Commit Graph

23581 Commits

Author SHA1 Message Date
431f14ef69 Android: Remove deprecated VideoRenderer and I420Frame
Bug: webrtc:9181
Change-Id: I9a38a35ae33ed385a9a5add0a5f51ec035019d91
Reviewed-on: https://webrtc-review.googlesource.com/71661
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23849}
2018-07-05 10:37:59 +00:00
4077814031 Removing /wd4334 from system_wrappers.
Bug: webrtc:9251
Change-Id: Ie500934e03337763d9d9a1bd436605e2d85191b9
Reviewed-on: https://webrtc-review.googlesource.com/87142
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23848}
2018-07-05 09:18:34 +00:00
e144f7fb83 Reland "Unit test for case where the number of active and configured spatial"
This reverts commit 425193b4a92f0df1f3fbea3626b9abf6a38f67ec.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> 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}

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

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:9472
Change-Id: I796909c553702a0fa19e5e16e4586f915569b134
Reviewed-on: https://webrtc-review.googlesource.com/87220
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23847}
2018-07-05 07:13:39 +00:00
7b92ceb0ee Ensure that input_frames_.size() <= kMaxBufferedInputFrames at enqueue time.
Bug: webrtc:9452
Change-Id: I6d415a2cb24461d7359ff30e6499d65d88d2b2f8
Reviewed-on: https://webrtc-review.googlesource.com/85371
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23846}
2018-07-05 07:08:59 +00:00
b1f48bcfd3 Roll chromium_revision fbdf9dddef..39b1860428 (572603:572705)
Change log: fbdf9dddef..39b1860428
Full diff: fbdf9dddef..39b1860428

Changed dependencies:
* src/base: 127a5a00ac..190ca34865
* src/build: 43e3305117..0cc28952ad
* src/ios: 5d591600cf..30df90fa70
* src/testing: 80b019a75e..caff2642b8
* src/third_party: c17ae2bb03..387fe8be07
* src/tools: f32b1d66f0..b1bad47542
DEPS diff: fbdf9dddef..39b1860428/DEPS

No update to Clang.

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

Change-Id: I965ff5606b099481ee39f3abcdbcad228a94c9fc
Reviewed-on: https://webrtc-review.googlesource.com/87205
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23845}
2018-07-05 05:12:19 +00:00
9a7eddd9fa Roll chromium_revision 1a5890105a..fbdf9dddef (572487:572603)
Change log: 1a5890105a..fbdf9dddef
Full diff: 1a5890105a..fbdf9dddef

Changed dependencies:
* src/base: 7e26365993..127a5a00ac
* src/ios: e923f6e8b0..5d591600cf
* src/testing: e62a1f4d12..80b019a75e
* src/third_party: f1f650b31a..c17ae2bb03
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5abd99f5f7..82213060d5
* src/third_party/harfbuzz-ng/src: 2cb075fe26..957e775663
* src/tools: a080e360f4..f32b1d66f0
DEPS diff: 1a5890105a..fbdf9dddef/DEPS

No update to Clang.

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

Change-Id: I7af9a9459bf78c2e153e1685624276027cfc6401
Reviewed-on: https://webrtc-review.googlesource.com/87163
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23844}
2018-07-04 17:08:16 +00:00
451b29c49c Make a copy of the frame if the processing has to be posted.
Since the frame is processed on the same thread as the decoding happens
on, keeping a reference to the frame may cause deadlocks on some
implementations.

Longer term, we should probably move the frame processing to a separate
thread but this is an easy fix for now.

Bug: b/110246814
Change-Id: I251737e2188e1755d45b35165586d1b0daf14595
Reviewed-on: https://webrtc-review.googlesource.com/87104
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23843}
2018-07-04 14:11:24 +00:00
f4aeb891b7 Android: Handle StartRecording() failure gracefully
This CL also adds a test to test the behavior when StartRecording()
fails, which is the case when e.g. the microphone is already in use.

