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Author SHA1 Message Date
Guo-wei Shieh
a7446d2a50 Change DTLS default from 1.0 to 1.2 for webrtc. 
This changes for standalone webrtc applications.

BUG=
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1548733002 .

Cr-Commit-Position: refs/heads/master@{#11211}
 2016-01-11 23:27:12 +00:00
Jon Hjelle
f6c318ebae Update API for Objective-C RTCMediaSource. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1538263002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11210}
 2016-01-11 22:39:05 +00:00
Jon Hjelle
e799badacc Move Objective-C video renderers to webrtc/api/objc. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1542473003 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11209}
 2016-01-11 21:47:17 +00:00
Jon Hjelle
81028796bc Update API for Objective-C RTCMediaStreamTrack. 
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1527143002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11208}
 2016-01-11 21:16:19 +00:00
Jon Hjelle
a2c353f815 Update API for Objective-C RTCStats. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1540113002 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11207}
 2016-01-11 21:11:45 +00:00
danilchap
7e8145f05d [rtp_rtcp] rtcp::Tmmbr moved into own file 
BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1575023002

Cr-Commit-Position: refs/heads/master@{#11206}
 2016-01-11 19:49:24 +00:00
lally
27ed3cc28c SCTP: Stopped accepting SSRCs higher than max. 
Seems to fix asan-related crash.

BUG=https://code.google.com/p/chromium/issues/detail?id=570261

Review URL: https://codereview.webrtc.org/1571853002

Cr-Commit-Position: refs/heads/master@{#11205}
 2016-01-11 18:24:35 +00:00
hbos
a9a1d2acaf H.264: Default flags and pulling in openh264 and ffmpeg. 
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg).
These will be used in a follow-up CL that adds an encoder and decoder implementation.
The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added.

BUG=468365

Review URL: https://codereview.webrtc.org/1575913003

Cr-Commit-Position: refs/heads/master@{#11204}
 2016-01-11 18:19:06 +00:00
Jon Hjelle
7823495698 Move RTCI420Frame to webrtc/api/objc/RTCVideoFrame with minor style changes. 
BUG=
R=tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1533193003 .

Patch from Jon Hjelle <hjon@andyet.net>.

Cr-Commit-Position: refs/heads/master@{#11203}
 2016-01-11 17:47:14 +00:00
kjellander
fd99dea4f6 Roll chromium_revision 42ab10e..8c958e0 (368534:368561) 
Change log: 42ab10e..8c958e0
Full diff: 42ab10e..8c958e0

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1575003002

Cr-Commit-Position: refs/heads/master@{#11202}
 2016-01-11 12:57:26 +00:00
danilchap
ef3d805f6e [rtp_rtcp] rtcp::Tmmbn moved into own file 
explicetly unchanged.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1578713002

Cr-Commit-Position: refs/heads/master@{#11201}
 2016-01-11 11:31:17 +00:00
kjellander
d36efeb622 Roll chromium_revision e738b54..42ab10e (368533:368534) 
Change log: e738b54..42ab10e
Full diff: e738b54..42ab10e

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1573883002

Cr-Commit-Position: refs/heads/master@{#11200}
 2016-01-11 03:56:01 +00:00
kjellander
4de003722d Roll chromium_revision 7d97c94..e738b54 (368514:368533) 
Change log: 7d97c94..e738b54
Full diff: 7d97c94..e738b54

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1572173002

Cr-Commit-Position: refs/heads/master@{#11199}
 2016-01-10 11:58:43 +00:00
kjellander
3c05e6c9c3 Disable EndToEndTest.TransportSeqNumOnAudioAndVideo for Dr Memory. 
It started failing at the roll in
https://codereview.webrtc.org/1556273002

BUG=webrtc:5402
TBR=marpan@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1574813002

Cr-Commit-Position: refs/heads/master@{#11198}
 2016-01-10 05:46:42 +00:00
guoweis
daa87497e8 Revert of Roll chromium_revision 7d97c94..951c006 (368514:368525) (patchset #1 id:1 of https://codereview.webrtc.org/1577573002/ ) 
Reason for revert:
Win DrMemory Full: video_engine_tests failed 1

https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/3465

Original issue's description:
> Roll chromium_revision 7d97c94..951c006 (368514:368525)
>
> Change log: 7d97c94..951c006
> Full diff: 7d97c94..951c006
>
> No dependencies changed.
> No update to Clang.
>
> TBR=
>
> Committed: https://crrev.com/6109fc13aadebf7c5a990bbc78e981ab215321a6
> Cr-Commit-Position: refs/heads/master@{#11195}

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1570403002

Cr-Commit-Position: refs/heads/master@{#11197}
 2016-01-10 03:27:35 +00:00
Guo-wei Shieh
db21f633a2 fix GN build break on native_client 
TBR=guidou@webrtc.org
BUG=

Review URL: https://codereview.webrtc.org/1576723002 .

