Commit Graph

24046 Commits

Author SHA1 Message Date
704a7bd55a Use rtc::saturated_cast instead of static_cast in VCMFecMethod
Bug: webrtc:9439
Change-Id: Ia76a37ab5ae4871c7437b1b4c242556cd33bee40
Reviewed-on: https://webrtc-review.googlesource.com/92701
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24214}
2018-08-07 17:08:42 +00:00
9129565879 Adds functionality to add delay spikes in SimulatedNetwork.
Bug: webrtc:9467
Change-Id: Ifddafa65a9e18a3131fc0415764599740fab2db4
Reviewed-on: https://webrtc-review.googlesource.com/92089
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24213}
2018-08-07 16:45:19 +00:00
5ca90f55ae Ensures that packets_lost is always positive.
This is a quick fix to ensure that we don't wrap the value.  A proper
solution would be to ensure that the packets_lost field is signed and
handled as signed at all places it's used.

Bug: webrtc:9598
Change-Id: I3622f2a61aa3af57db6292ef4c0a8e97c4833aa4
Reviewed-on: https://webrtc-review.googlesource.com/92881
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24212}
2018-08-07 16:36:06 +00:00
f0d5fc9601 Add steveanton@ and qingsi@ as rtc_base OWNERs
We both frequentyly work on this code and much of it is intertwined with
pc/ and p2p/ code which we have OWNERs for already.

NOTRY=True

Bug: None
Change-Id: If56ebca6ef44cf9b7837e8d4bc3afa367a5d5216
Reviewed-on: https://webrtc-review.googlesource.com/90084
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24211}
2018-08-07 15:47:05 +00:00
ffd9293cb7 Add dependency on google-truth for Android.
This dependency has been copied from Chromium.

Bug: webrtc:9594
Change-Id: Ida86d73a39ffa14c92dcfd4783d95e08857b3da5
Reviewed-on: https://webrtc-review.googlesource.com/92397
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24210}
2018-08-07 15:07:53 +00:00
5f7d00eb3d Release audio unit when ios audio device failed to initialize playout and recording.
TBR=henrika@webrtc.org

Bug: webrtc:9552
Change-Id: I7c3e0c1c2126603e7b1cc412cb37cac57eb3cdbf
Reviewed-on: https://webrtc-review.googlesource.com/90085
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24209}
2018-08-07 14:34:12 +00:00
c2a028887f Enable audio in video_quality_test.
Allows enabling audio for RunWithAnalyzer method, and prints out audio jitterbuffer performance stats. Also fixes for RunWithRenderer when enabling audio (seg-faulted).

Bug: b/112299470
Change-Id: Ic7c0de1c455891f38cca317001c6c216e82f6ec3
Reviewed-on: https://webrtc-review.googlesource.com/92800
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24208}
2018-08-07 13:49:04 +00:00
b3e2c8eb1b Roll chromium_revision 39d45f08f5..474eca0589 (580730:581204)
Change log: 39d45f08f5..474eca0589
Full diff: 39d45f08f5..474eca0589

Changed dependencies:
* src/base: d182366d3b..8385797d0c
* src/build: b6d04f7ca1..f24ca38e53
* src/ios: e369aedb22..037c6dcb8c
* src/testing: 067c5fe80f..75bb85f253
* src/third_party: c60fb24bae..51ecceccb2
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d30f10814b..3cb00fbd56
* src/third_party/depot_tools: 82bb756217..2ebf9fdade
* src/tools: 734ee5dbb6..f01fd4bf32
DEPS diff: 39d45f08f5..474eca0589/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None

Change-Id: I8055910abaf2de3a224564c1b719c2baefe90c24
Reviewed-on: https://webrtc-review.googlesource.com/92841
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24207}
2018-08-07 13:41:09 +00:00
8e06419ee9 Makes units constexpr when possible.
This makes the constructor and the unchecked create functions
constexpr on the unit classes Timestamp, TimeDelta, Datarate and
DataSize. This allows using the units in constexpr constants.
Unchecked access methods are made constexpr as well. Making them
usable in static asserts.

Constexpr create functions for checked construction is added in
a separate CL.

