Commit Graph

20146 Commits

Author SHA1 Message Date
aba85d1f53 Resolve circular dependency in rtc_media_base.
This one was pretty straightforward fortunately.

Bug: webrtc:6828
Change-Id: Ie7b5e71f1298c409dbca2c74eaa09c0986e41d8f
Reviewed-on: https://webrtc-review.googlesource.com/25821
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20914}
2017-11-28 15:28:58 +00:00
b874a35227 Complete moving i420 out from video_frame_api.
Bug: webrtc:7504
Change-Id: I2cbcc91bd6be4d55c0d78cf06c69fb8db2d35e65
Reviewed-on: https://webrtc-review.googlesource.com/22640
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20913}
2017-11-28 14:01:28 +00:00
3fed5dbed6 Reduce complexity of fake slide generator
The random square generator produces unrealistically complex frames in
some situations, leading to frames > 250kb even at max QP. This leads to
unmanageably long transmission delays.

Bug: None
Change-Id: I8f5a33d52fb5efa03de97e529ad598b75511f679
Reviewed-on: https://webrtc-review.googlesource.com/23561
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20912}
2017-11-28 13:59:09 +00:00
979d6f96a8 in RtcpTransceiver tests use dedicate RtcpParserTransport
class to pass packet to RtcpPacketParser

This helpers make tests setup cleaner and
makes explicit expectation on number of packets passed to the transport.

Bug: webrtc:8239
Change-Id: I2d5975be59327cee440e87dbd0701b93514c9726
Reviewed-on: https://webrtc-review.googlesource.com/22460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20911}
2017-11-28 13:52:08 +00:00
4fa5da54d5 Android: Generate JNI code for MediaStreamTrack
Bug: webrtc:8278
Change-Id: Id5ac6ecd4f65bed4ae4b2953ef58ebc390508d21
Reviewed-on: https://webrtc-review.googlesource.com/25963
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20910}
2017-11-28 13:14:08 +00:00
32f64d2ef9 rtp_encode: Fixing bug related to DTX
Bug: webrtc:2692
Change-Id: I7b884b22cab21b9dce77e5599f43431bbc899f5d
Reviewed-on: https://webrtc-review.googlesource.com/26027
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20909}
2017-11-28 12:35:38 +00:00
eb5554c8d2 Add configs iOS11 bots and trybots.
No-Try: true
Bug: webrtc:8570
Change-Id: I1de6b12bc9797577379b0ed863b7cc3e3703d4f3
Reviewed-on: https://webrtc-review.googlesource.com/26032
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20908}
2017-11-28 11:59:08 +00:00
327c43c29b Add sending Nack to RtcpTransceiver
Bug: webrtc:8239
Change-Id: Idf27bb05958d9eceaf601078019f05444232581f
Reviewed-on: https://webrtc-review.googlesource.com/26260
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20907}
2017-11-28 11:57:58 +00:00
2f061681cc Make PrintResultList receive a vector of doubles instead of a string.
Also, add more tests to perf_test_unittest.

Bug: webrtc:8566
Change-Id: I8864db7172fa207803d310c4a5fee4bf820a56bd
Reviewed-on: https://webrtc-review.googlesource.com/25823
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20906}
2017-11-28 11:52:38 +00:00
e9619f8f81 Add a new NetEq decoding unit test for Opus with DTX
This tests NetEq with a stream encoded with Opus using it's internal
DTX/CNG.

Also adding a new resource file which is encoded using Opus with DTX.

Bug: webrtc:8488
Change-Id: Icfba5bc5dc7f9c9d0e637a90f4df674e8ba40358
Reviewed-on: https://webrtc-review.googlesource.com/26028
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20905}
2017-11-28 10:45:38 +00:00
2492984441 Add TimeMicrosToNtp to calculate current NtpTime without Clock
Bug: webrtc:6733, webrtc:8239
Change-Id: I8ac4464cd7a7ec2b2dbad44430f1141a80ba39c1
Reviewed-on: https://webrtc-review.googlesource.com/25541
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20904}
2017-11-28 10:11:58 +00:00
fb09eeb8f1 Attempt to resolve crash in AudioDeviceIOS::UpdateAudioDeviceBuffer
Bug: b/69547732
Change-Id: I078175f96d55351ab0318aa2de96f4b859e752ea
Reviewed-on: https://webrtc-review.googlesource.com/24864
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20903}
2017-11-28 09:13:18 +00:00
8f00ad02fb WindowFinderTest.FindConsoleWindow is flaky on Win32 ASan
The root cause of the flakiness is unknown, the possible issue is that the
console window running the test case is hidden or minimized. So this change
adds a SetWindow(SW_MAXIMIZE) to ensure the console window is showing.