Bug: webrtc:9491
Change-Id: Ifce60ce5e9b7fa7521ca5c9fe20794233456b9ce
Reviewed-on: https://webrtc-review.googlesource.com/87105
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23842}
2018-07-04 14:08:54 +00:00
641ddf2915 Make rtc_event_log2text work on stdin if no input file specified
Bug: webrtc:9490
Change-Id: Ie235d156cef842b2333f621ae98e14aa1b4663a5
Reviewed-on: https://webrtc-review.googlesource.com/87101
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23841}
2018-07-04 12:42:01 +00:00
1f5de53a8c Fuzz key presses in APM
This calls webrtc::AudioProcessing::set_stream_key_pressed, which
opens up a lot of code paths in the transient suppressor.

The change breaks historical fuzzer test cases.

Bug: webrtc:9413
Change-Id: I1f593a98286c7e7c0fc6751d16df40ad813dbd70
Reviewed-on: https://webrtc-review.googlesource.com/86950
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23840}
2018-07-04 11:14:14 +00:00
f9c2952837 Removing warning suppression flags in common_audio/.
Bug: webrtc:9251
Change-Id: I9cae182ceb5e6bd3d6a34dc1a336ee3900f4cc98
Reviewed-on: https://webrtc-review.googlesource.com/86946
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23839}
2018-07-04 11:13:09 +00:00
e12c1fe8d9 Removing warning suppression flags from pc/.
Bug: webrtc:9251
Change-Id: Ic12126fc03309448fe71a17e6b65343949496f4f
Reviewed-on: https://webrtc-review.googlesource.com/86820
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23838}
2018-07-04 10:35:27 +00:00
36c69d5114 NetEq fuzzers: Set max length in BUILD config rather than in the code
This is the preferred way.

NOTRY=True

Bug: none
Change-Id: I305d4d9cb7b66e01427958e1416d672badd72af0
Reviewed-on: https://webrtc-review.googlesource.com/86948
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23837}
2018-07-04 10:24:03 +00:00
496cedfe56 AEC3: Reverberation model: Changes on the decay estimation.
In this CL we have introduced changes on the estimation of the decay involved in the exponential modeling of the reverberation. Specifically, the instantaneous ERLE has been tracked and used for adapting faster in the regions when the linear filter is performing well. Furthermore, the adaptation is just perform during render activity.


Change-Id: I974fd60e4e1a40a879660efaa24457ed940f77b4
Bug: webrtc:9479
Reviewed-on: https://webrtc-review.googlesource.com/86680
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23836}
2018-07-04 10:04:32 +00:00
cb96ad8f0e Add ParsedPayload::video_header() accessor.
Preparation CL to remove RTPTypeHeader.

Bug: none
Change-Id: I695acf20082b94744a2f6c7692f1b2128932cd79
Reviewed-on: https://webrtc-review.googlesource.com/86132
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23835}
2018-07-04 09:31:21 +00:00
d92288f5ba Add experimental shortened 2-temporal-layer setting
Also adjust to base-layer fraction for the shortened 3-tl pattern to be
60%, just like the 2-tl setting.

This CL removes direct use of the allocation matrix and moves it behind
a static getter.

Bug: webrtc:9477
Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5
Reviewed-on: https://webrtc-review.googlesource.com/86681
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23834}
2018-07-04 09:25:21 +00:00
c05bd738d6 Limit fuzzer input size for comfort noise decoder fuzzer
This avoids fuzzer timeouts on the bot.

NOTRY=True

Bug: chromium:857404
Change-Id: I480c53f005536029c667b9f41aab3ecaca14d125
Reviewed-on: https://webrtc-review.googlesource.com/86945
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23833}
2018-07-04 09:21:41 +00:00
265b868e23 Remove non-implemented function signatures from RtpFrameReferenceFinder.
Bug: none
Change-Id: I17e7cb6300cc6f4c82517d6a2059c7be6d4fb9ad
Reviewed-on: https://webrtc-review.googlesource.com/86882
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23832}
2018-07-04 09:07:22 +00:00
f0e0d75e6e Adding CheckNoWarningSuppressionFlagsAreAdded.
This PRESUBMIT check will ensure that WebRTC does not regress on
the amount of warning suppression flags used.

At the moment it only checks for //build/config/clang:extra_warnings.

