Commit Graph

23581 Commits

Author SHA1 Message Date
7687ad58b2 Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"
This is a reland of 80c4cca4915dbc6094a5bfae749f85f7371eadd1

Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
> 
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
> 
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
> 
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
>   no longer be reached.
> 
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}

Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
2018-07-02 10:20:33 +00:00
04b18cb365 Removes redundant delay based bwe.
This removes the legacy DelayBasedBwe to reduce code redundancy and
avoid the risk of applying changes on only one version.

Bug: webrtc:8415
Change-Id: I88aba03adbb77ee0ff0a97a8b3be6ddf028af48a
Reviewed-on: https://webrtc-review.googlesource.com/85364
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23798}
2018-07-02 09:11:33 +00:00
e0eda662ef Adding alessiob@ and minyue@ as owners of APM.
NOTRY=True

Bug: None
Change-Id: I690140661cf09e505a4e9e874912f05d83f14dcd
Reviewed-on: https://webrtc-review.googlesource.com/85284
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23797}
2018-07-02 07:45:31 +00:00
fceaca3233 Roll chromium_revision f06b8215fe..c20726850b (571725:571826)
Change log: f06b8215fe..c20726850b
Full diff: f06b8215fe..c20726850b

Changed dependencies:
* src/base: 05a0132eb3..0d31d15d4c
* src/build: b79f5b50ec..213a0e3999
* src/testing: f7b8fb322b..20f36f2392
* src/third_party: c59f378278..fc2824c48d
* src/third_party/depot_tools: d4c2a87998..024a331759
* src/tools: c468102a6f..0935834c72
DEPS diff: f06b8215fe..c20726850b/DEPS

No update to Clang.

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

Change-Id: Ibd9198711785229b820c15d2c9f1914f7079e74c
Reviewed-on: https://webrtc-review.googlesource.com/86575
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23796}
2018-07-02 07:07:00 +00:00
dc99e244ca Removing deadbeef@ from OWNERS files.
Since I'm leaving Google.

Bug: None
Notry: True
Change-Id: Ibb5c3e09fce007d149200dcb6cac74be53084764
Reviewed-on: https://webrtc-review.googlesource.com/86461
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23795}
2018-07-02 00:40:38 +00:00
5a482847e9 Roll chromium_revision 810d8218ca..f06b8215fe (571617:571725)
Change log: 810d8218ca..f06b8215fe
Full diff: 810d8218ca..f06b8215fe

Changed dependencies:
* src/base: fe70ab13e4..05a0132eb3
* src/buildtools: aec56e2607..0dd5c6f980
* src/ios: df92b7461a..e7915b2a8e
* src/testing: a70a01c4a1..f7b8fb322b
* src/third_party: 500196de14..c59f378278
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f76f0b4406..34f0d7e2e4
* src/third_party/depot_tools: a4dec94a1a..d4c2a87998
* src/tools: e4cbb07d3c..c468102a6f
DEPS diff: 810d8218ca..f06b8215fe/DEPS

No update to Clang.

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

Change-Id: I074d799aac0cf23fa8ff22423c375a62d7dd406a
Reviewed-on: https://webrtc-review.googlesource.com/86500
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23794}
2018-06-30 01:08:07 +00:00
20b4b0dcf0 Roll chromium_revision a714568fbe..810d8218ca (571512:571617)
Change log: a714568fbe..810d8218ca
Full diff: a714568fbe..810d8218ca

Changed dependencies:
* src/base: 8c01d4ef25..fe70ab13e4
* src/build: 06960ce32c..b79f5b50ec
* src/ios: dc2780aff3..df92b7461a
* src/testing: 5e63b2909b..a70a01c4a1
* src/third_party: 42df0ae52f..500196de14
* src/third_party/depot_tools: 406de133ef..a4dec94a1a
* src/tools: 0597286f57..e4cbb07d3c
DEPS diff: a714568fbe..810d8218ca/DEPS

No update to Clang.

