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770 Commits

Author SHA1 Message Date
buildbot@webrtc.org
ea77334c30 (Auto)update libjingle 75302540-> 75327856 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 21:52:48 +00:00
henrike@webrtc.org
1d8f780779 Stop building talk/sound since it is no longer used. 
BUG=N/A
R=pbos@webrtc.org
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 17:16:56 +00:00
glaznev@webrtc.org
1d53f64b0f Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. 
webrtc::VideoEngine::SetAndroidObjects and webrtc::VoiceEngine::SetAndroidObjects
are not compatible with WEBRTC_CHROMIUM_BUILD. Since neither VoiceEngine nor VideoEngine
are needed at the time it's better to disable it completely.

BUG=https://crbug.com/412276
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 16:58:25 +00:00
henrikg@webrtc.org
307d3dbdee Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." 
Speculative revert, seems to be reason for flaky Win FYI bot compile break.

> Expose VideoEncoders with webrtc/video_encoder.h.
> 
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
> 
> BUG=3070
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/21929004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 09:48:30 +00:00
sprang@webrtc.org
c665dcb205 Revert 7145 "Stop building talk/sound since it is no longer used." 
> Stop building talk/sound since it is no longer used.
> 
> BUG=N/A
> R=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/22319004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 08:29:53 +00:00
henrik.lundin@webrtc.org
1972ff8a6e Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. 
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.

This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions).  I've removed some of
these.

This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class.  Removed "virtual" in those
cases.

BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 06:20:28 +00:00
henrike@webrtc.org
4c876453c8 Stop building talk/sound since it is no longer used. 
BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 22:18:04 +00:00
glaznev@webrtc.org
3472dcd7b0 Fix frame rate selection for Android camera. 
- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.

BUG=2622
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 19:24:57 +00:00
henrike@webrtc.org
b2efb6771c Put base tests in webrtc_tests.gyp 
BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 17:28:19 +00:00
jiayl@webrtc.org
b6d69282f5 Enable shared socket for TurnPort. 
In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address.

TESTED=AppRtc loopback call generates both turn and stun candidates.

BUG=1746
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 16:31:34 +00:00
buildbot@webrtc.org
5d639b3ef3 (Auto)update libjingle 75141932-> 75179475 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 07:57:12 +00:00
jiayl@webrtc.org
7d4891d3f1 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7068

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:43:15 +00:00
fbarchard@google.com
54cf1505e2 ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. 
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 18:34:53 +00:00
jiayl@webrtc.org
22406fcc9b TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7070

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 15:44:05 +00:00
mallinath@webrtc.org
3d81b1b22a Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got 
reverted due to some internal compile failures.

In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.

Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093

TBR=juberti@webrtc.org
BUG=1179

Review URL: https://webrtc-codereview.appspot.com/22329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 14:38:10 +00:00
andresp@webrtc.org
4d19e05ab2 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. 
This needs to happen sooner or later as if webrtc/base/checks.h happens to be included transitively here it would collide.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 11:45:44 +00:00
pbos@webrtc.org
b420191743 Expose VideoEncoders with webrtc/video_encoder.h. 
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 10:40:56 +00:00
henrike@webrtc.org
8b0b21161a Revert 7093: "Implementing ICE Transports type handling in libjingle transport." 
TBR=mallinath@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/28419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 22:46:28 +00:00
pbos@webrtc.org
7118e61669 Finish work queue in SctpDataMediaChannelTest. 
Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).

R=jiayl@webrtc.org
BUG=3608,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/28399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 21:44:07 +00:00
jiayl@webrtc.org
0e52772aa9 Fix a bot-breaking memory leak from early returning in ParseMediaDescription. 
BUG=3791
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 21:43:43 +00:00
jiayl@webrtc.org
c172320bd2 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. 
This reverts commit r7068.