Cr-Commit-Position: refs/heads/master@{#11196}
 2016-01-09 21:12:11 +00:00
kjellander
6109fc13aa Roll chromium_revision 7d97c94..951c006 (368514:368525) 
Change log: 7d97c94..951c006
Full diff: 7d97c94..951c006

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1577573002

Cr-Commit-Position: refs/heads/master@{#11195}
 2016-01-09 19:59:36 +00:00
kjellander
0697db6f1b Roll chromium_revision 8a15a7f..7d97c94 (368391:368514) 
Change log: 8a15a7f..7d97c94
Full diff: 8a15a7f..7d97c94

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1577543002

Cr-Commit-Position: refs/heads/master@{#11194}
 2016-01-09 04:09:33 +00:00
Guo-wei Shieh
684e995464 Disable 2 video tests which fail on DrMemoryFull 
TBR=kjellander@webrtc.org
BUG=5417

Review URL: https://codereview.webrtc.org/1575433003 .

Cr-Commit-Position: refs/heads/master@{#11193}
 2016-01-09 03:03:50 +00:00
Taylor Brandstetter
f475d365a2 Properly handle different transports having different SSL roles. 
This meant splitting "transport_options" into audio/video/data options,
for when creating the answer, and giving "GetSslRole" a "transport_name"
parameter so we can retrieve the current role on a per-transport basis.

BUG=webrtc:4525
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1516993002 .

Cr-Commit-Position: refs/heads/master@{#11192}
 2016-01-08 23:36:06 +00:00
pkasting
25702cb162 Misc. small cleanups. 
* Better param names
* Avoid using negative values for (bogus) placeholder channel counts (mostly in tests).  Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases.
* Use arraysize()
* Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers
* reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead
* Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition
* Fix indenting
* Use uint32_t for timestamps (matching how it's already a uint32_t in most places)
* Spelling
* RTC_CHECK_EQ(expected, actual)
* Rewrap
* Use .empty()
* Be more pedantic about matching int/int32_t/
* Remove pointless consts on input parameters to functions
* Add missing sanity checks

All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first.

BUG=none
TEST=none

Review URL: https://codereview.webrtc.org/1534193008

Cr-Commit-Position: refs/heads/master@{#11191}
 2016-01-08 21:50:32 +00:00
kjellander
5de688ed34 Roll chromium_revision ede5d4f..8a15a7f (368310:368391) 
Change log: ede5d4f..8a15a7f
Full diff: ede5d4f..8a15a7f

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1573583002

Cr-Commit-Position: refs/heads/master@{#11190}
 2016-01-08 21:09:36 +00:00
minyue
49c454e748 Cleaning neteq_unittest resource files. 
BUG=webrtc:2692

Review URL: https://codereview.webrtc.org/1563983003

Cr-Commit-Position: refs/heads/master@{#11189}
 2016-01-08 19:30:18 +00:00
kjellander
f1685c771d Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac 
NOTRY=True
BUG=5407
TBR=stefan@webrtc.org,pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1569273003

Cr-Commit-Position: refs/heads/master@{#11188}
 2016-01-08 18:43:45 +00:00
stefan
e74eef19bd Add CreateSend/ReceiveTransport() methods to CallTest. 
This allows the test to create its own transports if it, for instance, needs to do demuxing.

BUG=webrtc:5416

Review URL: https://codereview.webrtc.org/1573453002

Cr-Commit-Position: refs/heads/master@{#11187}
 2016-01-08 14:47:21 +00:00
phoglund
37ebcf0ce5 Reland "Add APK targets to build libjingle tests for Android." 
patchset #10 id:180001 of https://codereview.webrtc.org/1511633002/

This reverts commit bc14164aad254e72ce4d1e381b912b7d3acf5391.

We have made more preparations downstream, so this should work now. Original CL by perkj@.

BUG=webrtc:2365
The work started from the work by kjellander@ in https://codereview.webrtc.org/1413663003/

Review URL: https://codereview.webrtc.org/1570513004

Cr-Commit-Position: refs/heads/master@{#11186}
 2016-01-08 13:05:01 +00:00
kjellander
b71b4f0c7a Update attributes to match gclibc's ansidecl.h 
To ease use of WebRTC in other codebases, update some macros
to match glibc's ansidecl.h, which uses double-underscores for attributes.