Error message:

** Presubmit ERRORS **
Usage of //build/config/clang:extra_warnings is discouraged in WebRTC.
If you are not adding this code (e.g. you are just moving existing code) or you want to add an exception,
you can add a comment on the line that causes the problem:

"-Wno-odr"  # no-presubmit-check TODO(bugs.webrtc.org/BUG_ID)

Affected files:

  api/BUILD.gn (line 30)


Bug: webrtc:9251
Change-Id: I059cbc648ca6f6806cf5e936e0b83b72ec4f3f50
Reviewed-on: https://webrtc-review.googlesource.com/86942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23831}
2018-07-04 09:01:32 +00:00
ec64217e56 AEC3: Simplified suppression gain calculation
This change simplifies the calculation of the suppression gains.
It also contains a new tuning of the suppressor.

The suppressor behavior is tuned by setting echo-to-nearend ratios
for when the suppressor is to be fully transparent and for when to
fully suppress. An echo-to-masker value determines when the signal
is masked by noise. These three values are specified for low and
high frequencies.

Change-Id: I108e83c8f2a35462085a3fabaebcc02fa3103607
Bug: webrtc:9482
Reviewed-on: https://webrtc-review.googlesource.com/86021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23830}
2018-07-04 07:07:55 +00:00
23cd45ac2e webrtcvideoengine_unittest: Use RtpHeaderParser class for parsing rtp header.
Removes parsing code in ParseRtpPacket.

Bug: none
Change-Id: I190b3f8abd5f1881cf19e14b7f31cc8f85b5f156
Reviewed-on: https://webrtc-review.googlesource.com/86823
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23829}
2018-07-04 06:59:45 +00:00
b3d2119af7 Roll chromium_revision 79cbcdf6fb..1a5890105a (572378:572487)
Change log: 79cbcdf6fb..1a5890105a
Full diff: 79cbcdf6fb..1a5890105a

Changed dependencies:
* src/base: b321921624..7e26365993
* src/build: 91b88ae14f..43e3305117
* src/ios: 9615de8dfb..e923f6e8b0
* src/testing: b3ea40231d..e62a1f4d12
* src/third_party: 3b7f89fc9f..f1f650b31a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/10e0a3798e..5abd99f5f7
* src/tools: fec28f646d..a080e360f4
DEPS diff: 79cbcdf6fb..1a5890105a/DEPS

No update to Clang.

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

Change-Id: I8ebaa2a2ebf190ea9cd613afb3ad76475fabf49d
Reviewed-on: https://webrtc-review.googlesource.com/87066
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23828}
2018-07-04 06:08:23 +00:00
57900cb933 Roll chromium_revision 6df2efa531..79cbcdf6fb (572277:572378)
Change log: 6df2efa531..79cbcdf6fb
Full diff: 6df2efa531..79cbcdf6fb

Changed dependencies:
* src/base: 7ffd231167..b321921624
* src/build: 798d88a968..91b88ae14f
* src/ios: bbb1a1380e..9615de8dfb
* src/testing: c2d62722b5..b3ea40231d
* src/third_party: 2ed97d2760..3b7f89fc9f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9f46241c6c..10e0a3798e
* src/third_party/libvpx/source/libvpx: 583859d739..03abd2c8f3
* src/tools: 62a39dba9a..fec28f646d
DEPS diff: 6df2efa531..79cbcdf6fb/DEPS

No update to Clang.

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

Change-Id: I7042e8b3b43cec65b7d5626a7945102d7407fb32
Reviewed-on: https://webrtc-review.googlesource.com/87040
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23827}
2018-07-03 23:09:20 +00:00
43745937a8 Adding shampson (me) as an owner to pc/ & api/.
With deadbeef removed from these OWNERS files, Steve is the only OWNER
on our team. I'm adding myself, because I have worked in these
directories and it makes sense to be able to distribute the code
reviews.

NOTRY=True

Bug: None
Change-Id: I48e88a07ee42254d937391f500f273656853d98b
Reviewed-on: https://webrtc-review.googlesource.com/86980
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23826}
2018-07-03 20:39:17 +00:00
defa7a8049 NetEq: Handle nested RED packets
This CL makes NetEq handle nested RED packets without crashing. Nested
RED packets mean that the block PT (see
https://tools.ietf.org/html/rfc2198.html#section-3) in the RED packet
is also set to the RED PT. This implies a nested RED packet, which is
not supported. Instead, all payloads in a RED packet that have the RED
PT will be discarded.

