Commit Graph

25323 Commits

Author SHA1 Message Date
4e58444b0d Roll chromium_revision 00f78b5b14..208bb982f7 (610831:610939)
Change log: 00f78b5b14..208bb982f7
Full diff: 00f78b5b14..208bb982f7

Changed dependencies
* src/base: 637e844c7c..ec47506add
* src/build: 233906226b..01d83ecc66
* src/ios: 79ee87f08d..f3ae5bb001
* src/testing: 024dcf2894..f69111fe10
* src/third_party: eec9815377..cddbcf9ebb
* src/tools: 2f3431f65a..6d09354f13
DEPS diff: 00f78b5b14..208bb982f7/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I73503c3f22ff0d012a6a45d5c2fe8183e26dfa9a
Reviewed-on: https://webrtc-review.googlesource.com/c/112161
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25791}
2018-11-26 21:46:11 +00:00
74cdf7874d add cstring include need for strncmp
Propose resolution of Issue 10011 : (GCC) build fails desktop_capturer.cc:66:66: error: ‘strncmp’ was not declared in this scope

Bug: webrtc:10011
Change-Id: I4afdfd96f8bbc8e39380a365138ab79e237568e3
Reviewed-on: https://webrtc-review.googlesource.com/c/111885
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25790}
2018-11-26 20:49:36 +00:00
e38a5a1acb Small cleanup to mediasession_unittest.cc
- Uses GMock EXPECT_THAT(..., ElementsAreArray(...)) instead of EXPECT_EQ
      with MAKE_VECTOR.
- Removes unnecessary std::string casts.

Bug: None
Change-Id: I5411ac0a2260176acb333efe9a47660aada03acd
Reviewed-on: https://webrtc-review.googlesource.com/c/111727
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25789}
2018-11-26 20:20:52 +00:00
a6e034adb9 Rebase std::is_trivially_* with absl::is_trivially_*
std::is_trivially_* is not available on certain old STL
implementations. Using absl implementation will allow
maximized compatibility.

Bug: webrtc:10054
Change-Id: I17ed0fff44328b3d7c51d14e8c4470f1df0e66ad
Reviewed-on: https://webrtc-review.googlesource.com/c/111728
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25788}
2018-11-26 19:20:27 +00:00
622eedaf0f Bump variable sizes in response to fuzzer bug
The fuzzers detected a possible overflow in the multiplication of sum and gainQ10.
Since gainQ10 cannot be larger than 2048000 (see WebRtcIsac_kQGain2Levels) and sum cannot be larger than 2^16, a int64 is large enough to hold the result.

Bug: chromium:904909
Change-Id: Icb12821d4006aaaaf70a5735d2abd2b96f7a2f0e
Reviewed-on: https://webrtc-review.googlesource.com/c/111921
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25787}
2018-11-26 16:16:50 +00:00
b24c00f02d Add AudioProcessingCaptureStats and a level estimator replacement
This adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.

Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}
2018-11-26 15:52:14 +00:00
2918d4e309 Roll chromium_revision 7579fcbc1c..00f78b5b14 (610728:610831)
Change log: 7579fcbc1c..00f78b5b14
Full diff: 7579fcbc1c..00f78b5b14

Changed dependencies
* src/build: b471e777a1..233906226b
* src/ios: 5e2f73ff81..79ee87f08d
* src/testing: 0e30008dc2..024dcf2894
* src/third_party: 62ff7e0d07..eec9815377
* src/third_party/harfbuzz-ng/src: 1f14107f71..574d888c8a
* src/tools: bb68855c3b..2f3431f65a
DEPS diff: 7579fcbc1c..00f78b5b14/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ie26dcdb301af94f796dd3db3f9ed601175cda2b5
Reviewed-on: https://webrtc-review.googlesource.com/c/112103
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25785}
2018-11-26 13:22:02 +00:00
e97719974b Delete ChannelSend::RegisterTransport, replacing by construction argument
Bug: webrtc:9719
Change-Id: If3960de660cfa7a65c8bf9375ceb0af0a67d376c
Reviewed-on: https://webrtc-review.googlesource.com/c/111256
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25784}
2018-11-26 13:08:41 +00:00
b2533031a0 Add magjed as owner of rtc_tools.
He works on video analysis code every now and then, and EngProd
isn't much help when reviewing here.