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

Change-Id: I2506d375debf94de8039352eb83749d986119564
Reviewed-on: https://webrtc-review.googlesource.com/86420
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23793}
2018-06-29 21:14:05 +00:00
cf5de1da07 Roll chromium_revision a88423acf9..a714568fbe (571410:571512)
Change log: a88423acf9..a714568fbe
Full diff: a88423acf9..a714568fbe

Changed dependencies:
* src/base: 311c937b26..8c01d4ef25
* src/build: c9333f9faf..06960ce32c
* src/buildtools: 9c9fd97928..aec56e2607
* src/ios: 34302909a8..dc2780aff3
* src/testing: b47e929d27..5e63b2909b
* src/third_party: b77d94a9b3..42df0ae52f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e2d6bbca62..f76f0b4406
* src/third_party/depot_tools: ae1f03388f..406de133ef
* src/tools: 6ff0d88db8..0597286f57
DEPS diff: a88423acf9..a714568fbe/DEPS

Clang version changed 335608:335864
Details: a88423acf9..a714568fbe/tools/clang/scripts/update.py

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

Change-Id: I85ce46b41d6cd2302fef17a86025e4b5eb913a8b
Reviewed-on: https://webrtc-review.googlesource.com/86380
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23792}
2018-06-29 17:20:14 +00:00
f9f49a323c Removes redundant AlrDetector.
This replaces the old AlrDetector used by the pacer with the one in
GoogCC. This reduces the risk of accidentally changing only one version.

Note that the pacer instance will be removed when moving over to the
task queue based send side congestion controller.

Bug: webrtc:8415
Change-Id: Id4b2000ee5a04b94565092c29a84572a7750d2f5
Reviewed-on: https://webrtc-review.googlesource.com/85363
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23791}
2018-06-29 16:28:04 +00:00
f222d2823d Adds srte@webrtc.org as modules/pacing/ OWNER.
Bug: webrtc:8415
Change-Id: I5ef199825dbb061ae91baa7f8781238433d72d67
Reviewed-on: https://webrtc-review.googlesource.com/86129
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23790}
2018-06-29 15:25:24 +00:00
2e79d2b398 AEC3: Misadjustment estimator of the linear filter.
In this work the performance of the linear filter is
estimated. The estimation aims at capture situations when the linear
filter is largely over-estimating the echo. In those circumstances,
the linear filter is scaled with the purpose of accelerating its
convergence.

Change-Id: I05ea3739d82838a6f08673432da92125c47943e0
Bug: webrtc:9466,chromium:857426
Reviewed-on: https://webrtc-review.googlesource.com/86133
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23789}
2018-06-29 15:05:14 +00:00
916ec7dadf Add Generic frame descritpor header extension
to list of extensions supported by RtpPacket.

Bug: webrtc:9361
Change-Id: Iabee824381be3776e17e95f32507058607fc0bc8
Reviewed-on: https://webrtc-review.googlesource.com/85346
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23788}
2018-06-29 15:02:44 +00:00
deee55b3d5 Calculate all audio samples in AudioMixerCalculateEnergy.
Bug: None
Change-Id: I1478bc6348f11d81a896a48007bc08228f4a5586
Reviewed-on: https://webrtc-review.googlesource.com/82880
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23787}
2018-06-29 14:47:13 +00:00
4c77dcd0cb Turn rtc::{Make,Wrap}Unique into aliases for their Abseil counterparts
We don't want to maintain our own versions. This CL is step one in
getting rid of them.

Bug: webrtc:9473
Change-Id: Ib8a54288509f4768b482367b738224869a5af559
Reviewed-on: https://webrtc-review.googlesource.com/86282
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23786}
2018-06-29 13:48:33 +00:00
425193b4a9 Revert "Unit test for case where the number of active and configured spatial"
This reverts commit 5eb6045ce5754ce815929c54dd27ab0bf3ae62ba.

Reason for revert: Test breaks downstream.

Original change's description:
> Unit test for case where the number of active and configured spatial
> layers differ.
> 
> Bug: webrtc:9472
> Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
> Reviewed-on: https://webrtc-review.googlesource.com/85644
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23782}

TBR=brandtr@webrtc.org,ssilkin@webrtc.org,mhoro@webrtc.org

Change-Id: Ib97cdb127e79ee969f7cb3f931cb7bd533f13af0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9472
Reviewed-on: https://webrtc-review.googlesource.com/86320
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23785}
2018-06-29 12:01:38 +00:00
7a29426142 Detach audio devices from thread on terminate.
To allow the AudioDeviceModule to be reinitialized on a different thread
after termination, detach AudioDeviceModule and the input/output devices
when Terminate is called. Also destroy the AudioDeviceBuffer.

