Commit Graph

20061 Commits

Author SHA1 Message Date
e40468ba3d Move some numeric utility code from rtc_base/ to rtc_base/numerics/
Specifically, I'm moving

  safe_compare.h
  safe_conversions.h
  safe_minmax.h

They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.

BUG=webrtc:8445

Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
2017-11-22 11:21:47 +00:00
9deda4828d Move const variables initialization to constants.cc
Bug: None
Change-Id: I73f0d343e242e959879c5396852270c78c17fc87
Reviewed-on: https://webrtc-review.googlesource.com/22420
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20828}
2017-11-22 11:14:37 +00:00
9b28a03b95 Optional: Use nullopt and implicit construction in /pc/peerconnection.cc
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=hbos@webrtc.org

Bug: None
Change-Id: I57a1ac8c2a05de403ff87b319c7a50fad17c1c96
Reviewed-on: https://webrtc-review.googlesource.com/23571
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20827}
2017-11-22 11:05:27 +00:00
34afa0b1f7 Adding win_asan to CQ.
The Chromium Roll https://webrtc-review.googlesource.com/24760
introduced a breakage on win_asan. In order to re-enable the roll
we have to be protected against these kind of failures.

Bug: webrtc:8562
Change-Id: I8de6268eb1d3f87cba53889f1ae2397b3dbf491e
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/25080
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20826}
2017-11-22 09:12:18 +00:00
9e6fd2bd47 Add streams() to RtpReceiverInterface and implementations.
This makes the receiver know about its associated set of streams, the
equivalent of the [[AssociatedRemoteMediaStreams]] slot in the spec,
https://w3c.github.io/webrtc-pc/#dfn-x%5B%5Bassociatedremotemediastreams%5D%5D

This does not change layers below peerconnection.cc. The streams are set
upon the receiver's construction and is not modified for the duration of
its lifetime.

When we support modifying the associated set of streams of a receiver
the receiver needs to know about it. The receiver's streams() should be
used in all places where a receiver's streams need to be known.

Bug: webrtc:8473
Change-Id: I31202973aed98e61fa9b6a78b52e815227b6c17d
Reviewed-on: https://webrtc-review.googlesource.com/22922
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20825}
2017-11-22 09:04:17 +00:00
dd41194ca4 Removing assert in rtc_test template.
The assert causes a error when package default visibility is used.

Bug: None
Change-Id: I0d9006da8d838bd7113856bbf33fa2060c118f67
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/24900
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20824}
2017-11-22 08:26:37 +00:00
d319534143 Move ADM initialization into WebRtcVoiceEngine
Bug: webrtc:4690
Change-Id: I3b8950fdb13835964c5bf41162731eff5048bf1a
Reviewed-on: https://webrtc-review.googlesource.com/23820
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20823}
2017-11-21 20:48:07 +00:00
37e489c985 Add network_type to local RTCIceCandidateStats
Details:
  * Add RTCNetworkType enum
  * Add network_type to local ice candidate stats

Bug: webrtc:8435
Change-Id: Idb872849c09ad49c8f759d02afdc825e397afa07
Reviewed-on: https://webrtc-review.googlesource.com/14680
Commit-Queue: Gary Liu <qinghualiu@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20822}
2017-11-21 19:58:37 +00:00
09d584457a Fix some uninitialized variables in convert_legacy_video_factory
This is causing compilation failure on Chrome on Windows using MSVC
64bits.

TBR=andersc@webrtc.org

Bug: chromium:787192
Change-Id: If82290a9d7373385de4820b34f2c7dc9225c3d95
Reviewed-on: https://webrtc-review.googlesource.com/24981
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20821}
2017-11-21 19:09:58 +00:00
36f8f3eaab Optional: Use nullopt and implicit construction in /pc
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

TBR=pthatcher@webrtc.org

Bug: None
Change-Id: If41c462dc3ddff664d0b70d249d760e2ca4c8ab3
Reviewed-on: https://webrtc-review.googlesource.com/23576
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20820}
2017-11-21 17:53:37 +00:00
10dd7ed81a Support for external VAD program in APM-QA
There is now an 'ExternalVad' class in the AnnotationsExtractor. The
Extractor takes an extra list of these in addition to the other
VADs. The external VAD runs an external program to generate the
annotations. Annotations are loaded and saved to a compressed Numpy format.

Also made a small fix to name a mixed file in a way so that files will not
be overwritten.

