Commit Graph

25551 Commits

Author SHA1 Message Date
7f7e973362 Roll chromium_revision 82a8b043ef..28d6168850 (611537:611644)
Change log: 82a8b043ef..28d6168850
Full diff: 82a8b043ef..28d6168850

Changed dependencies
* src/base: 0dd82ab83e..ef037f58bf
* src/ios: 6ee737e3ed..2b4e7b8003
* src/third_party: ec8029cd49..7d90e20fb8
* src/third_party/robolectric: version:3.5.1..Q-aurrjcIq02dl2ws-M-IhnIydurvTbUMR_xD_zBJ0YC
* src/tools: 5c6de96455..8bab01cf90
DEPS diff: 82a8b043ef..28d6168850/DEPS

No update to Clang.

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

Change-Id: I9730282c84c902ec93b09ea986fe4a6f6dcd0ac8
Reviewed-on: https://webrtc-review.googlesource.com/c/112260
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@{#25819}
2018-11-28 11:32:54 +00:00
de10eea6fc AEC3: Fix ENR in the dominant nearend detection
Correcting a mistake in the dominant nearend detection where
the meaning of the echo-to-nearend ratio was inversed.

Bug: webrtc:8671
Change-Id: I7f56369fad1784e256150c312b6b3dafcb9d0f71
Reviewed-on: https://webrtc-review.googlesource.com/c/112136
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25818}
2018-11-28 09:23:34 +00:00
cf69d2209b AEC3: Optimizing the Update method of the FilterAnalyzer class.
In this CL the analysis of the impulse response that is done in the FilterAnalyzed class is changed in order to reduce its complexity. Instead of analyzing the whole impulse response in each Update call a smaller region is analyzed. That region is changed at each Update call which implies that several calls are needed in order to analyze the complete impulse response.

Bug: webrtc:10032,chromium:909007
Change-Id: Ic58be34ba18485311c63e0fed9b6e892f9cb864c
Reviewed-on: https://webrtc-review.googlesource.com/c/111602
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25817}
2018-11-28 09:01:07 +00:00
154a262b61 Don't clear self.videoFrame when setting up OpenGL in the EAGL video view.
It makes sense to clean up self.videoFrame in -teardownGL, but if
we happen to have a frame available in -setupGL then it's OK to
keep using that frame. Clearing the frame here frequently causes
the screen view to go black for a moment when the app returns from
the background.

Bug: webrtc:10059
Change-Id: Ic62f872a0a582c807cee1e30ea9bb32f31ada341
Reviewed-on: https://webrtc-review.googlesource.com/c/112213
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25816}
2018-11-28 09:00:06 +00:00
2ba65c3a71 Fix webrtc-internal ios json config
Bug: webrtc:10047
Change-Id: I3a5f1e1731872a712a255a71f9039ea55e347274
Reviewed-on: https://webrtc-review.googlesource.com/c/112135
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25815}
2018-11-28 08:53:26 +00:00
ce1b140b8d Adding WinUWP compilation support to WebRTC.
Windows UWP allows an application to be built that targets
across all Windows 10 based systems and the Windows store.

Change-Id: I69694bb7e83fb01ad6db2438b065b55738cf01fd
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/110570
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25814}
2018-11-28 08:32:30 +00:00
3bc24bcd3e Roll chromium_revision b04e513f82..82a8b043ef (611432:611537)
Change log: b04e513f82..82a8b043ef
Full diff: b04e513f82..82a8b043ef

Changed dependencies
* src/base: b9901f9bc5..0dd82ab83e
* src/ios: eb9e8e09dc..6ee737e3ed
* src/third_party: 77f6c16720..ec8029cd49
* src/third_party/nasm: a0a6951e25..4ee6a69ce3
* src/tools: b47c631587..5c6de96455
DEPS diff: b04e513f82..82a8b043ef/DEPS

No update to Clang.

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

Change-Id: I87d788e4455b1a1ddb0d21d9a3327a0658d4f3e4
Reviewed-on: https://webrtc-review.googlesource.com/c/112224
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@{#25813}
2018-11-28 03:36:36 +00:00
5ec2c94b0b Reland "Delay call to Destroy until after SignalDone has finished firing."
This is a reland of c7b8da453c2776a3e80fca3ff2400831adad03af

Original change's description:
> Delay call to Destroy until after SignalDone has finished firing.
> 
> Bug: chromium:905542
> Change-Id: I0def33a3ac21bd0da4b77ef05d42a0e7ef66caec
> Reviewed-on: https://webrtc-review.googlesource.com/c/112163
> Commit-Queue: Zach Stein <zstein@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25807}

Bug: chromium:905542
Change-Id: I5dd861bcd3b2ebcb7ff3db0888da1520eaaaca72
Reviewed-on: https://webrtc-review.googlesource.com/c/112240
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Zach Stein <zstein@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25812}
2018-11-28 02:55:11 +00:00
0d007d7c4f Roll chromium_revision aed902039c..b04e513f82 (611312:611432)
Change log: aed902039c..b04e513f82
Full diff: aed902039c..b04e513f82

Changed dependencies
* src/base: 577be0657e..b9901f9bc5
* src/build: e1cd03d1a2..bbd67a350d
* src/ios: cd3f4cdae4..eb9e8e09dc
* src/testing: 1070bcfcd9..5a776bca05
* src/third_party: 23cd046553..77f6c16720
* src/tools: fb15cfbb94..b47c631587
DEPS diff: aed902039c..b04e513f82/DEPS

No update to Clang.