I have run the tests against win_asan for hundreds times during the
thanksgiving. So far, no flakiness were caught.

Bug: webrtc:8568
Change-Id: Ib2c93e9bd511257213254bdaa0079c14ea50f3e4
Reviewed-on: https://webrtc-review.googlesource.com/25286
Reviewed-by: Zijie He <zijiehe@chromium.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#20902}
2017-11-28 01:21:47 +00:00
99206df620 [desktopCapture] make FakeDesktopCapturer reachable by Chromium
To make desktopCapture autotest in chromium more meaningful, it's better
to creake fake capturer of each capture type. Here we can reuse the
FakeDesktopCapturer for window capture in chromium.

Bug: chromium:699201
Change-Id: Icbe134d99cbd4980bf27fe74c1c629a1469836ea
Reviewed-on: https://webrtc-review.googlesource.com/26360
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20901}
2017-11-28 00:55:07 +00:00
73da79cf66 Step 1 to remove MediaContentDirection
This change adds |transceiver_direction()| and
|set_transceiver_direction()| to MediaContentDescription so that
external users can switch off of MediaContentDirection.

This deprecates the use of |direction()| and |set_direction()|
for external users. Once everyone has moved off of those methods,
the signiture will change to return/set RtpTransceiverDirection.
Then external users can move back to these methods with the new
signature.

Bug: webrtc:8558
Change-Id: I7e3ba289d3a0ac738b364b0388621cc3e7bcf5d3
Reviewed-on: https://webrtc-review.googlesource.com/24743
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20900}
2017-11-28 00:15:17 +00:00
1d03a751b0 Remove cricket::RtpTransceiverDirection
Replaces cricket::RtpTransceiverDirection with
webrtc::RtpTransceiverDirection, which is part of the public API.

Bug: webrtc:8558
Change-Id: Ibfc9373e25187e98fb969e7ac937a1371c8fa4c7
Reviewed-on: https://webrtc-review.googlesource.com/24129
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20899}
2017-11-27 23:04:17 +00:00
9158ef6575 Reland "Add AddTransceiver and GetTransceivers to PeerConnection"
This reverts commit 8b13f96e2d4b0449e54a3665121a4302ceb56e80.

Original change's description:
> Revert "Add AddTransceiver and GetTransceivers to PeerConnection"
> 
> This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a.
> 
> Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
> 
> Original change's description:
> > Add AddTransceiver and GetTransceivers to PeerConnection
> > 
> > WebRTC 1.0 has added the transceiver API to PeerConnection. This
> > is the first step towards exposing this to WebRTC consumers. For
> > now, transceivers can be added and fetched but there is not yet
> > support for creating offers/answers or setting local/remote
> > descriptions. That support ("Unified Plan") will be added in
> > follow-up CLs.
> > 
> > The transceiver API is currently only available if the application
> > opts in by specifying the kUnifiedPlan SDP semantics when creating
> > the PeerConnection.
> > 
> > Bug: webrtc:7600
> > Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> > Reviewed-on: https://webrtc-review.googlesource.com/23880
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#20896}
> 
> TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org
> 
> Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:7600
> Reviewed-on: https://webrtc-review.googlesource.com/26400
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20897}

TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org

Change-Id: I19fdf08c54f09302794e998a0ffddb82ae0d7b41
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26401
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20898}
2017-11-27 22:27:49 +00:00
8b13f96e2d Revert "Add AddTransceiver and GetTransceivers to PeerConnection"
This reverts commit f93d2800d9b0d5818a5a383def0aaef3d441df3a.

Reason for revert: https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2Fios-device%2F5804%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout

Original change's description:
> Add AddTransceiver and GetTransceivers to PeerConnection
> 
> WebRTC 1.0 has added the transceiver API to PeerConnection. This
> is the first step towards exposing this to WebRTC consumers. For
> now, transceivers can be added and fetched but there is not yet
> support for creating offers/answers or setting local/remote
> descriptions. That support ("Unified Plan") will be added in
> follow-up CLs.
> 
> The transceiver API is currently only available if the application
> opts in by specifying the kUnifiedPlan SDP semantics when creating
> the PeerConnection.
> 
> Bug: webrtc:7600
> Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
> Reviewed-on: https://webrtc-review.googlesource.com/23880
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20896}

TBR=steveanton@webrtc.org,zhihuang@webrtc.org,hbos@webrtc.org,pthatcher@webrtc.org

Change-Id: Ie91ea4988dba25c20e2532114d3a9d859a932d4c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7600
Reviewed-on: https://webrtc-review.googlesource.com/26400
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20897}
2017-11-27 20:56:00 +00:00
f93d2800d9 Add AddTransceiver and GetTransceivers to PeerConnection
WebRTC 1.0 has added the transceiver API to PeerConnection. This
is the first step towards exposing this to WebRTC consumers. For
now, transceivers can be added and fetched but there is not yet
support for creating offers/answers or setting local/remote
descriptions. That support ("Unified Plan") will be added in
follow-up CLs.