Bug: webrtc:9574
Change-Id: I605ae2e8572195dbb2078c283056208be0f43333
Reviewed-on: https://webrtc-review.googlesource.com/91160
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24206}
2018-08-07 11:30:21 +00:00
29835996e9 Move spl_sqrt_floor dep to proper third_party directory
Bug: webrtc:8366
Change-Id: I326af5251dd88136dcc722e0ba1a2f9a8aebcf89
Reviewed-on: https://webrtc-review.googlesource.com/90405
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24205}
2018-08-07 10:33:13 +00:00
7008287219 Delete struct webrtc::PacketTime.
Replaced by a int64_t representing time in us.

Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
2018-08-07 10:07:15 +00:00
10d70caa13 Fix guards for headers in third party
Bug: webrtc:8366
Change-Id: I86309265c822dd4430c5578d813bdddc77102d05
Reviewed-on: https://webrtc-review.googlesource.com/90416
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24203}
2018-08-07 09:39:06 +00:00
9a29c03355 Fix random crashes - invariant broken in LinkedSet (LRU) implementation.
Root cause: IsNewSequenceNumber didn't respect strict weak ordering requirements.
            (e.g. 0, 0x1000, 0x2000, ... 0x9000 are increasing, but 0x9000 < 0)
Solution: Unwrap the sequence numbers into int64_t for proper sorting.

This CL also introduce a simpler interface,
which does a better job at hiding implementation details.

Bug: webrtc:9575
Change-Id: Ic9922426de32278e8b51c6ecef8e2efeb0997512
Reviewed-on: https://webrtc-review.googlesource.com/91165
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24202}
2018-08-07 09:18:41 +00:00
264bee8bab Remove memcheck.
Since the linux_memcheck trybot is no more, this CL removes all the
code needed to make it work.

Bug: webrtc:7737, webrtc:8356, webrtc:9570
Change-Id: I09a9467b8bf895146a3384c2c915b54662721af6
Reviewed-on: https://webrtc-review.googlesource.com/90863
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24201}
2018-08-07 07:40:08 +00:00
e22a26f6f1 Add 2 more OWNERS to tools_webrtc.
Bug: None
Change-Id: I3550652ac111363d2f0e29fb97e3804c8b5d92af
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/90409
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24200}
2018-08-07 06:06:19 +00:00
eb73a7bd16 Removes unnecessary webrtc_cc namespaces.
Bug: webrtc:9586
Change-Id: I6407ee465d725d7469c409e5bea1c55354ab7f95
Reviewed-on: https://webrtc-review.googlesource.com/92385
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24199}
2018-08-06 17:18:45 +00:00
13ef7d25f6 Adds feedback only mode to GoogCC.
This CL adds a factory for creating a GoogCC network controller that
can be used without RTCP specific messages. This prepares for enabling
use of other underlying protocols as long as they can provide per
packet feedback.

Bug: None
Change-Id: I6671181949d97abd18843d0f4edf75040cc3f007
Reviewed-on: https://webrtc-review.googlesource.com/84583
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24198}
2018-08-06 15:43:37 +00:00
f70bc5eeff Removes pause check from RoundRobinPacketQueue.
This CL removes a check in RoundRobinPacketQueue::FinalizePop. This
check will trigger if a the pause state is changed in PacedSender while
a packet is sent. This is a rare occurrence but would yield flaky
behavior. The check should not be required for the code to function
since the paused state is not read in FinalizePop other than for this
check.

Bug: webrtc:9586
Change-Id: Ib9476168eb637dc2f9710d0592bed92c4b03dacb
Reviewed-on: https://webrtc-review.googlesource.com/92090
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24197}
2018-08-06 15:22:41 +00:00
fd77b78821 Delete RtpReceiverImpl::CheckPayloadChanged.
Also delete related code in RtpReceiverAudio, RtpReceiverVideo and
RtpPayloadRegistry.

Only intended change in behavior is that packets with unknown payload
types are not discarded at this level of the stack. They are discarded
higher up, in Channel::ReceivePacket (audio) and
RtpVideoStreamReceiver::ReceivePacket (video).