Also did some minor changes to the unittests.
TBR=alessiob@webrtc.org

Bug: webrtc:7494
Change-Id: I7816b04466be16cd635ac6ceab18cd7aad5325a4
Reviewed-on: https://webrtc-review.googlesource.com/23623
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20819}
2017-11-21 16:44:19 +00:00
c61ce0d0cd Fixing some clang-tidy findings.
Bug: None
Change-Id: I949c1ff35284ce79c99e8f76148f63b8bba965a9
Reviewed-on: https://webrtc-review.googlesource.com/24041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20818}
2017-11-21 16:43:07 +00:00
89e712649d Optional: Use nullopt and implicit construction in /pc/rtcstatscollector.cc
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=hbos@webrtc.org

Bug: None
Change-Id: I435258d0b16660bcc58c236158c19507ef60f57b
Reviewed-on: https://webrtc-review.googlesource.com/23609
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20817}
2017-11-21 15:04:17 +00:00
7880758b90 Optional: Use nullopt and implicit construction in /media
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

TBR=pthatcher@webrtc.org

Bug: None
Change-Id: I6dd8677a65f897877fc848aefa7ab37d844e70ed
Reviewed-on: https://webrtc-review.googlesource.com/23573
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20816}
2017-11-21 14:46:37 +00:00
872cf38c4d Fix some dependencies for peerconnection_and_implicit_call_api.
Some targets depend on "api/peerconnectioninterface.h" which is part of
//api:peerconnection_and_implicit_call_api.
Furthermore, peerconnection_and_implicit_call_api depends among other
things of headers in //media:rtc_media_base, so we should add it as a
dependency as well.

Bug: webrtc:7504
Change-Id: Ifab69253d52532876509b3507917b1c93d99a2ac
Reviewed-on: https://webrtc-review.googlesource.com/24660
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20815}
2017-11-21 13:59:27 +00:00
ed1023db0e Android: Disable internal tracer by default.
This is a debug feature and should be disabled by default. Any client
that needs this functionality should call setEnableInternalTracer.

Bug: webrtc:8553
Change-Id: I78d718ebb95fc5cb8c464327b5b36e385ccfa9c0
Reviewed-on: https://webrtc-review.googlesource.com/24540
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20814}
2017-11-21 11:53:57 +00:00
215588132c rtc_test visibility must only be public.
"rtc_test" expands to "test" which is another template. The "test"
template overrides the visibility to public when targeting Android.

WebRTC wants to avoid this kind of situations and since a test should
be considered like an executable it is ok to force to public.

Bug: None
Change-Id: Iaf382721c60bb7345d4599a0279f94a4653247ec
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/24560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20813}
2017-11-21 11:50:57 +00:00
a7e418cd5f Add RequestKeyFrame with Pli to RtcpTransceiver
Add support for reduced size mode.

Bug: webrtc:8239
Change-Id: I1d646f0d7848af6632c9204ce5b96ae24cfc0ad3
Reviewed-on: https://webrtc-review.googlesource.com/23681
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20812}
2017-11-21 11:19:07 +00:00
de939432dc Revert "Revert "Encode log events periodically instead of for every event.""
This reverts commit 33c5c7f5e4f018a5103770021328fc530d451d75.

Reason for revert: Fix broken API change.

TBR=sprang@webrtc.org,solenberg@webrtc.org
TBRing because only pc/ and api/ have changed since last LGTMed version.

Original change's description:
> Revert "Encode log events periodically instead of for every event."
>
> This reverts commit b154c27e72fddb6c0d7cac69a9c68fff22154519.
>
> Reason for revert: Broke the internal project.
>
> Original change's description:
> > Encode log events periodically instead of for every event.
> >
> > Updated unit test to take output_period and random seed as parameters.
> > Updated the peerconnection interface to allow passing in an output_period.
> >
> > This is in preparation of some upcoming CLs that will change the format
> > to store batches of delta-encoded values.
> >
> >
> > Bug: webrtc:8111
> > Change-Id: Id5d9844dfad8d8edad346cd7cbebc7eadaaa5416
> > Reviewed-on: https://webrtc-review.googlesource.com/22600
> > Commit-Queue: Björn Terelius <terelius@webrtc.org>
> > Reviewed-by: Elad Alon <eladalon@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20736}
>
> Change-Id: I2257c46c014adb8c7c4fb28538635cabed1f2229
> Bug: webrtc:8111
> Reviewed-on: https://webrtc-review.googlesource.com/24160
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20738}