TBR=kjellander@webrtc.org
BUG=2108

Review URL: https://webrtc-codereview.appspot.com/23539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:44:36 +00:00
buildbot@webrtc.org
fd42f9dd6f (Auto)update libjingle 74955991-> 75042522 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 19:45:36 +00:00
mallinath@webrtc.org
7256d31d28 Implementing ICE Transports type handling in libjingle transport. 
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 04:08:44 +00:00
thorcarpenter@google.com
cc060563f3 Remove unnecessary include from testutils.cc. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 21:19:00 +00:00
buildbot@webrtc.org
992febb997 (Auto)update libjingle 74873066-> 74873164 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:39:08 +00:00
thorcarpenter@google.com
a3344cfda4 Fix webrtcvideoframe tests. 
R=fbarchard@google.com, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:34:13 +00:00
jiayl@webrtc.org
ddb85ab85b Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added

BUG=3592
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:31:56 +00:00
buildbot@webrtc.org
af5fa95258 (Auto)update libjingle 74857067-> 74860820 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:03:50 +00:00
buildbot@webrtc.org
7e3bd3d7de (Auto)update libjingle 74851128-> 74857067 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 11:45:42 +00:00
buildbot@webrtc.org
bc6fa1876e (Auto)update libjingle 74825992-> 74851128 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 11:08:01 +00:00
buildbot@webrtc.org
818b7b3ac9 (Auto)update libjingle 74825084-> 74825992 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 00:14:03 +00:00
jiayl@webrtc.org
dfbcf8161e Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice. 
BUG=3778
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 00:01:12 +00:00
henrike@webrtc.org
f1427c6731 Revert 7070 "TurnPort should retry allocation with a new address on error 
STUN_ERROR_ALLOCATION_MISMATCH."

TBR=jiayl@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/15359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 22:21:33 +00:00
glaznev@webrtc.org
4b234044d5 Reduce maximum video resolution for Android. 
HW video encoder and decoder can not be initialized
with 3840x2160 resolution.

BUG=3757,3738
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 19:50:07 +00:00
jiayl@webrtc.org
574f2f60fe TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 19:11:34 +00:00
jiayl@webrtc.org
52055a276d Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 17:12:25 +00:00
pbos@webrtc.org
ceb956b29d Abort Negotiate() if DoCreateOffer() fails. 
Addressing crash in test.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/19239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7066 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 15:27:49 +00:00
pbos@webrtc.org
bcb6bcfe6c Remove HybridVideoEngine. 
This is currently unused dead code.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:32:26 +00:00
thorcarpenter@google.com
95c2458766 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files. 
"gcl try" fails to upload these large files so adding them independently.

R=andrew@webrtc.org, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 23:17:36 +00:00
buildbot@webrtc.org
609f987488 (Auto)update libjingle 74696326-> 74723281 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 21:50:32 +00:00
buildbot@webrtc.org
fa4535b270 (Auto)update libjingle 74694022-> 74696326 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 16:49:04 +00:00
pbos@webrtc.org
26c0c41a06 Network up/down signaling in Call. 
BUG=2429
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 16:17:12 +00:00
pbos@webrtc.org
ebee401230 Remove flake in SendsLowerResolutionOnSmallerFrames. 
Speculative fix for break on Linux64 Release. It looks like the second
frame is being dropped which is likely because the two frames are sent
too close to eachother. Adding a delay of 33ms in between them to make
sure the second one isn't dropped.

R=minyue@webrtc.org
TBR=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7043 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 15:52:02 +00:00
pbos@webrtc.org
c4175b9fdf Set resolution based on incoming VideoFrames. 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/17269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 15:25:49 +00:00
buildbot@webrtc.org
72e448559d (Auto)update libjingle 74628537-> 74648573 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 00:43:48 +00:00
tkchin@webrtc.org
90750482fa Remove deprecated RTCVideoRenderer constructor. 
Removes -[RTCVideoRenderer initWithView]. Also, fix potential issue where we hold on to a video frame longer than the lifetime of its associated track.

BUG=3341
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7032 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-02 20:50:00 +00:00
pbos@webrtc.org
9f341283f6 Remove WebRtcVideoEngine::default_codec_format(). 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-02 16:33:09 +00:00
pbos@webrtc.org
03655143db Remove files from talk/PRESUBMIT.py. 
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-02 16:17:36 +00:00
thakis@chromium.org
44010f3e52 win: Replace custom assert() macro with regular assert.h 
The current code got added in libjingle r103; I don't see a good reason for it.
Things still build with plain old assert.h.

