20163 Commits

Author SHA1 Message Date
Philip Eliasson
deb866360a Revert "Add stereo codec header and pass it through RTP"
This reverts commit 20f2133d5dbd1591b89425b24db3b1e09fbcf0b1.

Reason for revert: Breaks downstream project.

Original change's description:
> Add stereo codec header and pass it through RTP
> 
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
> 
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
> 
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}

TBR=danilchap@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org,emircan@webrtc.org

Change-Id: I57f3172ca3c60a84537d577a574dc8018e12d634
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/26940
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20931}
2017-11-29 11:39:41 +00:00
Oskar Sundbom
f82000328d Optional: Use nullopt and implicit construction in /rtc_base/rate_statistics.cc
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=sprang@webrtc.org

Bug: None
Change-Id: I50d25d6174486928963c2e98455587a8a9f0bee6
Reviewed-on: https://webrtc-review.googlesource.com/23616
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20930}
2017-11-29 11:37:29 +00:00
Oskar Sundbom
903dcd733a Optional: Use nullopt and implicit construction in /p2p
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: Ia65be19b24c93db360a313f82a84bfae1a49bf2d
Reviewed-on: https://webrtc-review.googlesource.com/23605
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20929}
2017-11-29 11:13:39 +00:00
Patrik Höglund
ebe62408b5 Fix circular dependency in rtc_event_log.
Bug: webrtc:6828
Change-Id: Ief948b6799455cfda6cb89e2e632f5fd42df0881
Reviewed-on: https://webrtc-review.googlesource.com/25840
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20928}
2017-11-29 10:46:19 +00:00
Pengyu Liao
570cf968eb Fix playout (recording from caller point of view) functionality for FileAudioDevice.
Bug: webrtc:8585
Change-Id: Ied2cbea146560488b07ac74bd3c5009f8804f1a0
Reviewed-on: https://webrtc-review.googlesource.com/26440
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20927}
2017-11-29 10:26:20 +00:00
Magnus Jedvert
43d069a2cd Revert "Android: Generate JNI code for stats"
This reverts commit aede67a199ae0552074bfec4bb03cc9a6a5fba0f.

Reason for revert: Causes error:
JNI ERROR (app bug): local reference table overflow (max=512)'

Original change's description:
> Android: Generate JNI code for stats
> 
> This CL also unifies the functions for converting from C++ to Java, and
> generates the boiler plate for converting C++ vectors to Java arrays.
> 
> Bug: webrtc:8278
> Change-Id: I262e9162beae8a64ba0e8b6a27e1081207b03961
> Reviewed-on: https://webrtc-review.googlesource.com/26020
> Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20918}

TBR=magjed@webrtc.org,sakal@webrtc.org

Change-Id: Ieb26ed8577bd489a4dd4f7542d16a7d0e11f409f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8278
Reviewed-on: https://webrtc-review.googlesource.com/26900
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20926}
2017-11-29 10:09:10 +00:00
Danil Chapovalov
8d19e03e95 Simpliy RtcpTransceiver::SendImmediateFeedback signature
and add implementation comment

Bug: webrtc:8239
Change-Id: Id24937018d386e386b8241aca8f5d686e7cc527a
Reviewed-on: https://webrtc-review.googlesource.com/26600
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20925}
2017-11-29 09:51:20 +00:00
Gustaf Ullberg
2723fb162c Added ERL and ERLE metrics to UMA.
Bug: webrtc:8569
Change-Id: Ie820ebbe6ea1d8742c32a7aba540cfebd8757818
Reviewed-on: https://webrtc-review.googlesource.com/25560
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20924}
2017-11-29 09:06:59 +00:00
Henrik Lundin
abbff89b29 Add new UMA metric for NetEq target buffer delay
The UMA metric will log the same information that goes into the
googPreferredJitterBufferMs stat.

Bug: webrtc:8488
Change-Id: I4e4e1e362dd42377105d52d2c4cd49c1ecb1a90d
Reviewed-on: https://webrtc-review.googlesource.com/26740
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20923}
2017-11-29 08:56:29 +00:00
Steve Anton
4e70a72571 Replace MediaContentDirection with RtpTransceiverDirection
Bug: webrtc:8558
Change-Id: I410d17cce235e0b42038cf0b125fd916010f50ae
Reviewed-on: https://webrtc-review.googlesource.com/24745
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20922}
2017-11-28 23:44:28 +00:00
Tommi
36207d600a Remove "using rtc::scoped_ptr" from audiotrack.cc.
This is causing compilation issues in a chromium cl because of type conflicts.

BUG=none
TBR=henrikg@webrtc.org

Tbr-ing to fix build issue upstream and because there's no code change.