Bug: webrtc:8111
Change-Id: Ie69862cd52d11c1e15adeb6e2caacafe16863c80
Reviewed-on: https://webrtc-review.googlesource.com/24620
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20811}
2017-11-21 10:58:57 +00:00
63e6072a43 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.
(See: https://webrtc-review.googlesource.com/c/src/+/23820)

Bug: webrtc:4690
Change-Id: I474a327303aa0c9b5b34c2055ae3a35e466a7d9f
Reviewed-on: https://webrtc-review.googlesource.com/24720
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20810}
2017-11-21 10:51:02 +00:00
d6c98c020a Add check to see if output device has HogMode property (Mac OS X)
Bug: webrtc:8549
Change-Id: I952db26de02ccce8155762531cbae9411abafb28
Reviewed-on: https://webrtc-review.googlesource.com/24125
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20809}
2017-11-21 10:49:57 +00:00
212ad82494 Revert "Roll chromium_revision 3d1e7a12b7..8b6898fe4e (517813:518117)"
This reverts commit 12325c5b83d46e5114ce27c7db97ab5e8ab08ff9.

Reason for revert: It breaks win32_asan.

Original change's description:
> Roll chromium_revision 3d1e7a12b7..8b6898fe4e (517813:518117)
> 
> Change log: 3d1e7a12b7..8b6898fe4e
> Full diff: 3d1e7a12b7..8b6898fe4e
> 
> Changed dependencies:
> * src/base: 4c3d6272f8..b2b46604dd
> * src/build: 571871636f..6167734e03
> * src/ios: 865de15eaf..d6ba2ebe6e
> * src/testing: 14acaa118c..c5ed5cc935
> * src/third_party: d43ee09615..de6ce59aae
> * src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6cc352e216..8793942c5c
> * src/third_party/depot_tools: ddfead3483..7227d2175a
> * src/third_party/ffmpeg: 252244150a..168dfaa19b
> * src/tools: bce4039f1c..2c0f60c1ad
> DEPS diff: 3d1e7a12b7..8b6898fe4e/DEPS
> 
> No update to Clang.
> 
> TBR=buildbot@webrtc.org,
> BUG=None
> CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
> 
> Change-Id: I7115e7c9e4bc14b1fbe46554413d9f45954a590c
> Reviewed-on: https://webrtc-review.googlesource.com/24760
> Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
> Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20807}

TBR=buildbot@webrtc.org

Change-Id: I143941a4977c177ad1819ed8d2c12d05fe3c37d5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Cq-Include-Trybots: master.internal.tryserver.corp.webrtc:linux_internal
Reviewed-on: https://webrtc-review.googlesource.com/24840
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20808}
2017-11-21 10:40:48 +00:00
12325c5b83 Roll chromium_revision 3d1e7a12b7..8b6898fe4e (517813:518117)
Change log: 3d1e7a12b7..8b6898fe4e
Full diff: 3d1e7a12b7..8b6898fe4e

Changed dependencies:
* src/base: 4c3d6272f8..b2b46604dd
* src/build: 571871636f..6167734e03
* src/ios: 865de15eaf..d6ba2ebe6e
* src/testing: 14acaa118c..c5ed5cc935
* src/third_party: d43ee09615..de6ce59aae
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6cc352e216..8793942c5c
* src/third_party/depot_tools: ddfead3483..7227d2175a
* src/third_party/ffmpeg: 252244150a..168dfaa19b
* src/tools: bce4039f1c..2c0f60c1ad
DEPS diff: 3d1e7a12b7..8b6898fe4e/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I7115e7c9e4bc14b1fbe46554413d9f45954a590c
Reviewed-on: https://webrtc-review.googlesource.com/24760
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20807}
2017-11-21 06:13:34 +00:00
79e7960152 Add SDP semantics option to RTCConfiguration
This setting allows the user of PeerConnection to choose whether
to use Plan B (current) or Unified Plan (future) semantics.
Unified Plan semantics are not yet supported.

Bug: chromium:465349
Change-Id: I77a5c376c83f335f734488e11e619582a314bffe
Reviewed-on: https://webrtc-review.googlesource.com/22766
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20806}
2017-11-21 02:09:03 +00:00
03fdc1042f Remove an unused debug message.
TBR=zhihuang@webrtc.org

Bug: None
Change-Id: If4cbffbba1e25a839a89c2b5956138481cb00edd
Reviewed-on: https://webrtc-review.googlesource.com/24740
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20805}
2017-11-21 00:18:12 +00:00
f2d7beb1d4 Created the DtlsSrtpTransport.
The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel.
DtlsSrtpTransport is responsible for exporting keys from DtlsTransport
and setting up the wrapped SrtpTransport.