The custom assert was wrong: __debugbreak() is documented to return void,
so doing `cond ? true : __debugbreak()` shouldn't build (and it doesn't in
clang-cl). It's possible to make it build by writing
`cond ? true : (__debugbreak(), true)`, but just using the regular header
seems like a much better fix.

BUG=chromium:82385
Review URL: https://webrtc-codereview.appspot.com/19139004/


git-svn-id: http://webrtc.googlecode.com/svn/trunk@7007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-29 03:00:15 +00:00
jiayl@webrtc.org
bc3f333905 Add jiayl to talk OWNERS. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 23:24:36 +00:00
jiayl@webrtc.org
e21cc9ae2a When the peerconnection creates the offer with a constraint to disable the audio offering, stats will not get properly updated. 
constraints . SetMandatoryReceiveAudio (false);

The problem is that webrtc::GetTrackIdBySsrc returns false if audio is not available. However it should continue and check for the video track.

BUG=webrtc:3755
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 22:21:34 +00:00
niklas.enbom@webrtc.org
4431fd6ad5 Add 60 fps video support 
R=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 14:57:46 +00:00
buildbot@webrtc.org
1f8a23757a (Auto)update libjingle 74235596-> 74297316 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-28 10:52:44 +00:00
pbos@webrtc.org
75c3ec1763 Fix data races during VideoAdapterTest tear-down. 
Explicitly disconnect the VideoCapturer to avoid frames being
delivered during listener destruction. This manifested only on DrMemory
Full on Windows which I was able to repro locally.

BUG=3671
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 18:16:13 +00:00
buildbot@webrtc.org
573a1eef3d (Auto)update libjingle 74202294-> 74230205 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 17:21:19 +00:00
solenberg@webrtc.org
00f11f5e24 - Make local constant non-static. 
- Remove spammy log line.

BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-27 08:52:17 +00:00
guoweis@webrtc.org
7087857afd implement handling ALTERNATE-SERVER response from turn protocol as 
specified in RFC 5766, also created 2 test cases for both the normal
redirection case as well as when a pingpong situation happens, the
allocation should fail

BUG=1986 TURN ALTERNATE-SERVER support
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 21:37:49 +00:00
buildbot@webrtc.org
3533bfcb94 (Auto)update libjingle 74132319-> 74133664 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 15:50:23 +00:00
buildbot@webrtc.org
4470d78c9b (Auto)update libjingle 74128148-> 74132319 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 15:24:54 +00:00
pbos@webrtc.org
f21ac1fd46 Fix Win64 compile of videoadapter_unittest.cc. 
Missed an typecast in videoadapter_unittest.cc in r6979 due to
tryservers being clogged and me waiting for a windows, linux, mac and
tsanv2 bot to finish was not enough. Committing fix straight away to
un-break tree.

TBR=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/18279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 12:46:57 +00:00
pbos@webrtc.org
c9b3f77e65 Fix data races in VideoAdapterTest. 
Adressing clear races between the test thread and capturer thread shown
as heap-use-after-free in vpx_codec_destroy in
WebRtcVideoMediaChannelTest.SetSend (way later in the rest run).

When capturing a frame the test copied it to a separate frame that would
then be read by the test without synchronization, if the test didn't
manage to examine the frame in between captures the adapted frame would
be overwritten by the following frame during accesses to it.

The actual races are suppressed by race:webrtc/base/messagequeue.cc and
race:webrtc/base/thread.cc. These fixes reduce the suppression count
locally from around 3000 to 30 for VideoAdapterTest.*.

Also removing tsan suppressions for talk/base as it's been moved to
webrtc/base.

R=tommi@webrtc.org
BUG=3671

Review URL: https://webrtc-codereview.appspot.com/22169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 12:33:18 +00:00
pbos@webrtc.org
b648b9d85c Remove test constructor in WebRtcVideoEngine2. 
Removes the need for ::Construct().