Bug: None
Change-Id: I30b5f12584305d17d4c6a9682790fd0eda67d867
Reviewed-on: https://webrtc-review.googlesource.com/c/111881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25783}
2018-11-26 11:30:34 +00:00
856cf22996 In ReceiveStatistics use monotonic clock instead of ntp clock
for all time difference calculations.

Bug: None
Change-Id: I37f4a3c73ab275e661bedf991a471a1c2928180a
Reviewed-on: https://webrtc-review.googlesource.com/c/111884
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25782}
2018-11-26 10:31:44 +00:00
22027b92d6 Add a new Task Queue for WinUWP.
WinUWP cannot use the win task queue as post/peek message event loop
is not available. A replacement version written using stdlib compatible
with WinUWP is added as an alternative.

Change-Id: Ie9d6e6f11f395d1815d8f04633772a0c597ed30a
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/108520
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25781}
2018-11-26 10:29:12 +00:00
ff0581672e Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric
It never saw much use, and is blocking refactoring.

Histograms.xml-side cleanup:
https://chromium-review.googlesource.com/c/chromium/src/+/1344141

Bug: webrtc:7882
Change-Id: I112232a573fcd218dc7a51bfcdd7898847d14f18
Reviewed-on: https://webrtc-review.googlesource.com/c/111506
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25780}
2018-11-26 09:32:35 +00:00
8ce0d2b956 In ReceiveStatistic require callbacks during construction
Remove RegisterRtcpStatisticsCallback callback functions
saving taking an extra lock when calling callbacks.

Bug: None
Change-Id: Ib4537deffa0ab0abf597228e7c0fab7067614f6a
Reviewed-on: https://webrtc-review.googlesource.com/c/111821
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25779}
2018-11-26 09:17:21 +00:00
4c0cc5bc5f Reland Profile 2 to default profiles
This is a reland after chrome browser tests are updated.

Bug: webrtc:9376
Change-Id: I818bf5d447da7901ffe49f2c452decb89196e829
TBR: niklas.enbom@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/112060
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25778}
2018-11-26 07:48:03 +00:00
f1c194decd Roll chromium_revision d298cced6c..7579fcbc1c (610627:610728)
Change log: d298cced6c..7579fcbc1c
Full diff: d298cced6c..7579fcbc1c

Changed dependencies
* src/build: d6391e35ea..b471e777a1
* src/ios: 37ce0c91d6..5e2f73ff81
* src/testing: e78a57eb5b..0e30008dc2
* src/third_party: 6c250624d7..62ff7e0d07
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/43e8ebcaf2..17079a5cc1
* src/third_party/depot_tools: 25c4fce2ce..6c18a1afa1
* src/tools: 6b41cb2c41..bb68855c3b
DEPS diff: d298cced6c..7579fcbc1c/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ie17a099efe3aa3078eb217286444a924c2e6e4ed
Reviewed-on: https://webrtc-review.googlesource.com/c/112037
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25777}
2018-11-25 11:33:04 +00:00
05aee741f6 Roll chromium_revision f9be7d3d66..d298cced6c (610432:610627)
Also fix Android build by switching to the new `errorprone_args` parameter introduced in https://chromium-review.googlesource.com/1349672

Change log: f9be7d3d66..d298cced6c
Full diff: f9be7d3d66..d298cced6c

Changed dependencies
* src/base: c501123f4c..637e844c7c
* src/build: a608842209..d6391e35ea
* src/ios: 7ec4d4bb81..37ce0c91d6
* src/testing: 64a90737d6..e78a57eb5b
* src/third_party: 9e781ebbaf..6c250624d7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4176781039..43e8ebcaf2
* src/tools: 3c063fd823..6b41cb2c41
DEPS diff: f9be7d3d66..d298cced6c/DEPS

No update to Clang.