Bug: chromium:851662
Change-Id: I86ec257e60fb8076e3574ac5a4a1ca50196f6b34
Reviewed-on: https://webrtc-review.googlesource.com/86824
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23825}
2018-07-03 20:27:57 +00:00
ec20710250 Adding ICE configurations to the PC perf test.
This adds multiple ICE configurations to the PeerConnection ramp up
performance test. The configurations added are:
-TLS TURN
-UDP TURN
-UDP peer to peer
-TCP peer to peer

Bug: webrtc:7668
Change-Id: If110d99e4d83b56ac093a1e43956292f1916a1bf
Reviewed-on: https://webrtc-review.googlesource.com/85140
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23824}
2018-07-03 19:45:27 +00:00
c5762130a1 Roll chromium_revision ce19c6d80b..6df2efa531 (572160:572277)
Change log: ce19c6d80b..6df2efa531
Full diff: ce19c6d80b..6df2efa531

Changed dependencies:
* src/base: 372f1b7ce6..7ffd231167
* src/build: 7ac293430b..798d88a968
* src/ios: c3c2c951d7..bbb1a1380e
* src/testing: 408bbbaa8a..c2d62722b5
* src/third_party: e808c54bee..2ed97d2760
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/cbfa46069e..9f46241c6c
* src/third_party/harfbuzz-ng/src: 957e775663..2cb075fe26
* src/tools: 57c608d921..62a39dba9a
DEPS diff: ce19c6d80b..6df2efa531/DEPS

No update to Clang.

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

Change-Id: Ib7317fac60a2b6df9132e0d7f51faf768b1c4d03
Reviewed-on: https://webrtc-review.googlesource.com/86924
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23823}
2018-07-03 18:14:24 +00:00
13171bdba8 Adds debug printing for congestion controllers.
These are useful for plotting creating data files that can be used to
visualize and debug congestion controller behavior.

Bug: webrtc:9467
Change-Id: I75b03a309b4b7d562fefe82a828ae1e6a9f069c8
Reviewed-on: https://webrtc-review.googlesource.com/86126
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23822}
2018-07-03 17:00:24 +00:00
d000b0a32e Move RTC_CHECK_OP error message construction out of header file.
This simplifies the logic, prevents emitting code for every pair of
argument types to RTC_CHECK_OP and partially unblocks removing streams from
the check code altogether.

Bug: webrtc:8982
Change-Id: Ib6652ac9a342e4471c12574a79872833cc943407
Reviewed-on: https://webrtc-review.googlesource.com/86544
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23821}
2018-07-03 15:21:13 +00:00
588527b295 Add sprang@ as owner for simulcast.cc/h
The previous attempt caused issues:
https://webrtc-review.googlesource.com/c/src/+/86900

Let's try it with a separate file instead.

Bug: None
Change-Id: I57dc4dceca1ea4b73003d4202e9b75ee469e5adc
Reviewed-on: https://webrtc-review.googlesource.com/86940
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23820}
2018-07-03 15:00:33 +00:00
23f71a8144 Remove usage of //build/config/clang:extra_warnings.
Bug: webrtc:9251
Change-Id: I13522eafff1a4d6a9fe909c305efa0e4581a56c7
Reviewed-on: https://webrtc-review.googlesource.com/86880
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23819}
2018-07-03 12:35:49 +00:00
3bc977a420 Revert "Add sprang@ as owner of simulcast.cc/h"
This reverts commit 91fc422d0884da7b2b64549791b043258d7c5555.