The transceiver API is currently only available if the application
opts in by specifying the kUnifiedPlan SDP semantics when creating
the PeerConnection.

Bug: webrtc:7600
Change-Id: I0b8ee24b489b45bb4c5f60b699bd20c61af01d8e
Reviewed-on: https://webrtc-review.googlesource.com/23880
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20896}
2017-11-27 20:35:35 +00:00
92016ce5a4 Optional: Use nullopt and implicit construction in /common_video
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=magjed@webrtc.org

Bug: None
Change-Id: I0eddc997560894dc661f521f6096e2d834216cee
Reviewed-on: https://webrtc-review.googlesource.com/23608
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20895}
2017-11-27 17:03:12 +00:00
9299642fd0 Make ALR probing experiment default on.
Bug: webrtc:7694
Change-Id: I9d468ed13d2894c6d6ec9163d21959d51926cf33
Reviewed-on: https://webrtc-review.googlesource.com/23560
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20894}
2017-11-27 15:17:12 +00:00
ce15cd3e3e Disable event perf tests
TBR=tommi@webrtc.org

Bug: webrtc:8546
Change-Id: Ifad216f07d069f9d99f4bc541da445d564e9028d
Reviewed-on: https://webrtc-review.googlesource.com/26030
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20893}
2017-11-27 13:56:22 +00:00
83c4a02b76 Added metric for the delay in AEC3.
Bug: webrtc:8569
Change-Id: I659049a411654bd3a252ab29008fac467f903efd
Reviewed-on: https://webrtc-review.googlesource.com/25600
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20892}
2017-11-27 12:52:42 +00:00
6a0345b3b0 Android: Generate JNI code for MediaStream
Bug: webrtc:8278
Change-Id: I48d0615f3db3f22e7179a2d7c59b970a33678ada
Reviewed-on: https://webrtc-review.googlesource.com/25962
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20891}
2017-11-27 11:58:32 +00:00
b8ff8f7d40 Revert "Adding -Wno-deprecated-declarations to declarations deprecated in iOS 10"
This reverts commit 2a9dbe6e7e722cf069b72ccff0051b1517706996.

Reason for revert: The fix does not help, it still fails at runtime

Original change's description:
> Adding -Wno-deprecated-declarations to declarations deprecated in iOS 10
>
> This CL will unblock the Chromium Roll while deprecated declarations
> will be removed from the WebRTC codebase.
>
> Bug: webrtc:8570
> Change-Id: I55cf78040758369ce45176cf0a00df50a87eb972
> No-Try: True
> Reviewed-on: https://webrtc-review.googlesource.com/25641
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20858}

No-Try: True
Bug: webrtc:8570
Change-Id: Ib17ed05f8f239912661281329ecab175c23491d6
Reviewed-on: https://webrtc-review.googlesource.com/25964
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20890}
2017-11-27 11:54:52 +00:00
06483ca0ce Add support for the rtc::Event implementation in Chromium.
See also:
https://chromium-review.googlesource.com/c/chromium/src/+/789871

BUG=chromium:689520

Change-Id: I2a22d7aa4d5b91ad3481c64d15419c81ae1dd768
Reviewed-on: https://webrtc-review.googlesource.com/26021
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20889}
2017-11-27 11:50:12 +00:00
80610c4fd1 Android: Generate JNI code for IceCandidate
Bug: webrtc:8278
Change-Id: I4facd1f6babd6e8a9b35c86b6ad7420e52321f49
Reviewed-on: https://webrtc-review.googlesource.com/25960
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20888}
2017-11-27 11:17:02 +00:00
5b8c0a2a1e Optional: Use nullopt and implicit construction in /modules/congestion_controller
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: If2a98dc714d1755f07af1f70248cf41e4a9db750
Reviewed-on: https://webrtc-review.googlesource.com/23612
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20887}
2017-11-27 10:42:22 +00:00
d0e196bd26 Adding two tests:
1. Bitrate allocation strategy unit test 
2. Perf test determining minimal supported bitrate with and without strategy

Bug: webrtc:8243
Change-Id: Idf675fbadddb66c77b2582052d6497971eb99ad6
Reviewed-on: https://webrtc-review.googlesource.com/4880
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20886}
2017-11-27 10:38:22 +00:00
07109657a5 Old SetRemoteDescription signature default implementation.
So that third party projects don't still have to implement it when they
switch over to the new signature.