Bug: chromium:906803
Change-Id: I45b0cf87d9aefdb4b43a6335deb389ad3e8ab9d4
Reviewed-on: https://webrtc-review.googlesource.com/c/111926
Reviewed-by: Oleksandr Iakovenko <iakovenko@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25776}
2018-11-23 17:47:35 +00:00
9289edae6f Revert "Replace the IceConnectionState implementation."
This reverts commit 1e87b4f32b73526f9caaae2a7bccfbd0cd84dcb9.

Reason for revert: Breaks internal project

Original change's description:
> Replace the IceConnectionState implementation.
> 
> PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
> Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.
> 
> Bug: webrtc:6145
> Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
> Reviewed-on: https://webrtc-review.googlesource.com/c/108780
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25773}

TBR=kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,jonasolsson@webrtc.org

Change-Id: Icc4368d120a4167286fa6ba2e884a3650b453eff
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6145
Reviewed-on: https://webrtc-review.googlesource.com/c/111925
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25775}
2018-11-23 16:19:05 +00:00
4f00075435 Remove use of CodecSpecificInfo.codec_name
Bug: webrtc:9890
Change-Id: I68bb73530f335e82d0d3f7885702fc6bb120a7a5
Reviewed-on: https://webrtc-review.googlesource.com/c/111241
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25774}
2018-11-23 16:04:13 +00:00
1e87b4f32b Replace the IceConnectionState implementation.
PeerConnection::ice_connection_state() used to return a value based on both DTLS and ICE transports.
Now that we have PeerConnection::peer_connection_state() to fill that role we can change the implementation of ice_connection_state over to match the spec.

Bug: webrtc:6145
Change-Id: Ia4f348f728f24faf4b976c63dea2187bb1f01ef0
Reviewed-on: https://webrtc-review.googlesource.com/c/108780
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25773}
2018-11-23 15:05:18 +00:00
57f3ad0f8d Adds stable bandwidth estimate to GoogCC.
The intention is to provide a bandwidth estimate that only updates if
the actual available bandwidth is known to have changed. This will be
used in media streams to avoid changing the configuration (such as
frame size, audio frame length etc), just because the control target
rate changed.

Bug: webrtc:9718
Change-Id: I17ba5a2f9e5bd408a71f89c690d45541655a68e2
Reviewed-on: https://webrtc-review.googlesource.com/c/107726
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25772}
2018-11-23 14:55:37 +00:00
88ce4ef46e Don't buffer encoded frames.
Pass encoded frames to packetizer immediately if encoder is configured
to drop whole superframe.

Bug: webrtc:9950
Change-Id: Iedee9618bb146307efd5a86cb35bf14b5e64b341
Reviewed-on: https://webrtc-review.googlesource.com/c/109002
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25771}
2018-11-23 13:48:00 +00:00
885cf60106 Moves ProbeBitrateEstimator from DelayBasedBwe.
This prepares for providing an additional implementation of delay based
rate control. By moving the probe controller, less code will have to be
added in the upcoming CL.

Bug: webrtc:9718
Change-Id: I64eb2c8f5f7950b6e9d209f110dc0a757c710b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/111860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25770}
2018-11-23 13:43:51 +00:00
e3abb8134f Decouple //rtc_base:rtc_base_tests_utils from gunit.
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.

It also removes some unused dependencies in the WebRTC build graph.

Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 12:52:46 +00:00
8af8896596 Expose jitter buffer flushes metric in new getStats api.
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}
2018-11-23 11:41:43 +00:00
b357e54dd5 Add field trial config to disable pacer emergency stops.
Bug: none
Change-Id: Ie92c4ef82e5ce3e222ec85df21acfb233b16b85d
Reviewed-on: https://webrtc-review.googlesource.com/c/111883
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25767}
2018-11-23 11:06:25 +00:00
6d254bcd5e Delete unused method NetEq::PacketBufferStatistics
Bug: None
Change-Id: I9f87e445e2b5b54f78474489172f694abff38363
Reviewed-on: https://webrtc-review.googlesource.com/c/111660
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25766}
2018-11-23 09:39:32 +00:00
5f2ffeec22 Clean up deprecated APM stats
This seems to be the last piece of the puzzle.
Deprecation PSA:
https://groups.google.com/forum/#!msg/discuss-webrtc/NgqEPvkNuDE/7HtwnMmADgAJ

Bug: webrtc:8572
Change-Id: Ib04b843fe50b8f07742c85827af6881dcfdc2991
Reviewed-on: https://webrtc-review.googlesource.com/c/109005
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25765}
2018-11-23 09:17:11 +00:00
f40150d874 Removing ANA enabling field trials.
This is to let ANA config proto to fully control it.

Bug: b/119788974
Change-Id: Ib7842f784bdf879cb7d753c7077ce845f435a379
Reviewed-on: https://webrtc-review.googlesource.com/c/111741
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25764}
2018-11-22 22:26:28 +00:00
2c977b4cc5 Remove RSID from stream configs in new event log format.
RSID is only useful if we store the RSID header extension.
Since we don't do that at the moment, there is no need to
store RSID in the stream configs.

Bug: webrtc:8111
Change-Id: I978f335d05984346f225c4781a8bfaa228f3f4c8
Reviewed-on: https://webrtc-review.googlesource.com/c/111759
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25763}
2018-11-22 17:54:06 +00:00
14dfe7f288 [GN] Fix dependency rebasing in BUILD.gn files.
This CL ensures we properly points to deps shared with chromium,
e.g. '//third_party/abseil-cpp...' and not '../third_party/abseil-cpp...'

NB: This is only applied to dependencies which were missing,
    and doesn't fix existing ones.

Bug: webrtc:10037
Change-Id: If4bbb00df39401c65def9d56e36e5feb5d67b9dd
Reviewed-on: https://webrtc-review.googlesource.com/c/111600
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25762}
2018-11-22 17:21:42 +00:00
254d869c00 Routing BitrateAllocationUpdate to audio codec.
This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.

Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}
2018-11-22 17:06:52 +00:00
3890c1ae6d Roll chromium_revision 1500c78c93..f9be7d3d66 (610314:610432)
Change log: 1500c78c93..f9be7d3d66
Full diff: 1500c78c93..f9be7d3d66

Changed dependencies
* src/base: 433ad142d5..c501123f4c
* src/build: b23a9fbf16..a608842209
* src/ios: 8c753b13c9..7ec4d4bb81
* src/third_party: 652ccc488f..9e781ebbaf
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/aa21a922d4..4176781039
* src/third_party/depot_tools: d66dad7fc2..25c4fce2ce
* src/tools: f398409248..3c063fd823
DEPS diff: 1500c78c93..f9be7d3d66/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I05aae5c560ea9824bee5b96e7a8d861219d4ba8a
Reviewed-on: https://webrtc-review.googlesource.com/c/111800
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25760}
2018-11-22 16:34:06 +00:00
3955a5022c Metal: Don't render into an empty view.
Change-Id: I4f407ab77854fa50d3b30e0bf54c365aee51923d
Bug: webrtc:10040
Reviewed-on: https://webrtc-review.googlesource.com/c/111782
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25759}
2018-11-22 16:20:37 +00:00
777cf26328 AEC3: Clockdrift detection
This change introduces a clockdrift detector operating on the estimated
delay of the echo path delay estimator. Each time the delay estimate
changes it is compared to previous estimates. If the estimates are
slowly increasing or decreasing, clockdrift is detected.

Four different patterns are considered clockdrift:
- k, k+1, k+2, k+3
- k, k+2, k+1, k+3
- k, k-1, k-2, k-3
- k, k-2, k-1, k-3

A delay estimate history matching the three last elements in one of the
patterns is considered probable clockdrift. Matching all four elements
is considered verified clockdrift.

If the delay is constant for some time after clockdrift is detected the
clockdrift detector will revert to no detected clockdrift.