The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL.

Bug: webrtc:7013
Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713
Reviewed-on: https://webrtc-review.googlesource.com/6700
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20804}
2017-11-20 23:18:22 +00:00
4eb0188cb6 Android: Generate JNI code for VideoDecoder
Bug: webrtc:8278
Change-Id: I985fa63b0c5a9cdd0fb1817730646bcd4b30288a
Reviewed-on: https://webrtc-review.googlesource.com/24221
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20803}
2017-11-20 22:48:02 +00:00
4171afb186 Use RtpTransceivers in PeerConnection
Moves ownership of the RtpSenders/RtpReceivers/BaseChannels to
RtpTransceiver objects. For now, there can only be one
RtpTransceiver for audio and one for video. Future work to
implement Unified Plan will relax this restriction.

Bug: webrtc:7600
Change-Id: I9dfe324de61e2b363948858da72624396e27fc1a
Reviewed-on: https://webrtc-review.googlesource.com/21461
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20802}
2017-11-20 22:41:44 +00:00
cada60193d Reland "Android: Generate JNI code for androidnetworkmonitor_jni"
This reverts commit 522c1bc6bb945d66bf77f175da48e1644d74511e.

Reason for revert: Reland with a temporary fix.

Original change's description:
> Revert "Android: Generate JNI code for androidnetworkmonitor_jni"
> 
> This reverts commit 768e1c0ea1f2077675df31915132a0557aca400e.
> 
> Reason for revert: Breaks AppRTCMobile.
> 
> Original change's description:
> > Android: Generate JNI code for androidnetworkmonitor_jni
> > 
> > Bug: webrtc:8278
> > Change-Id: I8447b2de5ec2610760f7112b6f86e54d94325322
> > Reviewed-on: https://webrtc-review.googlesource.com/24520
> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20796}
> 
> TBR=magjed@webrtc.org,sakal@webrtc.org
> 
> Change-Id: I45f8d19abd81386872b7c095ac7eca21fa06077c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8278,webrtc:8556
> Reviewed-on: https://webrtc-review.googlesource.com/24622
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20800}

TBR=magjed@webrtc.org,sakal@webrtc.org

Change-Id: I7cc404993addb17f0397127a10aac67476ef6ff4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278, webrtc:8556
Reviewed-on: https://webrtc-review.googlesource.com/24623
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20801}
2017-11-20 20:56:00 +00:00
522c1bc6bb Revert "Android: Generate JNI code for androidnetworkmonitor_jni"
This reverts commit 768e1c0ea1f2077675df31915132a0557aca400e.

Reason for revert: Breaks AppRTCMobile.

Original change's description:
> Android: Generate JNI code for androidnetworkmonitor_jni
> 
> Bug: webrtc:8278
> Change-Id: I8447b2de5ec2610760f7112b6f86e54d94325322
> Reviewed-on: https://webrtc-review.googlesource.com/24520
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20796}

TBR=magjed@webrtc.org,sakal@webrtc.org

Change-Id: I45f8d19abd81386872b7c095ac7eca21fa06077c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278,webrtc:8556
Reviewed-on: https://webrtc-review.googlesource.com/24622
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20800}
2017-11-20 20:51:50 +00:00
eeb2765f6c Implement Opus bandwidth adjustment behind a FieldTrial
Bug: webrtc:8522
Change-Id: I3a32ebfecd27ff74b507c2cee9e16aab17153442
Reviewed-on: https://webrtc-review.googlesource.com/22210
Commit-Queue: Alejandro Luebs <aluebs@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20799}
2017-11-20 20:04:19 +00:00
64eaa99cfc On-fly calculation of quality metrics.
Calculation of quality metrics required writing of decoded video
to file. There were two drawbacks with that approach. First, frame
drops significantly affected metrics because comparison was done
against the last decoded frame. Second, simulcast/SVC required
writing of multiple files. This might be too much data to dump.

On-fly metrics calculation is done in frame decoded callback.
Calculation time is excluded from encoding/decoding time. If CPU
usage measurement is enabled metrics calculation is disabled since
it affects CPU usage. The results are reported in Stats::PrintSummary.