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

Change-Id: I19e9fbd096f8da2d144b74ce8bbb17db224b0bf6
Reviewed-on: https://webrtc-review.googlesource.com/c/112218
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@{#25811}
2018-11-28 01:13:10 +00:00
72d2ddd36f Fix raddr on srflx and relay candidates
Bug: chromium:905690
Change-Id: Ic16d21672db5d456d7a9727ea5194ec26338c9d0
Reviewed-on: https://webrtc-review.googlesource.com/c/111441
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Justin Uberti <juberti@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25810}
2018-11-28 00:13:23 +00:00
889b587226 Revert "Delay call to Destroy until after SignalDone has finished firing."
This reverts commit c7b8da453c2776a3e80fca3ff2400831adad03af.

Reason for revert: introduced a flaky test

Original change's description:
> Delay call to Destroy until after SignalDone has finished firing.
> 
> Bug: chromium:905542
> Change-Id: I0def33a3ac21bd0da4b77ef05d42a0e7ef66caec
> Reviewed-on: https://webrtc-review.googlesource.com/c/112163
> Commit-Queue: Zach Stein <zstein@webrtc.org>
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25807}

TBR=zstein@webrtc.org,qingsi@webrtc.org

Change-Id: Iae31fdfe16823f1061b433037eb1e498d8b33f28
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:905542
Reviewed-on: https://webrtc-review.googlesource.com/c/112217
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25809}
2018-11-27 23:43:26 +00:00
ab20b3f364 Roll chromium_revision 81c26a093b..aed902039c (611047:611312)
Change log: 81c26a093b..aed902039c
Full diff: 81c26a093b..aed902039c

Changed dependencies
* src/base: 3c9ac70552..577be0657e
* src/build: 076d347a56..e1cd03d1a2
* src/ios: aa06ff9dc9..cd3f4cdae4
* src/testing: eeb483a834..1070bcfcd9
* src/third_party: 3a20b86d38..23cd046553
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/f241a59dcc..6965d25602
* src/third_party/depot_tools: 6c18a1afa1..44d4b29082
* src/tools: 627b8cb5af..fb15cfbb94
* src/tools/swarming_client: b6e9e23e4e..157bec8a25
DEPS diff: 81c26a093b..aed902039c/DEPS

No update to Clang.

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

Change-Id: I2ae5dbcc633fbe1cccbd0bc28011ccbbdcb2828f
Reviewed-on: https://webrtc-review.googlesource.com/c/112215
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@{#25808}
2018-11-27 22:58:35 +00:00
c7b8da453c Delay call to Destroy until after SignalDone has finished firing.
Bug: chromium:905542
Change-Id: I0def33a3ac21bd0da4b77ef05d42a0e7ef66caec
Reviewed-on: https://webrtc-review.googlesource.com/c/112163
Commit-Queue: Zach Stein <zstein@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25807}
2018-11-27 19:55:48 +00:00
14f252a1e4 AEC3: Add metrics for API call jitter
Bug: webrtc:10021,chromium:907234
Change-Id: Ic0e6ba01c8dfdd5ca8230c8579bf149693e5f151
Reviewed-on: https://webrtc-review.googlesource.com/c/111580
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25806}
2018-11-27 19:52:08 +00:00
10403ae87c Add PeerConnection option to configure minimum audio jitter buffer delay.
Note that this value will override the minimum delay that is used for audio/video sync.

Bug: webrtc:10053
Change-Id: Ia129f6c9ee9da5d00a3d955afaaa6e8f0c2bee33
Reviewed-on: https://webrtc-review.googlesource.com/c/112121
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25805}
2018-11-27 19:49:48 +00:00
c7f1a0af92 Fix webrtc-internal configs to run perf tests on separate bots
Test execution was broken by specifying pool.
No need to do it once we specify bot ids in included json configs.

Bug: webrtc:10047
Change-Id: Ica5b891b796eec69573cc39d1d72617a68169499
Reviewed-on: https://webrtc-review.googlesource.com/c/112129
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25804}
2018-11-27 16:36:30 +00:00
8b5560218a Batch RTC event log output if using the new wire format.
The new wire format doesn't have much effect on compression unless
the log is encoded in reasonably large batches.

PeerConnection has two functions to start logging; one which takes
an output period (or batch size) in milliseconds and one which uses
a default period instead. This CL changes the default batch size to
5 seconds if the the new format is enabled as a field trial.