Bug: webrtc:8995
Change-Id: I807997120bb40a95b0575c55db6e20a0cac651bf
Reviewed-on: https://webrtc-review.googlesource.com/92087
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24196}
2018-08-06 15:08:12 +00:00
0047bce0a9 Remove linux_internal_compile_lite from CQ.
TBR: phoglund@webrtc.org
No-Try: True
Bug: None
Change-Id: I83a7613f3fa81e36ea09dcb13082316de39867d6
Reviewed-on: https://webrtc-review.googlesource.com/92623
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24195}
2018-08-06 14:59:31 +00:00
508e23421f Remove unnecessary //base:base_java dependencies.
WebRTC code shouldn't depend on Chromium Android base code.

Bug: None
Change-Id: Ie094f26e4ee855769c9c5276bbb47242aae9c217
Reviewed-on: https://webrtc-review.googlesource.com/92387
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24194}
2018-08-06 12:04:35 +00:00
15f0a12b83 Allow releaseCallback to be null in JavaI420Buffer#wrap.
R=magjed

Bug: None
Change-Id: I3d57198dd0b8e0575af61b0dac439e3753a2360a
Reviewed-on: https://webrtc-review.googlesource.com/92386
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24193}
2018-08-06 11:39:52 +00:00
46671402ff Roll chromium_revision d6a13562c8..39d45f08f5 (580628:580730)
Change log: d6a13562c8..39d45f08f5
Full diff: d6a13562c8..39d45f08f5

Changed dependencies:
* src/base: 449b9ac452..d182366d3b
* src/build: a959e72727..b6d04f7ca1
* src/ios: 5bf0d11ec8..e369aedb22
* src/testing: aa21329c42..067c5fe80f
* src/third_party: 265f32ab48..c60fb24bae
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/27928c385e..d30f10814b
* src/third_party/depot_tools: 29b7b99e5d..82bb756217
* src/third_party/libvpx/source/libvpx: 2d79df4940..6fd9d0244c
* src/tools: 82c5a09abd..734ee5dbb6
DEPS diff: d6a13562c8..39d45f08f5/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: I7694d14a32191561a0b34cade9f0867a56b62a44
Reviewed-on: https://webrtc-review.googlesource.com/92522
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24192}
2018-08-04 03:26:10 +00:00
ff52e88a74 Revert "Extract color space from Vp8 decoder"
This reverts commit fad2aa23b406ca5d85b8aa9ab891f2067e51c782.

Reason for revert: There seems to be a mismatch with Chrome's default for VP8.

Original change's description:
> Extract color space from Vp8 decoder
> 
> Makes use of ColorSpace class to extract info from Vp8 stream.
> 
> Bug: webrtc:9522
> Change-Id: Id9d46eeea5497c4da31db27bfcf2743612ae4157
> Reviewed-on: https://webrtc-review.googlesource.com/90183
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24086}

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

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

Bug: webrtc:9522
Change-Id: Ie589963159c9e7ccbc52bf3fdfcbc383656a4ca9
Reviewed-on: https://webrtc-review.googlesource.com/92500
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24191}
2018-08-04 00:26:21 +00:00
8bf477cb2a Roll chromium_revision 62642dd6fe..d6a13562c8 (580526:580628)
Change log: 62642dd6fe..d6a13562c8
Full diff: 62642dd6fe..d6a13562c8

Changed dependencies:
* src/base: 6b0c28c299..449b9ac452
* src/build: c1f37fdd0b..a959e72727
* src/ios: 7addb925d2..5bf0d11ec8
* src/testing: 8c05ec074f..aa21329c42
* src/third_party: 66bea27c42..265f32ab48
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d38bbdff04..27928c385e
* src/third_party/libFuzzer/src: d62662686b..9dfdc2758f
* src/tools: 8449a143a6..82c5a09abd
DEPS diff: 62642dd6fe..d6a13562c8/DEPS

No update to Clang.