Reason for revert:
owners.SyntaxErrorInOwnersFile: /b/s/w/ir/cache/builder/presubmit/src/media/OWNERS:11 syntax error: per-file globs cannot span directories or use escapes: "per-file engine/simulcast*=sprang@webrtc.org"

Original change's description:
> Add sprang@ as owner of simulcast.cc/h
> 
> Bug: None
> Change-Id: I41817d76726f526afcde5c934abd1f401b180a3c
> Reviewed-on: https://webrtc-review.googlesource.com/86682
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23812}

TBR=sprang@webrtc.org,stefan@webrtc.org

Change-Id: Ia95e48769c4c2c81b3e3758038c8bfcb8c352589
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/86900
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23818}
2018-07-03 11:55:00 +00:00
e40b437401 Discard frame self-dependency when parsing genric frame descriptor
Bug: chromium:859281
Change-Id: Ieb96f633a93f4f2e498bb1949339e239184bce9d
Reviewed-on: https://webrtc-review.googlesource.com/86545
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@{#23817}
2018-07-03 10:28:05 +00:00
a436bb4a99 Roll chromium_revision f6935ecdd2..ce19c6d80b (572058:572160)
Change log: f6935ecdd2..ce19c6d80b
Full diff: f6935ecdd2..ce19c6d80b

Changed dependencies:
* src/base: acf85db8da..372f1b7ce6
* src/ios: 39abda7785..c3c2c951d7
* src/testing: c03efa3c5e..408bbbaa8a
* src/third_party: 3f1a8b2a7f..e808c54bee
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/153acbd707..cbfa46069e
* src/third_party/depot_tools: 605dd3126a..5484b866dc
* src/tools: 916b90567c..57c608d921
DEPS diff: f6935ecdd2..ce19c6d80b/DEPS

No update to Clang.

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

Change-Id: I944a967d227fd9e4578559c14b3c798f2102473b
Reviewed-on: https://webrtc-review.googlesource.com/86809
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23816}
2018-07-03 10:09:25 +00:00
46f858a626 Fix fuzzer-found overflow in AGC1
Much like https://bugs.chromium.org/p/chromium/issues/detail?id=855900,
the int32 gain table isn't always small enough for plain multiplication
with an int16.

This appears fixable through regular fixed-point arithmetic (multiply
out[i][n] by integer and fractional part of gain separately), but it's
less readable.

Bug: chromium:858989
Change-Id: Ie5aac25fd0cca4e51858cba69bde06c54a5d31bf
Reviewed-on: https://webrtc-review.googlesource.com/86602
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23815}
2018-07-03 09:56:34 +00:00
a8eb1e619e roll_deps: Accept any prefix (like 'git_revision:'), not only 'version:' for CIPD
`gclient setdep` was changed in https://chromium-review.googlesource.com/1123940
to support any prefix as well, but note that that was a backwards incompatible
change, because it now requires the prefix to be passed. So we just stop stripping
the prefix in this CL.

Also clarify the error when a CIPD dep is present in WebRTC and missing in Chromium.

No-Try: True
TBR: phoglund@webrtc.org
Bug: webrtc:9470, chromium:858978
Change-Id: I5e42bbda04db37a628a0ac1de69667b9a30dd793
Reviewed-on: https://webrtc-review.googlesource.com/86280
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23814}
2018-07-03 09:41:53 +00:00
a3b6601c9b Make ReceiveSendsFromThread use Dispatch
The ReceiveSendsFromThread function calls the OnMessage function.
However, instead we should be calling the Dispatch function which does the same thing as the OnMessage function except that it also does additional logging.
This logging is being missed for the cases where we call functions on a thread using the Invoke function.
Calling Dispatch fixes the issue and makes sure that this code path is consistent with other paths of posting to a thread like Post function which goes through Dispatch ultimately.

Bug: None
Change-Id: I75a5c8b464226cf4de60a3d19dff48f9e6197cca
Reviewed-on: https://webrtc-review.googlesource.com/85885
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23813}
2018-07-03 09:09:43 +00:00
91fc422d08 Add sprang@ as owner of simulcast.cc/h
Bug: None
Change-Id: I41817d76726f526afcde5c934abd1f401b180a3c
Reviewed-on: https://webrtc-review.googlesource.com/86682
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23812}
2018-07-03 07:28:03 +00:00
b8926b05c2 Roll chromium_revision a1981d69db..f6935ecdd2 (571936:572058)
Change log: a1981d69db..f6935ecdd2
Full diff: a1981d69db..f6935ecdd2

Changed dependencies:
* src/base: 21429bcfa1..acf85db8da
* src/ios: c07ee7f40e..39abda7785
* src/testing: 1764a90c9d..c03efa3c5e
* src/third_party: c19717098e..3f1a8b2a7f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3836c556c4..153acbd707
* src/third_party/depot_tools: 621c9d28c3..605dd3126a
* src/tools: 79432e6c3f..916b90567c
DEPS diff: a1981d69db..f6935ecdd2/DEPS

No update to Clang.