NOTRY=True

Review URL: https://codereview.webrtc.org/1571653002

Cr-Commit-Position: refs/heads/master@{#11185}
 2016-01-08 12:51:47 +00:00
kjellander
004851caca Roll chromium_revision 32569c6..ede5d4f (368258:368310) 
Change log: 32569c6..ede5d4f
Full diff: 32569c6..ede5d4f

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1567393002

Cr-Commit-Position: refs/heads/master@{#11184}
 2016-01-08 11:59:15 +00:00
henrik.lundin
e1ca167217 Add tracing to NetEqImpl::GetAudio 
BUG=webrtc:5167
R=pbos@webrtc.org
NOTRY=true

Review URL: https://codereview.webrtc.org/1571693002

Cr-Commit-Position: refs/heads/master@{#11183}
 2016-01-08 11:50:14 +00:00
andrew
ec80f03b3c Check the mic volume only periodically on Mac. 
Ask the OS for the mic volume every 1 second rather than with every 10
ms chunk. The previous behavior was consuming ~2% of the CPU load of
a voice engine call, and is now negligible.

This is consistent with the webrtc Windows Core Audio implementation,
as well as the Chromium Mac implementation:
https://code.google.com/p/chromium/codesearch#chromium/src/media/audio/agc_audio_stream.h

TEST=voe_cmd_test with AGC continues to work well on Mac.

Review URL: https://codereview.webrtc.org/1564223002

Cr-Commit-Position: refs/heads/master@{#11182}
 2016-01-08 09:16:25 +00:00
perkj
fbeb97e01f Fix clang warning in peerconnection_jni.cc 
TEST= export GYP_DEFINES="OS=android clang=1" ...
      ninja -C out/Debug AppRTCDemo
BUG=webrtc:5399

Review URL: https://codereview.webrtc.org/1561073005

Cr-Commit-Position: refs/heads/master@{#11181}
 2016-01-08 08:43:15 +00:00
asapersson
59bac1a4c5 Fix for stats updated twice when switching content type (realtime <-> screenshare). Add unittest. 
BUG=

Review URL: https://codereview.webrtc.org/1543933004

Cr-Commit-Position: refs/heads/master@{#11180}
 2016-01-08 07:36:06 +00:00
kjellander
95ab30ce88 Roll chromium_revision 6dd04c2..32569c6 (368115:368258) 
Change log: 6dd04c2..32569c6
Full diff: 6dd04c2..32569c6

Changed dependencies:
* src/third_party/libyuv: 1ccbf8f..fc52d8d
DEPS diff: 6dd04c2..32569c6/DEPS

No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1570713002

Cr-Commit-Position: refs/heads/master@{#11179}
 2016-01-08 07:31:13 +00:00
kjellander
a2b1e03c66 Disable AudioDeviceAPITest.MicrophoneVolumeTests on Linux. 
NOTRY=True
BUG=webrtc:5414
TBR=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1572503002

Cr-Commit-Position: refs/heads/master@{#11178}
 2016-01-08 07:26:11 +00:00
Taylor Brandstetter
893505d0fb Adding unit test to ensure TURN server priorities are unique. 
BUG=webrtc:5209
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1570563002 .

Cr-Commit-Position: refs/heads/master@{#11177}
 2016-01-07 23:12:53 +00:00
Taylor Brandstetter
e5ba13bc09 Adding a way for a Java RtpSender to set a track without taking ownership. 
This means that the track will still have a reference count after the
PeerConnection and RtpSender have been destroyed.

R=glaznev@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1566103003 .

Cr-Commit-Position: refs/heads/master@{#11176}
 2016-01-07 23:11:33 +00:00
kjellander
ced8ec9069 Roll chromium_revision bd5949f..6dd04c2 (368055:368115) 
Change log: bd5949f..6dd04c2
Full diff: bd5949f..6dd04c2

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1568963002

Cr-Commit-Position: refs/heads/master@{#11175}
 2016-01-07 21:22:52 +00:00
A.Brauckmann
bedc17be5c Fixing integer underflow in FileAudioDevice (webrtc issue 4554) 
Problem is described here:
https://code.google.com/p/webrtc/issues/detail?id=4554

Review URL: https://codereview.webrtc.org/1295603002

Cr-Commit-Position: refs/heads/master@{#11174}
 2016-01-07 20:38:36 +00:00
Marco
69387930e0 vp9 tests: Adjust some parameters and re-enable the tests. 
Tests were failing on android with new libvpx.
vp9 speed setting was changed to 8 recently and some recent changes
in libvpx require update for the tests to pass.