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

Change-Id: I0c9beef104d834cc6364b369ff92758d8caddbe2
Reviewed-on: https://webrtc-review.googlesource.com/92460
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24190}
2018-08-03 20:07:05 +00:00
50eb4c44ad Adds BBR field trial to CallTest.
Bug: webrtc:8415
Change-Id: Ie0db059390fe4e079f1faa90f74f4ef53b192b6f
Reviewed-on: https://webrtc-review.googlesource.com/92383
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24189}
2018-08-03 18:07:51 +00:00
3dc427fe34 Fix remaining target name for AppRTCMobile_stubbed_video_io_test_apk.
Fixes a target name that was missed in the last CL.

TBR=phoglund

Bug: webrtc:9588
No-Try: True
Change-Id: I704325666b758cee7eb080f8628fc839ab89831d
Reviewed-on: https://webrtc-review.googlesource.com/92389
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24188}
2018-08-03 14:17:55 +00:00
c5a2862a20 Roll chromium_revision 06e87fb3dc..62642dd6fe (580149:580526)
Change log: 06e87fb3dc..62642dd6fe
Full diff: 06e87fb3dc..62642dd6fe

Changed dependencies:
* src/base: f9ce552913..6b0c28c299
* src/build: fbf9211933..c1f37fdd0b
* src/ios: ff92b8db88..7addb925d2
* src/testing: 340252637e..8c05ec074f
* src/third_party: a42c5d9439..66bea27c42
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c988ddf7b1..d38bbdff04
* src/third_party/depot_tools: 6f812e132d..29b7b99e5d
* src/tools: 0a1519743e..8449a143a6
DEPS diff: 06e87fb3dc..62642dd6fe/DEPS

No update to Clang.

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

Change-Id: I4f9e15d24a9f55352c3a774242f0d6b9c8d6148d
Reviewed-on: https://webrtc-review.googlesource.com/92422
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24187}
2018-08-03 14:04:33 +00:00
827c63a5d7 Remove Syzygy from deps
It is unused;
it was removed in 96d692b3f7
and this is blocking DEPS roll.

Bug: None
Change-Id: Idb4ae4e43c35787e2f34111356b68e41f0bdd201
Reviewed-on: https://webrtc-review.googlesource.com/92388
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24186}
2018-08-03 13:37:55 +00:00
688f7f8fc2 Fix gitignore pattern to never exclude sha1 files
For example, currently "resources/audio_coding/F02_tlm10.OUT20.sha1" would have been ignored by the pattern "**/*.OUT*".

No-Try: True
Bug: None
Change-Id: I91243a301950485cb61d5f72a00af08372ec7951
Reviewed-on: https://webrtc-review.googlesource.com/92085
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24185}
2018-08-03 12:45:45 +00:00
d54f5f5c76 Rename rtc_instrumentation_test_apk targets to end with _test_apk.
This matches Chromium pattern of naming instrumentation test apks with
a name ending in _test_apk. The old naming confuses generate_gradle.py.

Renames:
 - AppRTCMobileTest
   -> AppRTCMobile_test_apk
 - AppRTCMobileTestStubbedVideoIO
   -> AppRTCMobile_stubbed_video_io_test_apk
 - libjingle_peerconnection_android_unittest
   -> android_instrumentation_test_apk

Bug: webrtc:9588
TBR: phoglund
Change-Id: Idb82dc4bd089bc7c90e9373f7c3d572f9fd2d95a
Reviewed-on: https://webrtc-review.googlesource.com/92380
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24184}
2018-08-03 12:44:40 +00:00
0cbe05cc86 Android: Remove custom matrix helper functions in RendererCommon
Bug: webrtc:9487
Change-Id: I2b5720d55cae9684a7ef2b14cabce262a5321ef0
Reviewed-on: https://webrtc-review.googlesource.com/87820
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24183}
2018-08-03 10:57:07 +00:00
489767830b Remove definition of FEATURE_ENABLE_VOICEMAIL, FEATURE_ENABLE_PSTN.
FEATURE_ENABLE_VOICEMAIL is never used.

FEATURE_ENABLE_PSTN is used in //third_party/libjingle_xmpp but WebRTC
doesn't depend on it, so it is reasonable to assume that no one needs
to define it.