Bug: webrtc:7452
Change-Id: I50ef77c531f33d4efa0567d0475dd8280337bed9
Reviewed-on: https://webrtc-review.googlesource.com/86127
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23784}
2018-06-29 12:00:17 +00:00
43d0b98fe5 Clean up RateControlInput struct, used by bandwidth estimation.
Remove unused member noise_var from RateControlInput struct.

Rename incoming_bitrate to estimated_throughput_bps to reflect
that the AimdRateControl may be running on either the send side
or the receive side.

Bug: webrtc:9411
Change-Id: Ie1ae0c29dc9559ef389993144e69fcd41684f1a4
Reviewed-on: https://webrtc-review.googlesource.com/83728
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Anastasia Koloskova <koloskova@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23783}
2018-06-29 10:47:37 +00:00
5eb6045ce5 Unit test for case where the number of active and configured spatial
layers differ.

Bug: webrtc:9472
Change-Id: I5cf292a12d73777ca0fd5771eb1a4756626f640c
Reviewed-on: https://webrtc-review.googlesource.com/85644
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23782}
2018-06-29 10:38:57 +00:00
4236991952 Set gtest_enable_absl_printers to true.
Starting from [1], gtest can pretty print absl types. In order to
enable the feature WebRTC has to set gtest_enable_absl_printers to true
in the .gn file.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1027711

Bug: None
Change-Id: I74eb9a48c361f1523dd8d45510297e101a4d14cd
Reviewed-on: https://webrtc-review.googlesource.com/85345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23781}
2018-06-29 09:36:17 +00:00
a91decab4f Implement PacketDuration() for FakeDecoderFromFile.
Bug: None
Change-Id: Ie4ab1ce737608706f12f298f793f76571805ca91
Reviewed-on: https://webrtc-review.googlesource.com/86160
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23780}
2018-06-29 08:32:36 +00:00
c54f706993 Roll chromium_revision ecf8a6133e..a88423acf9 (569618:571410)
Change log: ecf8a6133e..a88423acf9
Full diff: ecf8a6133e..a88423acf9

Changed dependencies:
* src/base: f7595e419a..311c937b26
* src/build: 69593eb8fa..c9333f9faf
* src/buildtools: 5941c1b3df..9c9fd97928
* src/ios: 181b18c878..34302909a8
* src/testing: 8354b28f74..b47e929d27
* src/third_party: 46683344d7..b77d94a9b3
* src/third_party/android_tools: https://chromium.googlesource.com/android_tools.git/+log/3545ab5b98..130499e252
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/6ff2ba80b7..fec83fc78d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/87eefd4f11..e2d6bbca62
* src/third_party/depot_tools: c5a26a769e..ae1f03388f
* src/third_party/freetype/src: 7915fd51f1..a632fb547e
* src/third_party/libvpx/source/libvpx: 8648a64c83..583859d739
* src/third_party/libyuv: bc383e76d6..4d67b3e851
* src/third_party/r8: 1.0.30..1.2.28-cr0
* src/third_party/usrsctp/usrsctplib: 159d060dce..7a8bc9a90c
* src/tools: 592ddd1d14..6ff0d88db8
DEPS diff: ecf8a6133e..a88423acf9/DEPS

Clang version changed 334100:335608
Details: ecf8a6133e..a88423acf9/tools/clang/scripts/update.py

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

Change-Id: If3229d875265bca1bffffd01a793098ad2106f9f
Reviewed-on: https://webrtc-review.googlesource.com/86240
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23779}
2018-06-29 07:33:16 +00:00
e19a4e115f Revert "Pull GN via CIPD package."
This reverts commit 77cc8182aef6ec97ecd4c115fae5de4f511efa57.

Reason for revert: Breaks DEPS Auto-roller.

