on: https://webrtc-review.googlesource.com/34900
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21504}
The macros confuses automatic tooling, Qt Creator fails to identify the
tests defined with the special macros used before.
The value for readers of defining the macros is not obvious either.
Macros can sometime make code more compact and therefore quicker to
overview. However they also increases ambiguity of the code and the
reader will have to look up their definition to know what they do.
In this case I argue that the slight decrease in code size does not
outweigh the cost of lost tooling support.
Bug: None
Change-Id: Ic496fbe1fefdc5acd3f50ec99e2c804bb6065c3d
Reviewed-on: https://webrtc-review.googlesource.com/33540
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21503}
All of the PlaysOutAudioAndVideoInSync* tests were reporting metrics under
the same name ("sync_convergence_time/synchronization") so that only one of
the tests (whichever ran last) had its metrics reported to the dashboard,
while the others were silently ignored.
I added a suffix to differentiate between them.
Bug: webrtc:8566
Change-Id: Ia51f0441d28b202581c5b22ef5ea683091557ab8
Reviewed-on: https://webrtc-review.googlesource.com/36541
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21501}
For the RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx and
RampUpTest.UpDownUpTransportSequenceNumberRtx [1,2], the generated metric names
are the same:
- ramp_up_down_up_3streams_rtx.first_rampup
- ramp_up_down_up_3streams_rtx.second_rampup
- ramp_up_down_up_3streams_rtx.rampdown
So only one of the two tests (whichever ran last) has its metrics reported to
the perf dashboard, while the others has its metrics ignored.
[1] https://webrtc.googlesource.com/src/+/master/call/rampup_tests.cc#571
[2] https://webrtc.googlesource.com/src/+/master/call/rampup_tests.cc#579
Bug: webrtc:8691
Change-Id: I632dfe32d3b4729f1b0233c44d03c2894ee8c027
Reviewed-on: https://webrtc-review.googlesource.com/36941
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21499}
VideoSourceInterface is clearly an integral part of
mediastreaminterface.h already, so moving that interface to api makes
sense. This also resolves a circular dependency in call/.
Bug: webrc:6828
Change-Id: Ic1862f118363b0b55a235a9c0c35d9adc647184c
Reviewed-on: https://webrtc-review.googlesource.com/37500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21498}
This is the old-style-stats equivalent of CL 34360.
Bug: webrtc:8616
Change-Id: I12573eb305a8f1ecf8134b87ab14e33eaec5ba22
Reviewed-on: https://webrtc-review.googlesource.com/37080
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21497}
I updated some dependency enforcement rules to allow examples and pc
to depend on common_video. I reckoned depending on common_video is
not controversial when they already dependend on media/base, which
is a lower-level abstraction.
Bug: webrtc:6828
Change-Id: I77dbeb10187b4e70dda1d873a29994fa76070758
Reviewed-on: https://webrtc-review.googlesource.com/34187
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21495}
WebRTC.Audio.InitRecordingDurationMs and
WebRTC.Audio.StartRecordingDurationMs UMA stats are added on Android
to measure the time consumed on these two methods where the main part
of the work is done in Java.
Bug: b/67854242
Change-Id: I2d5487511402db18009d66a39c66d3f10d98cdd6
Reviewed-on: https://webrtc-review.googlesource.com/37420
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21494}
The attribute android_manifest for android_library targets has been
removed in [1]. This CL renames it to android_manifest_for_lint (to
avoid lint errors) in all the rtc_android_library targets.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/848079
Bug: webrtc:8707
Change-Id: Ifa127790937fa49ed52d6aab0c7ce5ab03e1177b
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/37440
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21493}
Diff patch set 1 and 2 to see actual differences to the last
patch.
Bug: webrtc:6828
Change-Id: Ie0c85d41df47c2a2505bc71b20fdb3834bdeaf12
Reviewed-on: https://webrtc-review.googlesource.com/36920
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21492}
Setting ffmpeg_branding = "Chrome" is what causes a compilation error
(see [1], where h264_cabac.c is included):
../../third_party/ffmpeg/libavcodec/x86/cabac.h:193:9: error: inline assembly requires more registers than available
BRANCHLESS_GET_CABAC("%0", "%q0", "(%4)", "%1", "%w1",
[...]
See: https://build.chromium.org/p/tryserver.webrtc/builders/linux32_dbg/builds/50.
Also, from the compililation error, this might be a bug on clang?
...
clang: note: diagnostic msg: PLEASE submit a bug report to http://llvm.org/bugs/ and include the crash backtrace, preprocessed source, and associated run script.
...
[1] https://cs.chromium.org/chromium/src/third_party/ffmpeg/ffmpeg_generated.gni?l=222
Bug: webrtc:7413
No-Try: true
Change-Id: I5c785d8f6f61c72a5e7665367023fec017b18d3e
Reviewed-on: https://webrtc-review.googlesource.com/37360
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21489}
This was causing ICE pings to continue going out on PeerConnections
that use DataChannels, even after closing the PeerConnection.
This CL adds a two-line fix, and an integration test that will catch
this and similar issues.
Bug: webrtc:7655
Change-Id: I589a2a1aaf6433c1d65be69a1267e1b52a33534b
Reviewed-on: https://webrtc-review.googlesource.com/37145
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21488}
This prevents the programmer from accidentally adding LOG_START and LOG_END events to the log without actually starting the log. This also makes it easier to ensure that the LOG_START event always ends up first and the LOG_END event always ends up last in the log file.
Bug: webrtc:8111
Change-Id: I4e6c9306f8559ff184b5185f8728409f8dcebfa0
Reviewed-on: https://webrtc-review.googlesource.com/34400
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21486}
Use the PrintResult* functions from test/testsupport/perf_test.h
instead of using printf directly.
Bug: webrtc:8566
Change-Id: Icc3418402e5fbe4e695a64d0523e1f64aa27edf8
Reviewed-on: https://webrtc-review.googlesource.com/36420
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21483}
For DualStreamsTest, the name of the metric reported
("dualstreams_moderately_restricted_screenshare") was repeated 4 times:
- Conference_Restricted/0
- Conference_Restricted/1
- ModeratelyRestricted_SlidesVp8_3TL_Simulcast_Video_Simulcast_High/0
- ModeratelyRestricted_SlidesVp8_3TL_Simulcast_Video_Simulcast_High/1
So only one of those tests (whichever ran last) has its metrics reported
to the perf dashboard, while the others have their metrics ignored.
I added the "/0" or "/1" as part of the metric name, to differentiate
between them.
Bug: webrtc:8566
Change-Id: I088807b66f9b7957571dccdb8fe3df0d87486bb0
Reviewed-on: https://webrtc-review.googlesource.com/36400
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21481}
It will be fully removed soon because it duplicates the actions of win_rel
No-Try: True
Bug: webrtc:8664
Change-Id: I949d608b8de1f29b850fcaf036ffb7a0ef2bf28f
Reviewed-on: https://webrtc-review.googlesource.com/36501
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21471}