Bug: webrtc:8473
Change-Id: I329814ad6e899def7bad97939e8643380a268f91
Reviewed-on: https://webrtc-review.googlesource.com/26022
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20885}
2017-11-27 10:25:42 +00:00
9c78058f50 Optional: Use nullopt and implicit construction in /rtc_base
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: I2c1e50090e48c892fda74f1678231a7624cac9fa
Reviewed-on: https://webrtc-review.googlesource.com/23575
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20884}
2017-11-27 10:17:52 +00:00
80adac053c Set video codec names in public API using existing cricket constants
For better consistency between the Objective-C API constant definitions
and the existing constants defined in the underlying core, re-use the
available video codec-name constants from cricket to define the peer
constants in the public API.

BUG=None

Change-Id: I8d5ddc2c1bd6670810fca1665aaf9a116620a34e
Reviewed-on: https://webrtc-review.googlesource.com/25360
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20883}
2017-11-27 09:01:42 +00:00
09819ec373 Improve the rtc::Buffer docs
Bug: none
Notry: true
Change-Id: Ie18d2c30617a168eb7cf8b1497e924eeb5083892
Reviewed-on: https://webrtc-review.googlesource.com/25822
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20882}
2017-11-25 19:03:41 +00:00
a319931f3c Generate JNI code for built in Integer class
TBR=sakal@webrtc.org

Bug: webrtc:8279
Change-Id: Ic2e5d169f11a0d4a88086d61f737f33ab9e86447
Reviewed-on: https://webrtc-review.googlesource.com/24220
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20881}
2017-11-25 12:16:20 +00:00
18b823e0f5 Revert "Let PrintResultMeanAndError accept a string."
This reverts commit b54bc06079a66a379595068f6ea56d56a4fd95c0.

Reason for revert: Downstream projects should be fixed now

Original change's description:
> Let PrintResultMeanAndError accept a string.
> 
> Some downstream projects still use it. I'll update them and then revert
> this change.
> 
> Bug: webrtc:8566
> Change-Id: Ib4e56348c40a3645f3049382b47089ca6c675e96
> Reviewed-on: https://webrtc-review.googlesource.com/25841
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20872}

TBR=phoglund@webrtc.org,ehmaldonado@webrtc.org

Change-Id: I5d2a59cbfb6a148fc6c621a69fa23397ba2c6991
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8566
Reviewed-on: https://webrtc-review.googlesource.com/25920
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20880}
2017-11-24 18:39:44 +00:00
c4c8876f70 Android: Generate JNI code for RTCStats
Bug: webrtc:8278
Change-Id: I183cec54ec3e97894db7f26e365eb9941a1ab458
Reviewed-on: https://webrtc-review.googlesource.com/25660
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20879}
2017-11-24 18:38:40 +00:00
6bd39025ec Optional: Use nullopt and implicit construction in /modules/video_coding
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

Bug: None
Change-Id: Iedebf4dc56a973306e7d7e7649525879808dc72b
Reviewed-on: https://webrtc-review.googlesource.com/23578
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20878}
2017-11-24 18:36:09 +00:00
56d460902e Use the new AudioProcessing statistics everywhere.
The new interface uses optionals instead of default values, and only values that are actually used are included. To make it easy to add/remove stats in the future, the struct itself is copied around on all layers, instead of copying the values one by one. This CL also fixes a bug which caused several APM statistics to get stuck at a fixed level when there are no more receive streams (after some period where there were receive streams). Since APM doesn't know this happens, an argument was added to the GetStats call to pass this information down to APM.

Bug: webrtc:8563, b/67926135
Change-Id: I96cc008353355bb520c4523f5c5379860f73ee24
Reviewed-on: https://webrtc-review.googlesource.com/25621
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20877}
2017-11-24 18:17:39 +00:00
606c882972 Optional: Use nullopt and implicit construction in /voice_engine
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=henrikg@webrtc.org

Bug: None
Change-Id: I8cb71d015813312ccfc1b6fc7a2a27f391aaf4d1
Reviewed-on: https://webrtc-review.googlesource.com/23614
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20876}
2017-11-24 17:09:19 +00:00
3928dbc1e7 Optional: Use nullopt and implicit construction in /rtc_tools/event_log_visualizer
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=terelius@webrtc.org

Bug: None
Change-Id: I18b44f6c1bd3ccb4288807eee883b87afa7900f4
Reviewed-on: https://webrtc-review.googlesource.com/23569
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20875}
2017-11-24 16:51:29 +00:00
69d276d7dc Removed residual echo complexity unittest.
This test produces a consistent stream of false positive alerts, and I have been unable to make it more robust, despite several attempts. It also has never managed to catch a real regression, so I think it is better to remove it.