Bug: webrtc:8524
Change-Id: Id54fb21f2f95deeb93757afaf46bde7d7ae18dac
Reviewed-on: https://webrtc-review.googlesource.com/22560
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20798}
2017-11-20 16:13:59 +00:00
c5975bf25d Add performance tests for rtc::Event.
Bug: webrtc:8546
Change-Id: Ic069b213b5c0d83123a6506f10c597cf73396459
Reviewed-on: https://webrtc-review.googlesource.com/23960
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20797}
2017-11-20 16:09:59 +00:00
768e1c0ea1 Android: Generate JNI code for androidnetworkmonitor_jni
Bug: webrtc:8278
Change-Id: I8447b2de5ec2610760f7112b6f86e54d94325322
Reviewed-on: https://webrtc-review.googlesource.com/24520
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20796}
2017-11-20 16:00:39 +00:00
202be3957d Android: Generate JNI code for androidvideotracksource
Bug: webrtc:8278
Change-Id: I43b53c68ebaf2c3f9c27ea7ef510d7b016c1df93
Reviewed-on: https://webrtc-review.googlesource.com/23243
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20795}
2017-11-20 15:25:59 +00:00
36de62e830 Avoid flagging Opus DTX frames as speech.
Background: After 20 consecutive DTX frames, Opus encodes the background
noise in a normal frame and then goes back to outputting DTX frames.

Currently all Opus frames are flagged as containing speech.

This CL is has two effects on outgoing Opus packets:
1. DTX frames are flagged as non-speech.
2. A non-DTX frame that follows 20 consecutive DTX frames is flagged as
   non-speech.

Bug: webrtc:8088
Change-Id: Ic36cf8c9d0a34f55ed4e57858362ad91e3897dda
Reviewed-on: https://webrtc-review.googlesource.com/23760
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20794}
2017-11-20 14:53:40 +00:00
e5b5f4638d Roll chromium_revision 92f669180c..3d1e7a12b7 (517788:517813)
Change log: 92f669180c..3d1e7a12b7
Full diff: 92f669180c..3d1e7a12b7

Changed dependencies:
* src/base: 474649f29b..4c3d6272f8
* src/ios: 1e7c47ae2c..865de15eaf
* src/third_party: 82e8cb8ce5..d43ee09615
* src/tools: b0592cf79b..bce4039f1c
DEPS diff: 92f669180c..3d1e7a12b7/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I0b0b1ccd260bbb6a9625e5fa7156ad48803e7940
Reviewed-on: https://webrtc-review.googlesource.com/24481
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20793}
2017-11-20 14:39:00 +00:00
22d43f3352 iOS: Don’t change video rotation in FaceUp or FaceDown.
Previously, if using the device in landscape and then tilting the phone
into FaceUp orientation, the video rotation would reset to portrait.

Bug: webrtc:8492
Change-Id: I3e11e3adecabf99249ba3a8d5532291580a93f2e
Reviewed-on: https://webrtc-review.googlesource.com/24021
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20792}
2017-11-20 14:28:30 +00:00
82ed988a1c Allow multiple Init() calls on FakeAudioDevice
This is temporarily needed while landing https://webrtc-review.googlesource.com/c/src/+/23820 and updating clients.

Bug: webrtc:4690
Change-Id: Ib0bd6a6a063a8a54c80b73853b2c042dfb02c44a
Reviewed-on: https://webrtc-review.googlesource.com/24501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20791}
2017-11-20 13:37:00 +00:00
c6a003c192 Android: Simplify GetJavaEnumName
We can call name() on the base class java.lang.Enum instead of the
specific enum subclass.

Bug: webrtc:8278
Change-Id: I12f8354d99e48b4a5e3d3eaff74a85ac417df0b3
Reviewed-on: https://webrtc-review.googlesource.com/24440
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20790}
2017-11-20 13:20:30 +00:00
ae02609645 Add parallel stats interface with optional stats to APM.
This new parallel GetStatistics function uses Optionals to indicate if stats are valid or not, and no longer relies on default values. It also takes an argument to indicate if receive streams are present, and if not several stats will not be set.