Bug: webrtc:9564
Change-Id: Idfb04081f497ef52fc5c140ffb82fa2dc7b9824d
Reviewed-on: https://webrtc-review.googlesource.com/92081
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24182}
2018-08-03 10:08:21 +00:00
7d745e5a89 Reland "Remove RTPVideoHeader::h264() accessors."
Downstream projects have been updated, so this can now be relanded.
This is a revert (and rebase) of: https://webrtc-review.googlesource.com/c/src/+/88820

Bug: none
Change-Id: I424664ddef7aeebd3c6c94ae67c7f70a342dc9a4
Reviewed-on: https://webrtc-review.googlesource.com/92082
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24181}
2018-08-03 09:16:50 +00:00
9f6450d5a5 JNI generation: Replace base::subtle::AtomicWord with std::atomic<>
This also rolls up //base in DEPS, because it needs to be landed together with
54f759310c

Bug: chromium:867475
Change-Id: I5792cb0610d2df46a9368fd3b1846583aa134b38
Reviewed-on: https://webrtc-review.googlesource.com/90404
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24180}
2018-08-03 08:54:30 +00:00
e8b4e7e53c Roll chromium_revision 4dd959af57..06e87fb3dc (579686:580149)
Change log: 4dd959af57..06e87fb3dc
Full diff: 4dd959af57..06e87fb3dc

Changed dependencies:
* src/base: 4b0b86b8b4..e15177f81a
* src/build: e9eade234d..fbf9211933
* src/ios: 69485848c6..ff92b8db88
* src/testing: 39667a68df..340252637e
* src/third_party: 74ddeed04a..a42c5d9439
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5c2d9e757a..c988ddf7b1
* src/third_party/gtest-parallel: cb3514a085..fe7f791f14
* src/tools: e26055aead..0a1519743e
DEPS diff: 4dd959af57..06e87fb3dc/DEPS

Clang version changed 337439:338452
Details: 4dd959af57..06e87fb3dc/tools/clang/scripts/update.py

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

Change-Id: I626f49a90bcf67b2ab10970bdfb02ddb75dc7387
Reviewed-on: https://webrtc-review.googlesource.com/92100
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24179}
2018-08-03 08:00:09 +00:00
7d984ee1d7 Don't allocate an array of size 0, it causes a UBSan failure
Bug: webrtc:9587
Change-Id: I56bdf3c5c8744044b2d0d1fa3531fca504ea200f
Reviewed-on: https://webrtc-review.googlesource.com/92091
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24178}
2018-08-03 07:49:46 +00:00
54bd8f54e9 Remove dead code.
This code never executes as we always get passed a nil codecSpecificInfo.

Bug: webrtc:9580
Change-Id: I5c5311c20877494978df45d409a53ad5b0e86a9b
Reviewed-on: https://webrtc-review.googlesource.com/92083
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24177}
2018-08-03 07:10:14 +00:00
e20867ff6d Add AsyncResolverFactory interface and basic implementation.
The factory is plumbed down to P2PTransportChannel and will eventually
be used to resolve hostnames. Uses of PacketSocketFacotry::CreateAsyncResolver
will eventually be migrated to use this factory instead.

Bug: webrtc:4165
Change-Id: I1c48b2ffb8649609a831eba291f67ce544bb10eb
Reviewed-on: https://webrtc-review.googlesource.com/91300
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24176}
2018-08-02 21:20:15 +00:00
da2ec40590 Always sends probes when they are generated.
This changes makes the usage of the new probe controller reflect how the
old probe controller was used. That is probes are now sent as soon as
they are generated. This is to avoid regressions in performance doe to
the timing of the sent probes.

Bug: chromium:868776
Change-Id: I722585689258c9b01e8f1dc47249b284a05a2793
Reviewed-on: https://webrtc-review.googlesource.com/91441
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24175}
2018-08-02 15:36:12 +00:00
dc6e68b4a7 Delete class TelephoneEventHandler and related code.
Followup to https://webrtc-review.googlesource.com/91125.