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

Change-Id: I9625bb5206fd6ddb0ecfb69d1e6b2032ea604e76
Reviewed-on: https://webrtc-review.googlesource.com/86800
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23811}
2018-07-03 01:11:03 +00:00
98badbcd9f Add VP9 profile negotiation to SDP
This CL adds VP9 profile information in SDP. It adds the necessary fields and
enums to codec containers.

Additional profiles will be followed.

Bug: webrtc:9376
Change-Id: I78574714f06f8087262a71dd64c01f31a229dd54
Reviewed-on: https://webrtc-review.googlesource.com/81960
Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23810}
2018-07-02 23:38:41 +00:00
0ea751539e Fix a bug in TurnServer that causes flakiness in webrtc_perf_tests.
When a TCP TURN port is destroyed, a TURN refresh request with zero
lifetime is first sent to release the TURN allocation at the server,
and the underlying TCP connection is closed afterwards.

The closing of the TCP connection is handled first by the
VirtualSocketServer in our test infrastructure, and the corresponding
server socket is asynchronously destroyed at the TURN server. The
refresh request is however still passed to this server socket and
further signaled to the TURN server, which fails a DCHECK. The
server implementation should disable any firing of signals from a
server socket to be destroyed.

The bug id is set to None since this is a one-liner CL.

TBR=pthatcher@webrtc.org

Bug: None
Change-Id: Ib457b3800511a322ef69d67c71f2de05f3d67967
Reviewed-on: https://webrtc-review.googlesource.com/86501
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23809}
2018-07-02 18:51:23 +00:00
312466a204 Roll chromium_revision c20726850b..a1981d69db (571826:571936)
Change log: c20726850b..a1981d69db
Full diff: c20726850b..a1981d69db

Changed dependencies:
* src/base: 0d31d15d4c..21429bcfa1
* src/build: 213a0e3999..7ac293430b
* src/ios: e7915b2a8e..c07ee7f40e
* src/testing: 20f36f2392..1764a90c9d
* src/third_party: fc2824c48d..c19717098e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/34f0d7e2e4..3836c556c4
* src/third_party/depot_tools: 024a331759..621c9d28c3
* src/tools: 0935834c72..79432e6c3f
DEPS diff: c20726850b..a1981d69db/DEPS

No update to Clang.

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

Change-Id: I3d654931c2b8217e739c18c06f696fed1e44f10b
Reviewed-on: https://webrtc-review.googlesource.com/86662
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23808}
2018-07-02 18:11:51 +00:00
9f1de69008 Add ADAPTER_TYPE_ANY in AdapterType.
ADAPTER_TYPE_ANY can be used to set the network ignore mask if an
application does not want candidates from the any address ports, the
underlying network interface types of which are not determined in
gathering. The ADAPTER_TYPE_ANY is also given the maximum network cost
so that when there are candidates from explicit network interfaces,
these candidates from the any address ports as backups, if they ever
surface, are not preferred if the other candidates have at least the
same network condition.

Bug: webrtc:9468
Change-Id: I20c3a40e9a75b8fb34fad741ba5f835ecc3b0d92
Reviewed-on: https://webrtc-review.googlesource.com/85880
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23807}
2018-07-02 17:59:11 +00:00
6b33e60213 In ULP FEC fuzzer test, make sure sequence number is not the same as previous sequence number.
Bug: chromium:859265
Change-Id: I9acb9a177dfed3830ead0ba5a16ee4310f4d2b5b
Reviewed-on: https://webrtc-review.googlesource.com/86547
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23806}
2018-07-02 15:51:10 +00:00
4d01146f16 Prepare AGC2 for analog gain changes.
1. Adds support for Reset calls in AGC2. The AGC will be reset during
   analog gain changes.
2. Allows AdaptiveModeLevelEstimator to return estimates > 0. This can
   happen if the signal gain is too high. It's needed for letting the
   analog AGC know that the gain is too high.