Bug: webrtc:8111
Change-Id: I638f6114325251b6a9acf4f863afe2688a3b0522
Reviewed-on: https://webrtc-review.googlesource.com/c/112130
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25803}
2018-11-27 16:11:59 +00:00
352ce5c419 Expose delayed packet outage as a cumulative metric of samples in the new getStats API.
The stat will be exposed through origin trial described in:
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI

Change-Id: Ib191a11c6bd9e617abbe9dd82239b0c5b4e6b4e0
Bug: webrtc:10043
Reviewed-on: https://webrtc-review.googlesource.com/c/111922
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25802}
2018-11-27 15:10:09 +00:00
59cfd35438 Address vptr race condition while PeerConnection is destructed.
Auxiliary threads (worker, network) are still active
while PeerConnection is destructed, leading to race condition
in tests such as:
  * RTCStatsIntegrationTest.GetStatsFromCaller
  * RTCStatsIntegrationTest.GetsStatsWhileDestroyingPeerConnection

This CL prevents the conflict to happen by explicitly
closing the PeerConnection.

Bug: webrtc:9847
Change-Id: I40880bb9b193201711031b8c4563c6bbd4983c71
Reviewed-on: https://webrtc-review.googlesource.com/c/104820
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25801}
2018-11-27 14:06:37 +00:00
09d6588d93 Change HdrMetadataExtension to ColorSpaceExtension
Bug: webrtc:8651
Change-Id: Ica6f8c6bd13bb07f89700b9c0a359b9a58feefbb
Reviewed-on: https://webrtc-review.googlesource.com/c/111758
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25800}
2018-11-27 14:05:31 +00:00
53382cb19f Move RtcpStatistics from common_types.h to a new header file
New location is modules/rtp_rtcp/include/rtcp_statistics.h.

Bug: webrtc:5876
Change-Id: I85f55b58658588228ed77175226b3479352fd5de
Reviewed-on: https://webrtc-review.googlesource.com/c/111961
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25799}
2018-11-27 13:46:42 +00:00
6b3d18164b Remove unused BWE field trial strings.
Bug: None
Change-Id: I38d2e5495ddfe0b9f1493efc38ef7df95e7fd207
Reviewed-on: https://webrtc-review.googlesource.com/c/111258
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25798}
2018-11-27 13:05:43 +00:00
65c921cb4f Add setters to ColorSpace class
Color space information will be transmitted as uint8_t. It's therefore
necessary to convert from uint8_t to the corresponding enums.

Bug: webrtc:8651
Change-Id: Ib7e7f9f6b4d7e0c291d283822180144944f3ea1e
Reviewed-on: https://webrtc-review.googlesource.com/c/111757
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25797}
2018-11-27 13:00:50 +00:00
196c5ba045 Specific pacing configuration.
Allow configuration of pacing parameters, independently of ALR configuration.

Bug: webrtc:10038
Change-Id: I2a7f650d1e6d58f9936fc481988bc4bb6fba7220
Reviewed-on: https://webrtc-review.googlesource.com/c/112127
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25796}
2018-11-27 12:36:40 +00:00
ba2840ce4e Various VP9 high fps fixes
- Enable flexible mode in loopback tools and quality tests
- Ensure duplicate references are not set by the sender in video header
- Reset first active spatial layer on keyframe in encoder
- Make vp9 encoder to not generate spatial references for first active
  layer with external reference control in svc flexible mode

Bug: webrtc:10049
Change-Id: If9ff576ea8a1a2fef6116b17b5b5adff08c5f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/112080
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25795}
2018-11-27 12:20:56 +00:00
af52b68116 Populate VideoSendTime extension network2 field when configured
before this CL it was only configured when pacer is used.
This CL sets it also when pacer is not used.

Move block for setting TransmissionOffset/AbsoluteTime extensions after pacer_ check
to stress in pacer case there are set(overwritten) in another function.

Bug: None
Change-Id: I06a6dd6ec689a25439a75b3baa71340535cd1ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/112126
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25794}
2018-11-27 10:54:40 +00:00
31a43319c8 Roll chromium_revision 208bb982f7..81c26a093b (610939:611047)
Change log: 208bb982f7..81c26a093b
Full diff: 208bb982f7..81c26a093b

Changed dependencies
* src/base: ec47506add..3c9ac70552
* src/build: 01d83ecc66..076d347a56
* src/ios: f3ae5bb001..aa06ff9dc9
* src/testing: f69111fe10..eeb483a834
* src/third_party: cddbcf9ebb..3a20b86d38
* src/tools: 6d09354f13..627b8cb5af
DEPS diff: 208bb982f7..81c26a093b/DEPS

No update to Clang.

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

Change-Id: I994001245fca4b194d8d4799b80d1099e04c186f
Reviewed-on: https://webrtc-review.googlesource.com/c/112204
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@{#25793}
2018-11-27 09:32:26 +00:00
c5095e52ed Fix wrong forward declaration namespace.
Bug: webrtc:9862
Change-Id: I51b5f189a83c44ad34f85bac8832e462276afcf3
Reviewed-on: https://webrtc-review.googlesource.com/c/111924
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25792}
2018-11-27 08:20:05 +00:00
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