Bug: b/67926135
Change-Id: I175de1c65c414bea6ec9ca8b0b16f07cb2308d9f
Reviewed-on: https://webrtc-review.googlesource.com/17942
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20789}
2017-11-20 13:13:20 +00:00
0ce0988503 Expose audio record source setting in WebRtcAudioRecord.
Landing https://webrtc-review.googlesource.com/c/src/+/23881 on behalf
of stevengatto@

TBR=glaznev

Bug: webrtc:8545
Change-Id: I4358b93d2f4d934c497c4d3ee7e86e1fbc7a5fae
Reviewed-on: https://webrtc-review.googlesource.com/24460
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20788}
2017-11-20 13:06:21 +00:00
e9d482bcfc Roll chromium_revision 27b69bb27b..92f669180c (517785:517788)
Change log: 27b69bb27b..92f669180c
Full diff: 27b69bb27b..92f669180c

Changed dependencies:
* src/third_party: f42f9f8f0b..82e8cb8ce5
DEPS diff: 27b69bb27b..92f669180c/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I8c793714986eb7eb152206944ec0147d46f19522
Reviewed-on: https://webrtc-review.googlesource.com/24267
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20787}
2017-11-20 10:58:30 +00:00
aa8b67da9d Optional: Use nullopt and implicit construction in /modules/audio_processing
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=henrik.lundin@webrtc.org

Bug: None
Change-Id: I733a83f702fe11884d229a1713cfac952727bde8
Reviewed-on: https://webrtc-review.googlesource.com/23601
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20786}
2017-11-20 10:19:30 +00:00
28dfeb7f24 Roll chromium_revision 38c60c130c..27b69bb27b (517782:517785)
Change log: 38c60c130c..27b69bb27b
Full diff: 38c60c130c..27b69bb27b

Changed dependencies:
* src/third_party: 21742359cb..f42f9f8f0b
DEPS diff: 38c60c130c..27b69bb27b/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I709ba2631109e9664a0af0a6c69de90a1297ea12
Reviewed-on: https://webrtc-review.googlesource.com/24436
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20785}
2017-11-20 10:03:20 +00:00
53893175cd Roll chromium_revision 3b4c8d5365..38c60c130c (517770:517782)
Change log: 3b4c8d5365..38c60c130c
Full diff: 3b4c8d5365..38c60c130c

Changed dependencies:
* src/base: b638059557..474649f29b
* src/third_party: 2ac1392726..21742359cb
DEPS diff: 3b4c8d5365..38c60c130c/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I62c14ac401a5461b84cf2a965e213eba963b21ed
Reviewed-on: https://webrtc-review.googlesource.com/24435
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20784}
2017-11-20 09:13:10 +00:00
156af4ae61 neteq_rtpplay: Add buffer size (target and current) to print-out
Bug: none
Change-Id: Id940471235e9f54e1e46569c74255759a891395d
Reviewed-on: https://webrtc-review.googlesource.com/24100
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20783}
2017-11-20 08:07:30 +00:00
702d5d390f Roll chromium_revision ccd0bd543f..3b4c8d5365 (517759:517770)
Change log: ccd0bd543f..3b4c8d5365
Full diff: ccd0bd543f..3b4c8d5365

Changed dependencies:
* src/third_party: 7d6301c2e2..2ac1392726
* src/tools: 0262309153..b0592cf79b
DEPS diff: ccd0bd543f..3b4c8d5365/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I164800883f94b083643c2380fa532b26d17ae96f
Reviewed-on: https://webrtc-review.googlesource.com/24433
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20782}
2017-11-20 07:48:00 +00:00
dae2fb8b87 Roll chromium_revision fe4e8d4a86..ccd0bd543f (517754:517759)
Change log: fe4e8d4a86..ccd0bd543f
Full diff: fe4e8d4a86..ccd0bd543f

Changed dependencies:
* src/third_party: 212917edef..7d6301c2e2
DEPS diff: fe4e8d4a86..ccd0bd543f/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I0f41184a879e7a70692382d6caaeba3e556e77d5
Reviewed-on: https://webrtc-review.googlesource.com/24431
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20781}
2017-11-20 06:35:00 +00:00
e9cd220eb1 Roll chromium_revision 91eaa1ed39..fe4e8d4a86 (517752:517754)
Change log: 91eaa1ed39..fe4e8d4a86
Full diff: 91eaa1ed39..fe4e8d4a86

Changed dependencies:
* src/third_party: 9aed422455..212917edef
* src/tools: 269f9b1951..0262309153
DEPS diff: 91eaa1ed39..fe4e8d4a86/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I9e35232c5b1a6986178d689e1bf8cae80c5184e3
Reviewed-on: https://webrtc-review.googlesource.com/24429
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20780}
2017-11-20 05:27:30 +00:00