Bug: webrtc:7494
Change-Id: I38def17c21cc01c36aaea79a2401d8c2f289407b
Reviewed-on: https://webrtc-review.googlesource.com/79360
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23805}
2018-07-02 15:25:49 +00:00
5afa61cf15 NetEq: Fold GetDecisionSpecialized into GetDecision
Now that there is only one implementation of the decision logic, there
is no longer any need to have GetDecisionSpecialized being separate.

Bug: webrtc:9421
Change-Id: Id364ce09ac05d106652d749502058056f11bba27
Reviewed-on: https://webrtc-review.googlesource.com/86604
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23804}
2018-07-02 14:51:09 +00:00
9f2e624024 Break out NetEqEventLogInput to separate source files
Building NetEqEventLogInput requires protobuf support, while building
NetEqRtpDumpInput located in the same file does not. This makes both
classes unusable when protobuf support is lacking. With this CL, the
NetEqEventLogInput is broken out into separate files, to allow usage
of NetEqRtpDumpInput even when protobufs are not supported.

Bug: webrtc:9421
Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f
Reviewed-on: https://webrtc-review.googlesource.com/84587
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23803}
2018-07-02 14:15:29 +00:00
64cb83bbd9 Flags and settings for AGC2 in AgcManagerDirect.
This CL adds two flags to audioproc_f. The flags control
AgcManagerDirect. The flags are
'--experimental_agc_agc2_level_estimator' and
'--experimental_agc_agc2_digital_adaptive'.

After this CL, the flags are be applied to AgcManagerDirect. The flags
have no effect in release-mode. They cause a crash in debug-mode.

In an upcoming CL, '--experimental_agc_agc2_level_estimator 1' will
replace the speech level estimation in ExperimentalAgc with that of
AGC2.

'--experimental_agc_agc2_digital_adaptive 1' will replace the digital
gain selection and application with that of AGC2.

These audioproc_f will activate both new settings:

./out/Target/audioproc_f --agc 1 --experimental_agc 1
--experimental_agc_agc2_digital_adaptive 1
--experimental_agc_agc2_level_estimator 1 --simulate_mic_gain 1
--simulated_mic_kind 2

See also https://webrtc-review.googlesource.com/c/src/+/79360

Bug: webrtc:7494
Change-Id: If0e65893dffdddb312e553787b8cedaf9a334323
Reviewed-on: https://webrtc-review.googlesource.com/86548
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23802}
2018-07-02 13:20:39 +00:00
5c71e74331 Add AGC1-compliant fake recording device.
The AGC submodule of APM changes analog gain. These gain changes are
typically ignored by the test tool audioproc_f.

There is an option of the test tool to take action on the gain
changes.  It's the '--simulate_mic_gain' option. The option converts
the analog gain to a digital gain. The digital gain is applied to the
capture stream.

This change adds a new simulated microphone kind. The new microphone
has a gain curve defined by
modules/audio_processing/agc/gain_map_internal.h. That gain curve
defines how AGC1 expects a microphone to behave.

Bug: webrtc:7494
Change-Id: Ifb3f54a8c6f8c001a711fa977f39f32413069780
Reviewed-on: https://webrtc-review.googlesource.com/86128
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23801}
2018-07-02 12:29:36 +00:00
c167673c4d Add more ApmDataDumper dumps to AGC.
We dump the compression level from AgcManagerDirect.

We use the same names and structure as in
GainControlForExperimentalAgc.

This is to get Apm dump file names to match in the upcoming AGC
changes: https://webrtc-review.googlesource.com/c/src/+/79360

TBR: alessiob@webrtc.org
Bug: webrtc:7494
Change-Id: I1e6260ea48ffc43f709e4b0c97f843ad9c3d1824
Reviewed-on: https://webrtc-review.googlesource.com/86546
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23800}
2018-07-02 11:00:13 +00:00