Original change's description:
> Pull GN via CIPD package.
> 
> The gn binary will be downloaded into third_party/gn.
> 
> The part about gn_win will be true only after the buildtools_revision
> will be updated by the Chromium roll.
> 
> This CL has been copied from https://chromium-review.googlesource.com/c/chromium/src/+/1117264/9/DEPS.
> 
> Bug: None
> Change-Id: I3fee1d9f6c39e508871798eeeb60d74ab7bc41d1
> Reviewed-on: https://webrtc-review.googlesource.com/86123
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23765}

TBR=mbonadei@webrtc.org,oprypin@webrtc.org

Change-Id: I660196e48a626e87ec5ed722b2a169620494d74c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/86220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23778}
2018-06-29 06:31:48 +00:00
776199a55a Enable PeerConnectionEndToEndTest.CallWithLegacySdp on ASan.
Bug: None
Change-Id: I9f695bd0a13b0130f4d773803e010b69020c2ac1
Reviewed-on: https://webrtc-review.googlesource.com/86131
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23777}
2018-06-29 05:00:46 +00:00
82d171c824 Skip PeerConnectionEndToEndTest.CallWithCustomCodec on Win ASan builds.
Bug: None
Change-Id: Iaee0bdee03e23aae916a641c6230e14ae229c6df
Reviewed-on: https://webrtc-review.googlesource.com/86130
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23776}
2018-06-29 04:57:36 +00:00
fc63c9e273 AEC3: Allow filter adaptation even though the estimated echo is saturated
This CL removes the constraint that freezes the filter adaptation
whenever the estimated echo or the prediction error is saturated. This
allows for much more rapid filter recovery in cases where the echo path
gain for some reason changes, such as when the analog AGC gain is
adjusted or the loudspeaker volume is changed.

TBR: devicentepena@webrtc.org
Bug: webrtc:9466,chromium:857426
Change-Id: Ic0b3b03f41f12e9a607aaadd2ee91cbaa16cac52
Reviewed-on: https://webrtc-review.googlesource.com/86124
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23775}
2018-06-28 22:45:18 +00:00
7750de906a Port RtcEventLog encoder unittests to the new parser API.
The Copy() function previously did not copy the logging timestamp.
To be able to use Copy() in this test, we add private copy
constructors for RtcEvents which the Copy() can use to copy
everything including the timestamp.

Also adds missing test for RtcEventAlrState,
RtcEventIceCandidatePairConfig and RtcEventIceCandidatePair.

Bug: webrtc:8111
Change-Id: I3901231735baa4e671173c921eada0a4be6de7c9
Reviewed-on: https://webrtc-review.googlesource.com/86042
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23774}
2018-06-28 15:31:23 +00:00
0601d68ac8 Adds field trial for disabling pacer queue draining.
This CL adds a field trial that disables the feature that the pacer will
ignore the pacing rate and send extra fast to drain the queues if the
pacer queue starts to fill up. BBR assumes that the pacing rate will be
respected and sending more increase the risk of overestimating the
bandwidth.

Bug: webrtc:8415
Change-Id: Ibba315360dafef1c317d14a83199172f9f8cc6aa
Reviewed-on: https://webrtc-review.googlesource.com/80964
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23773}
2018-06-28 13:46:22 +00:00
6c618c7002 AEC3: Avoid entering non-linear mode when the filter is slightly diverged
This CL changes the behavior when the main filter diverges.
Instead of entering non-linear mode, the AEC continues to operate in
linear mode but estimates the residual echo differently. R2 is S2
scaled by a factor of 10.

Bug: chromium:857018,webrtc:9462
Change-Id: I41212efe164ad319cf38a163cdf9d3ea151e0997
Reviewed-on: https://webrtc-review.googlesource.com/85981
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23772}
2018-06-28 13:35:18 +00:00
c75b35ab40 Fixed crash when PCF is destroyed before DataChannel in ObjC
Bug: webrtc:9231
Change-Id: Ifad698b366be61d33ffca81cf4f8ca8aba2988a2
Reviewed-on: https://webrtc-review.googlesource.com/86040
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23771}
2018-06-28 12:54:22 +00:00
33b61ee81e Delete unused file.
Bug: None
Change-Id: I9a29f6cb8bba4000a636e47e7381cebc255fe3d6
Reviewed-on: https://webrtc-review.googlesource.com/84421
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23770}
2018-06-28 12:53:17 +00:00
b2a7478221 Fix usage logging of TURN and STUN servers
Also adds tests, and adds a bit of logging in ParseIceServers.