The level of clockdrift is reported via an UMA histogram.

Bug: webrtc:10014
Change-Id: I1cce4d593e101a8b3fa99df6935e59b4243cb97a
Reviewed-on: https://webrtc-review.googlesource.com/c/111381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25758}
2018-11-22 16:02:44 +00:00
f259078009 Use cropping aligning in video quality analysis tool
TBR=phoglund

Bug: webrtc:9642
Change-Id: I32e54473ef6699b862b36c36c7d975b381db6ed2
Reviewed-on: https://webrtc-review.googlesource.com/c/99580
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25757}
2018-11-22 15:44:14 +00:00
ebb50c217d Fix setting max reordering threshold in ReceiveStatistics
By ensuring new max reordering threshold applies to future statisticians too.

Bug: b/38179459
Change-Id: I0df32fb893a930b93faaf2161cd03626f9544a74
Reviewed-on: https://webrtc-review.googlesource.com/c/111752
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25756}
2018-11-22 15:31:20 +00:00
286df00f72 Add tool for aligning cropped region of video files
This class adds logic for aligning what part of a test video has been
encoded from a reference video. It does that by cropping and zooming in
on a region of the reference video that most closely matches the test
video. A small cropping does not have much impact on human perception,
but it has a big impact on PSNR and SSIM calculations.

For example, if the test video is cropped with one row in the top and
bottom, adjusting for this improves average PSNR from 27.7146 to
29.3357 and average SSIM from 0.934891 to 0.95318 in an example test
video.

TBR=phoglund

Bug: webrtc:9642
Change-Id: I02cfe0e2261fb58df8cdb1e15ba93285e3dc4538
Reviewed-on: https://webrtc-review.googlesource.com/c/99480
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25755}
2018-11-22 15:30:15 +00:00
8e668633c7 Remove cricket::UdpTransport.
This code is never built by GN, and the header is never included.

Bug: webrtc:9855
Change-Id: I7f79c2b16e4a833fa7788be87dbdf9b41247c9e4
Reviewed-on: https://webrtc-review.googlesource.com/c/111755
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25754}
2018-11-22 14:05:38 +00:00
94c94205f7 Remove cricket::BundleFilter.
This code is never built by GN, and the header is never included.

Bug: webrtc:9855
Change-Id: I16e6a54cc95629917d454f91d9bcc99fc55d8a00
Reviewed-on: https://webrtc-review.googlesource.com/c/111754
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25753}
2018-11-22 14:04:33 +00:00
eccfc47ffa Cleanup AimdRateController and remove RateControlRegion enum.
- Rename avg_max_bitrate_kbps to link_capacity_estimate_kbps and change
  the type to optional.
- Remove the RateControlRegion enum. The old code seems to have the invariant
  that the region is kRcMaxUnknown iff avg_max_bitrate_kbps is uninitialized.
- Change floats to double.

Bug: webrtc:9942
Change-Id: Ic071a11ec4950053ec92beaa06f28f43192521d7
Reviewed-on: https://webrtc-review.googlesource.com/c/111247
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25752}
2018-11-22 13:51:28 +00:00
42d2e4bbb1 Increase test timeouts in TCPChannelClientTest
This test fails on slow runners when TCPChannelClient has not
yet finished communication, but the test thread times out checking
that mock methods are called.

Bug: webrtc:9955
Change-Id: Ia91ada6b01ca1bab48afa57fe76aedd08770a641
Reviewed-on: https://webrtc-review.googlesource.com/c/111383
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25751}
2018-11-22 13:16:18 +00:00
00dfe932a7 Remove superfluous constructor from dltsTransport
Addressing TODO left in the code.