Bug: chromium:788318
Change-Id: I7e9731834f67af1ef2fa15a655e620bd64a4cfde
Reviewed-on: https://webrtc-review.googlesource.com/25824
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20874}
2017-11-24 16:15:59 +00:00
7bd6cccb40 Android: Generate JNI code for DataChannel
Bug: webrtc:8278
Change-Id: I107c839656500971cbd3da7557e14776759c318a
Reviewed-on: https://webrtc-review.googlesource.com/25820
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20873}
2017-11-24 15:41:29 +00:00
b54bc06079 Let PrintResultMeanAndError accept a string.
Some downstream projects still use it. I'll update them and then revert
this change.

Bug: webrtc:8566
Change-Id: Ib4e56348c40a3645f3049382b47089ca6c675e96
Reviewed-on: https://webrtc-review.googlesource.com/25841
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20872}
2017-11-24 15:23:29 +00:00
a6092a9ea7 Deprecated the Get BitrateController method
GetBandwidthObserver should be used instead as it exposes a smaller interface.

Bug: webrtc:8415
Change-Id: I29ca795657e205186d7ebd929e756038a294b5f7
Reviewed-on: https://webrtc-review.googlesource.com/23900
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20871}
2017-11-24 15:07:44 +00:00
012b56b5d4 Reland "Add file capturer to AppRTCMobile on simulator."
This is a reland of 5adcd198752b651f7b7e9199a91f9b873b7d7237
Original change's description:
> Add file capturer to AppRTCMobile on simulator.
> 
> To achieve this, the CL does the following
> - Adds sample mp4 video
> - Refactors the existing RTCFileVideoCapturer to achieve continious
> capture and adds tests.
> 
> Bug: webrtc:8406
> Change-Id: Ibc0891176c58ec9053b42e340d2113036e7199ec
> Reviewed-on: https://webrtc-review.googlesource.com/12180
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20598}

Bug: webrtc:8406
Change-Id: I93be89b86e342a9a8195e19ebaf4aef1410d2c20
Reviewed-on: https://webrtc-review.googlesource.com/23200
Reviewed-by: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Daniela Jovanoska Petrenko <denicija@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20870}
2017-11-24 15:06:39 +00:00
09b9faed53 APM reports AEC3 ERL and ERLE metrics also via the old GetStatistics function.
The audio processing module reports the metrics 'echo return loss'
and 'echo return loss enhancement' for AEC3 via the old, soon to be
replaced, GetStatistics fuction.

Bug: webrtc:8533
Change-Id: I6b2286b5cdf8f20ebf14f82f1180f6bfb6c00c68
Reviewed-on: https://webrtc-review.googlesource.com/25642
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20869}
2017-11-24 14:13:59 +00:00
81af414ffb Replacing the legacy tool RTPjitter with a new rtp_jitter
This new tool provides the similar functionality as the legacy tool, but
is implemented using less legacy helpers. It also replaces RTPtimeshift
and RTPchange. The most significant change versus the old RTPjitter tool
is that the new tool takes the timing data in the form of integers in a
text file (instead of the binary data file used by the old tool). This
should make it easier to create custom timing files when needed.

Bug: webrtc:2692
Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287
Reviewed-on: https://webrtc-review.googlesource.com/25700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20868}
2017-11-24 13:38:59 +00:00
0e1d7989a5 Optional: Use nullopt and implicit construction in /api
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.

This CL was uploaded by git cl split.

R=pthatcher@webrtc.org

Bug: None
Change-Id: Icdc7e9e4395eeac053483c69e53501e5aa222107
Reviewed-on: https://webrtc-review.googlesource.com/23567
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20867}
2017-11-24 13:18:09 +00:00
f9d303c042 Make PrintResultMeanAndError receive two doubles instead of a string.
Bug: webrtc:8566
Change-Id: Ida925b030bff24275d34c0e888ee362e94c46b21
Reviewed-on: https://webrtc-review.googlesource.com/25540
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20866}
2017-11-24 12:14:48 +00:00
3e189a6dc3 Remove static library loading from WebRTC Android SDK.
Bug: webrtc:7474
Change-Id: Ie75a5c12638be82d7bd91073744946ac21c48155
Reviewed-on: https://webrtc-review.googlesource.com/22962
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20865}
2017-11-24 11:46:39 +00:00