Change-Id: Ia34ae3844fe3f57f047cb44422fa591f752b7bda
Reviewed-on: https://webrtc-review.googlesource.com/26680
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20921}
2017-11-28 21:29:18 +00:00
Emircan Uysaler
20f2133d5d Add stereo codec header and pass it through RTP
- Defines CodecSpecificInfoStereo that carries stereo specific header info from
encoded image.
- Defines RTPVideoHeaderStereo that carries the above info to packetizer,
see module_common_types.h.
- Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
header.
- Uses new data containers in StereoAdapter classes.

This CL is the step 3 for adding alpha channel support over the wire in webrtc.
See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
CL that gives an idea about how it will come together.
Design Doc: https://goo.gl/sFeSUT

Bug: webrtc:7671
Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
Reviewed-on: https://webrtc-review.googlesource.com/22900
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20920}
2017-11-28 18:43:43 +00:00
Oleh Prypin
ceec4d0a97 Roll chromium_revision 5df4e1bfe7..5bd5874cbf (518692:519731) + iOS fix
Pin iOS version to 9.0, to avoid the upstream change to 10.0

Change log: 5df4e1bfe7..5bd5874cbf
Full diff: 5df4e1bfe7..5bd5874cbf

Changed dependencies:
* src/base: 4843599735..fc034c4143
* src/build: 99653454ee..f0766940d5
* src/ios: 7287c9baeb..49bd74cee7
* src/testing: 60c665fffe..373652d16f
* src/third_party: 6a94dad699..34c5bb433a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2ffb57b102..230a61040f
* src/third_party/depot_tools: 7d9d9233cb..1b30125fbc
* src/third_party/ffmpeg: abead8cbcf..9cb03e5705
* src/third_party/icu: 5ed26985c0..741688ebf3
* src/third_party/libvpx/source/libvpx: ea14a1a965..cbe62b9c2d
* src/tools: 4e917ad6ac..8d915c324e
DEPS diff: 5df4e1bfe7..5bd5874cbf/DEPS

No update to Clang.

TBR=marpan@webrtc.org
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Bug: webrtc:8570, webrtc:8580
Change-Id: I36440e0e1a3b3daac767e196acd5bfadb4eb6d9c
Reviewed-on: https://webrtc-review.googlesource.com/26026
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20919}
2017-11-28 18:42:38 +00:00
Magnus Jedvert
aede67a199 Android: Generate JNI code for stats
This CL also unifies the functions for converting from C++ to Java, and
generates the boiler plate for converting C++ vectors to Java arrays.

Bug: webrtc:8278
Change-Id: I262e9162beae8a64ba0e8b6a27e1081207b03961
Reviewed-on: https://webrtc-review.googlesource.com/26020
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20918}
2017-11-28 17:27:58 +00:00
Rasmus Brandt
3fb614bc93 Remove unused UlpfecGenerator::BuildRedPacket.
BUG=none

Change-Id: I998e23beee9c46dc696631195790e8821d1cc967
Reviewed-on: https://webrtc-review.googlesource.com/24821
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20917}
2017-11-28 16:18:28 +00:00
Oskar Sundbom
248ccf8ad4 Optional: Use nullopt and implicit construction in /
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=tommi@webrtc.org

Bug: None
Change-Id: I0ca1b624859a6561e227480b7dac8c254d26ad57
Reviewed-on: https://webrtc-review.googlesource.com/23562
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20916}
2017-11-28 16:03:48 +00:00
Oskar Sundbom
5e1a7496bb Optional: Use nullopt and implicit construction in /common_audio
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=tina.legrand@webrtc.org

Bug: None
Change-Id: Iea6f04db7c1f92fe9da2c855bb60ad2f70c371d3
Reviewed-on: https://webrtc-review.googlesource.com/23615
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20915}
2017-11-28 15:46:48 +00:00
Patrik Höglund
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
Patrik Höglund
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
Erik Språng
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
Danil Chapovalov
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
Magnus Jedvert
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
Henrik Lundin
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
Edward Lemur
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
Danil Chapovalov
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
Edward Lemur
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
Henrik Lundin
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
Danil Chapovalov
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
henrika
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
Zijie He
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
braveyao
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
Steve Anton
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
Steve Anton
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
Steve Anton
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
Steve Anton
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
Steve Anton
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
Oskar Sundbom
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
Erik Språng
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
Kári Tristan Helgason
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
Per Åhgren
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
Magnus Jedvert
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
Oleh Prypin
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
Tommi
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
Magnus Jedvert
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
Oskar Sundbom
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
Alex Narest
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
Henrik Boström
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
Oskar Sundbom
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
VladimirTechMan
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
Karl Wiberg
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