TBR=stefan@webrtc.org
BUG=webrtc:5401

Review URL: https://codereview.webrtc.org/1569903002 .

Cr-Commit-Position: refs/heads/master@{#11173}
 2016-01-07 20:00:44 +00:00
hjon
6f5ca080b8 Update API for Objective-C RTCMediaConstraints. 
BUG=

Review URL: https://codereview.webrtc.org/1543033003

Cr-Commit-Position: refs/heads/master@{#11172}
 2016-01-07 17:29:37 +00:00
Stefan Holmer
9fea80f50d Add audio streams to CallTest and a first A/V call test. 
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers.

Audio streams are using a fake audio device with file input.

The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code.

R=pbos@webrtc.org
TBR=kjellander@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1542653002 .

Cr-Commit-Position: refs/heads/master@{#11171}
 2016-01-07 16:43:31 +00:00
pbos
ecd21b481f Add ImplementationName to SimulcastEncoderAdapter. 
BUG=webrtc:4897
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1555673002

Cr-Commit-Position: refs/heads/master@{#11170}
 2016-01-07 16:03:13 +00:00
Peter Boström
01f364ef06 Remove always-on options in OveruseFrameDetector. 
BUG=webrtc:5132
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1556703002 .

Cr-Commit-Position: refs/heads/master@{#11169}
 2016-01-07 15:38:33 +00:00
henrika
30166cb1a8 iOS stability improvement for device switching, including BT devices 
BUG=webrtc:5058

Review URL: https://codereview.webrtc.org/1554163002

Cr-Commit-Position: refs/heads/master@{#11168}
 2016-01-07 15:23:08 +00:00
Peter Boström
7776e782d6 Remove unused methods in VideoCodingModule. 
Also voids ::Codec which always passed.

BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1464313004 .

Cr-Commit-Position: refs/heads/master@{#11167}
 2016-01-07 14:42:58 +00:00
Peter Boström
3886fc8f57 Use pointer to generated FEC packet. 
Removes multiple index lookups to generated_fec_packets_ speeding up
FecTest.FecTest with >2x in both Debug and Release, improving
performance but also readability.

On Debug this means that the slowest test in modules_tests now takes
~15-20 seconds instead of 50+ seconds, reducing the overall bottleneck.

BUG=webrtc:4712
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1552563003 .

Cr-Commit-Position: refs/heads/master@{#11166}
 2016-01-07 14:33:57 +00:00
henrika
46ea3ce580 AudioDeviceTest.StartPlayoutOnTwoInstances now verifies two active playing streams 
TBR=tkchin_webrtc
BUG=b/25343768

Review URL: https://codereview.webrtc.org/1527143007 .

Cr-Commit-Position: refs/heads/master@{#11165}
 2016-01-07 14:03:07 +00:00
kjellander
a49ad9856b Roll chromium_revision 4662d4f..bd5949f (368042:368055) 
Change log: 4662d4f..bd5949f
Full diff: 4662d4f..bd5949f

No dependencies changed.
No update to Clang.

TBR=

Review URL: https://codereview.webrtc.org/1562313002

Cr-Commit-Position: refs/heads/master@{#11164}
 2016-01-07 13:57:50 +00:00
kwiberg
cea7c2f783 Replace manual casting to rvalue reference with calls to std::move 
Review URL: https://codereview.webrtc.org/1570473002

Cr-Commit-Position: refs/heads/master@{#11163}
 2016-01-07 13:52:09 +00:00
kjellander
a46a4c92d0 Roll chromium_revision 2a70cb1..4662d4f (367468:368042) 
I had to fix the audio_device BUILD.gn which was forgotten back
in https://codereview.webrtc.org/1536923003. It also contained a few
missing source files and one library.

Change log: 2a70cb1..4662d4f
Full diff: 2a70cb1..4662d4f

Changed dependencies:
* src/buildtools: 6d0c448..0f8e6e4
* src/third_party/libsrtp: 8a7662a..ebfcc9a
DEPS diff: 2a70cb1..4662d4f/DEPS

No update to Clang.

TBR=henrika@webrtc.org
NOTRY=True

Review URL: https://codereview.webrtc.org/1565093002

Cr-Commit-Position: refs/heads/master@{#11162}
 2016-01-07 10:54:28 +00:00