Bug: chromium:718508
Change-Id: Id41ccb7cccbdab5af76e380b32b4d8ba0c4a0a72
Reviewed-on: https://webrtc-review.googlesource.com/86121
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23769}
2018-06-28 12:52:07 +00:00
72b751af0b Add PeerConnection GetRtpSender/ReceiverCapabilities
Those are static functions in the spec, so implemented as member functions
of the PeerConnectionFactory instead.

Bug: webrtc:7577, webrtc:9441
Change-Id: Iccb24180e096e713d24e7e25ecfd5d7bbd7638f9
Reviewed-on: https://webrtc-review.googlesource.com/85341
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23768}
2018-06-28 12:40:07 +00:00
0d4070a18c Remove incorrect test from api/units/
The behavior of division-by-zero is undefined, so the DivisionByZeroFails test isn't correct. As we don't need any specific behavior on division-by-zero we leave the current code untouched.
Additionally, since the DivisionFailsOnLargeSize EXPECT_DEATH checks rely on DCHECKs, we only run those when DCHECKs are enabled.

Bug: webrtc:9443
Change-Id: I0fdd7be55a7bc76b4203b2f6d5cd0ed8ac5cc688
Reviewed-on: https://webrtc-review.googlesource.com/85362
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23767}
2018-06-28 11:37:20 +00:00
183e09d23c Correct data histogram entry for incoming DC
Both incoming and outgoing datachannels should cause
the DATA_ADDED flag to be set.

This CL also moves all tests into their own file, and
improves scaffolding.

Bug: chromium:718508
Change-Id: I5c4c257ccb6f26799f7593bce8b27ebf59015b1e
Reviewed-on: https://webrtc-review.googlesource.com/85348
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23766}
2018-06-28 10:33:23 +00:00
77cc8182ae Pull GN via CIPD package.
The gn binary will be downloaded into third_party/gn.

The part about gn_win will be true only after the buildtools_revision
will be updated by the Chromium roll.

This CL has been copied from https://chromium-review.googlesource.com/c/chromium/src/+/1117264/9/DEPS.

Bug: None
Change-Id: I3fee1d9f6c39e508871798eeeb60d74ab7bc41d1
Reviewed-on: https://webrtc-review.googlesource.com/86123
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23765}
2018-06-28 10:00:40 +00:00
0bd7bf0de3 Adding ABWENoTWCC field trial
Bug: webrtc:8243
Change-Id: I80c598f6cf42c831e73ca98f68e726cf892549ce
Reviewed-on: https://webrtc-review.googlesource.com/85980
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23764}
2018-06-28 09:51:00 +00:00
546bdeda77 Add return after NOT_REACHED() in eventlog unittest.
Bug: webrtc:9457
Change-Id: If4728d05d832f72871c25ddce93a72be5089be40
Reviewed-on: https://webrtc-review.googlesource.com/86122
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23763}
2018-06-28 09:48:30 +00:00
64b17c2aca Remove StreamStatistician::IsPacketInOrder
this function is now only used in combination with StreamStatistician::IsRetransmitOfOldPacket
but IsRetransmitOfOldPacket internally checks if packet is in_order, thus making extra check unnecessary

In addition to making code simpler, removing this checks avoids
taking two extra CritSection on common code path of incoming rtp packet.

Bug: webrtc:8016
Change-Id: I050004e256b5698ce700e3416aa86b55f446a270
Reviewed-on: https://webrtc-review.googlesource.com/85361
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23762}
2018-06-28 08:44:40 +00:00
968b1dd0d7 Use field trial parser for BBR Experiment.
Bug: webrtc:8415
Change-Id: If6336b16fa55c6bd891252fc3b9c0bcce56e2fd1
Reviewed-on: https://webrtc-review.googlesource.com/83620
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23761}
2018-06-28 07:52:58 +00:00
e275174b1b Adding "is_standardized" flag to RTCStatsMember.
This will allow us to add unstandardized stats for the benefit of
native applications, and easily filter them out in chromium (without
having to maintain a whitelist that lists out every member
individually).

Unstandardized stats are declared as "RTCNonStandardStatsMember",
to make it clear in the declaration (in rtcstats_objects.h) whether
something is standardized or not.