Bug: webrtc:7135
Change-Id: I7011cc65ac756931d8134763da57ec1bc9c584d6
Reviewed-on: https://webrtc-review.googlesource.com/91163
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24174}
2018-08-02 15:02:23 +00:00
ee1e74fb86 Fix occasional crash in iOS ADM.
RTCNativeAudioSessionDelegateAdapter has a raw pointer to AudioDeviceIOS,
and receives callbacks from RTCAudioSession and forwards them to AudioDeviceIOS.

During teardown of these components the situation can occur that the dtor for
AudioDeviceIOS has been called but the ObjC runtime has not yet dealloced
RTCNativeAudioSessionDelegateAdapter, so it's still receiving callbacks while
the pointer it keeps to AudioDeviceIOS has been invalidated.

This occasionally triggers a crash when WebRTC is shutting down.

The fix in this CL is to make sure to deregister the adapter from RTCAudioSession
_before_ the dtor for AudioDeviceIOS returns.

Bug: webrtc:9523
Change-Id: Ica85420d76efc63940472bc43e3ec71d16036ccf
Reviewed-on: https://webrtc-review.googlesource.com/90245
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24173}
2018-08-02 14:25:37 +00:00
ab4a530b87 Delete telephone-event handling from RTPReceiverAudio.
Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24172}
2018-08-02 12:55:40 +00:00
31f1596c7c Fix a warning about comparison of integers of different signs
Bug: webrtc:9583
Change-Id: I94df0efa0a0fd5666a42dbd6997ac1b00b3844bd
Reviewed-on: https://webrtc-review.googlesource.com/92040
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24171}
2018-08-02 12:23:36 +00:00
f8d81d33ed Add members for the codec agnostic descriptor to RTPVideoHeader.
TBR=danilchap@webrtc.org

Bug: webrtc:9361, webrtc:9582
Change-Id: I0303fc89bafab59e68ec81979e0e4372e79a4f51
Reviewed-on: https://webrtc-review.googlesource.com/91866
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24170}
2018-08-02 09:12:31 +00:00
e468ba77ae Delete MockRenderer and replace it with identical MockSink class.
This was left by a mistake in a previous refactoring.

R=magjed

Bug: None
Change-Id: Ia2b469e730844780fa3b9ce5540d4bdd4d10b556
Reviewed-on: https://webrtc-review.googlesource.com/91480
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24169}
2018-08-02 08:29:31 +00:00
f5cf02ac2c Use randomly generated IDs for candidate pairs in ICE event logs.
We used to use an integer-valued hash as the candidate pair ID in ICE
event logs, and the preimage of this hash contains address information.

Bug: None
Change-Id: Ib24aa89164600c62e0b0a7d771af379ace80a0e3
Reviewed-on: https://webrtc-review.googlesource.com/91920
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24168}
2018-08-01 20:08:43 +00:00
d3b7ec2e91 Allow all "token" chars from RFC 4566 when checking for legal mid names.
Previously only alphanumeric characters were allowed.

Bug: webrtc:9537
Change-Id: I3fd793ad88520b25ecd884efe3a698f2f0af4639
Reviewed-on: https://webrtc-review.googlesource.com/89388
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24167}
2018-08-01 18:20:42 +00:00
78026754a7 AEC3: Utilize shadow filter output to respond to audio path changes
This CL adds functionality to use the shadow filter output instead
of the main filter output for cases when the former is better than
the latter. One case when that happens is when there have been an
echo path change, either in the acoustic path, in the audio buffers
or due to some active audio processing effects being applied on
the device.

The CL causes less echo leaks, in particular on devices with
active render processing.

Bug: webrtc:9581,chromium:869821
Change-Id: Icb8df1b94141598da82dc188051ac59e43338938
Reviewed-on: https://webrtc-review.googlesource.com/91820
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24166}
2018-08-01 15:20:33 +00:00
d2f4e8bd90 Explicitly add -mfpu=neon to all targets that use NEON
Remove obsolete comment about Chromium not defining NEON for Android.

Semi-related fix: don't use `rtc_remove_configs` directly, `suppressed_configs` is the "public interface".

Bug: webrtc:9579
Change-Id: I512628feb462a29432f1356cfef00efe1ddaf84f
Reviewed-on: https://webrtc-review.googlesource.com/91761
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24165}
2018-08-01 13:15:42 +00:00