Bug: none
Change-Id: If7ea70c727f6b7f6496cdb0f6d81fb53dd23ef0a
Reviewed-on: https://webrtc-review.googlesource.com/c/111748
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25750}
2018-11-22 11:46:30 +00:00
44727b48d6 Cleanup rtcp StreamStatistician::OnRtpPacket
inline InOrder check
remove it from IsRetransmit check as redundant
avoid call to IsRetransmitOfOldPacket when packet arrived in order
take current time once
Remove packet overhead counting as unused

Bug: None
Change-Id: Icd8bf69b5076e4469c349529c9ac79a1b15d9515
Reviewed-on: https://webrtc-review.googlesource.com/c/111746
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25749}
2018-11-22 11:42:13 +00:00
af228ee761 Disable flaky tests CallPerfTest.CaptureNtpTimeWithNetworkDelay on WIN.
See linked bug.

TBR=stefan@webrtc.org

Bug: webrtc:8291
Change-Id: I0e5896a6e5bbb6979d59032d1a033f209d45918e
Reviewed-on: https://webrtc-review.googlesource.com/c/111749
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25748}
2018-11-22 11:23:37 +00:00
5486bcd0d0 Remove SetChannelParameters function from API classes.
Followup to https://webrtc-review.googlesource.com/c/src/+/108861

Bug: webrtc:9946
Change-Id: Ia6e7fa3942c21aefeadb7b214c85cff93fbc2ef6
Reviewed-on: https://webrtc-review.googlesource.com/c/109860
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25747}
2018-11-22 11:12:10 +00:00
ecd62056e3 Disable GoogCcNetworkControllerTest.DetectsHighRateInSafeResetTrial
Test is flaky. See linked bug.

TBR=srte@webrtc.org

Bug: webrtc:10036
Change-Id: I21dd0daceaca6071364cb3aec50da79480f4dfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/111747
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25746}
2018-11-22 10:51:11 +00:00
8ac05ccaa7 Adds trial to use link capacity estimate in Opus encoder.
Since the link capacity is designed to be a more stable value, we don't
need the smoothing. This allows us to react faster to changes in link
capacity while still avoiding to react to changes in target bitrate due
to normal control behavior.

Bug: webrtc:9718
Change-Id: I2fbf6bb882f312a7b28ea43d27057886d035ac45
Reviewed-on: https://webrtc-review.googlesource.com/c/111511
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25745}
2018-11-22 09:21:12 +00:00
2ff3f49700 Move webrtc::CreatePeerConnectionFactory definition next to decl.
This CL moves webrtc::CreatePeerConnectionFactory definitions out of
pc:create_pc_factory and merges it with its declaration in the api/
directory.

In order to avoid circular dependencies a new build target is created:
* api:create_peerconnection_factory

Bug: webrtc:9862
Change-Id: Ie215c94460cba026f5bf7d11c9a5aa03792064af
Reviewed-on: https://webrtc-review.googlesource.com/c/111186
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25744}
2018-11-22 09:07:51 +00:00
d51b3553db Delete unused NetEq Rtcp stats.
Bug: webrtc:7135
Change-Id: Ib3ca9e02b051b8b41c2eac4e43a4f1f37999bf75
Reviewed-on: https://webrtc-review.googlesource.com/c/111640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25743}
2018-11-22 08:00:54 +00:00
7c36c71326 Roll chromium_revision 6931f4c0d0..1500c78c93 (610209:610314)
Change log: 6931f4c0d0..1500c78c93
Full diff: 6931f4c0d0..1500c78c93

Changed dependencies
* src/base: ab495eac8b..433ad142d5
* src/build: dcab93e47d..b23a9fbf16
* src/buildtools: da9b2941cb..04161ec8d7
* src/ios: 205368975a..8c753b13c9
* src/testing: 6a892ba0d8..64a90737d6
* src/third_party: 2c4678ac79..652ccc488f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d69ae20edf..aa21a922d4
* src/third_party/depot_tools: a85a4b01ee..d66dad7fc2
* src/tools: bcd323f441..f398409248
DEPS diff: 6931f4c0d0..1500c78c93/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I1b63aae3051a2eab1dd25c545a5468992291ef14
Reviewed-on: https://webrtc-review.googlesource.com/c/111734
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25742}
2018-11-22 06:31:21 +00:00