Bug: webrtc:9410
Change-Id: I7c9804c261b7af96738e94dadeaa4b8a56b9ef2c
Reviewed-on: https://webrtc-review.googlesource.com/83743
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23760}
2018-06-28 00:43:46 +00:00
d059f2c446 Add steveanton@ as api/ and ortc/ OWNER
NOTRY=True

Bug: None
Change-Id: If64cc510402d294763806dca49e38e4758fd4dea
Reviewed-on: https://webrtc-review.googlesource.com/86084
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23759}
2018-06-28 00:24:46 +00:00
d1003d74b2 A new PeerConnection level perf test.
This test creates a one way audio and video call, allows for bandwidth
estimation to ramp up and then runs the call for 10 seconds. The
average bandwidth estimate over this time is recorded as a perf metric.
This is done at the PeerConnection level with the intention to catch
regressions related to ICE configurations. Stats are taken from
PeerConnection for BWE, and the network simulation is done with a
VirtualSocketServer.

Bug: webrtc:7668
Change-Id: Ib8a449da80fc74be1e505ac34c0c6b7479cb58db
Reviewed-on: https://webrtc-review.googlesource.com/78361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23758}
2018-06-27 23:19:05 +00:00
42f0d78f1e Roll back checking in the third_party directory
This goes back to using a subtree mirror of Chromium's third_party directory (managed by gclient).

The related scripts for syncing the files are also deleted.

The plan is to solve the conflict by creating third_party directories in subdirectories of WebRTC rather than the repo root.

Bug: webrtc:8366
Change-Id: I0b9f6a86c6d4075e2fa12c2db19aa54682ddb11f
Reviewed-on: https://webrtc-review.googlesource.com/85300
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23757}
2018-06-27 13:04:08 +00:00
67c8bcf804 Revert two instances of num_active_spatial_layers.
The variable, num_active_spatial_layers, is used to construct ssData.
This CL reverts two instances of num_active_spatial_layers not
related to ssData construction.

Bug: None
Change-Id: I4d90d4578684dfdf8bd5a39c7a2fe778fce4414c
Reviewed-on: https://webrtc-review.googlesource.com/85643
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23756}
2018-06-27 10:49:00 +00:00
bcf91808a2 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.

Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
2018-06-27 10:33:40 +00:00
81f5197512 Fix pylint presubmit errors and warnings from untouched modules.
BUG=None

Change-Id: I619dab14875e19477beb8bfb566ed1f34009c025
Reviewed-on: https://webrtc-review.googlesource.com/85960
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23754}
2018-06-27 09:31:29 +00:00
d9711098b0 Extract fft to separate target to be able to move it to third_party
fft.c is third party library and have to be moved to proper third_party
directory. So this CL will extract it to separate gn target to be able
then to move it to proper location.

Bug: webrtc:8366
Change-Id: I228ebab3c821aa7095f7aa460c23c2ea0fb98f01
Reviewed-on: https://webrtc-review.googlesource.com/85640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23753}
2018-06-27 09:08:19 +00:00
2c74d85c16 Adds enum field trial parser.
Removed the need to create a custom parser function and reuses some of
the code to reduce binary overhead of enums.

Bug: webrtc:9346
Change-Id: I51c9da713ed5456a86a2afbcf0991477bb83b894
Reviewed-on: https://webrtc-review.googlesource.com/83623
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23752}
2018-06-27 08:54:00 +00:00
b3f5aed433 Remove the flag PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS.
We now always enable any address ports, only using them if they end up
using interfaces that weren't otherwise accessible. This flag is no
longer used by downstream projects.

TBR=deadbeef@webrtc.org

Bug: None
Change-Id: I6e4e93958cbc4300811bafb103f1a2e8732274ed
Reviewed-on: https://webrtc-review.googlesource.com/85860
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23751}
2018-06-27 00:43:09 +00:00
d6eb71ef2c Use the sparse histogram in RTC_HISTOGRAM_ENUMERATION_SPARSE.
A stub of sparse histogram factory getter is added so that Chromium can
provide an implementation using base::SparseHistogram for the metrics
macro RTC_HISTOGRAM_ENUMERATION_SPARSE. The default implementation in
WebRTC reuses the non-sparse version.

Bug: None
Change-Id: Ia091ca7aaacb6baa92027cd99d821bbc8da8d780
Reviewed-on: https://webrtc-review.googlesource.com/85740
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23750}
2018-06-27 00:41:29 +00:00