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 11:08:06 +00:00
kjellander@webrtc.org
b96ea2aab5 Remove former team members from OWNERS and WATCHLISTS 
Remove the following (CCed) former team members from all
OWNERS files and the WATCHLISTS file:
* fischman@
* leozwang@
* mikhal@
* pwestin@
* wu@

BUG=
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 06:12:08 +00:00
buildbot@webrtc.org
204cd56007 (Auto)update libjingle 74064646-> 74072040 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 21:10:18 +00:00
kjellander@webrtc.org
e9bfed0648 Move constant so it is not stripped out for TSAN bots. 
BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 19:46:26 +00:00
buildbot@webrtc.org
857130fd5b (Auto)update libjingle 74039473-> 74044292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 16:07:12 +00:00
solenberg@webrtc.org
6556a59db1 As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests. 
Also, caused some issues with other peerconnection_unittest tests, so changed the design of those.

BUG=
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 14:35:40 +00:00
buildbot@webrtc.org
b4c7b09c13 (Auto)update libjingle 73927775-> 74032598 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-25 12:11:58 +00:00
buildbot@webrtc.org
3740d74106 (Auto)update libjingle 73927658-> 73927775 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 22:27:04 +00:00
buildbot@webrtc.org
309a611670 (Auto)update libjingle 73891518-> 73927658 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 22:24:54 +00:00
buildbot@webrtc.org
2b0554f0e7 (Auto)update libjingle 73794259-> 73891518 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 14:08:15 +00:00
pbos@webrtc.org
97fdeb8329 Remove static initializer in WebRtcVideoEngine2. 
Blocks import into chromium.

R=tommi@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/18249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6954 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-22 10:36:23 +00:00
phoglund@webrtc.org
7bd5fefb17 Making sure muc members get recorded. 
This is an upstream of a change I made; will describe in a separate
email thread.

Essentially, the members map wasn't getting populated in the callclient
example, so it was always empty. Now it will be populated correctly.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-21 09:53:28 +00:00
henrik.lundin@webrtc.org
6908b84179 Disable two tests in TurnPortTest 
The tests are flaky.

BUG=3720
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-20 09:47:58 +00:00
buildbot@webrtc.org
95bbd18696 (Auto)update libjingle 73627179-> 73695227 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-20 07:49:30 +00:00
buildbot@webrtc.org
5a60aed80f (Auto)update libjingle 73626701-> 73627179 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6930 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-19 15:11:45 +00:00
buildbot@webrtc.org
84532e59dd (Auto)update libjingle 73626167-> 73626701 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6929 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-19 15:05:18 +00:00
henrike@webrtc.org
0481f15f02 (Auto)update libjingle 73399579-> 73626167 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6928 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-19 14:56:59 +00:00
houssainy@google.com
d5b292e450 Active connection stats [LocalAddress,RemoteAddress,LocalCandidateType...etc] 
is now printed in the head-up display in Android appRTC.

This printing will be usefull in debugging switching ICE candidates.

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

Review URL: https://webrtc-codereview.appspot.com/13189005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-19 11:43:32 +00:00
buildbot@webrtc.org
353cd37ae9 (Auto)update libjingle 73370064-> 73399579 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6911 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-15 18:26:12 +00:00
tommi@webrtc.org
5b06b06cc0 Revert 6897 (i.e. Reland 6863) - "Revert 6863 "Refactor StatsCollector and associated..." 
The bot that had the problem was using an old version of STL, so relanding.

> Revert 6863 "Refactor StatsCollector and associated types."
> 
> Breaks chrome compilation on Mac:
> 
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
> error: no matching constructor for initialization of
> 'webrtc::StatsReport'
>           _Tp __x_copy = __x;
>               ^          ~~~
> /Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::_M_insert_aux' requested here
>           _M_insert_aux(end(), __x);
>           ^
> ../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
> note: in instantiation of member function
> 'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
> >::push_back' requested here
>   reports.push_back(report1);
>           ^
> ../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
> note: candidate constructor not viable: requires 0 arguments, but 1
> was provided
>   StatsReport() : timestamp(0) {}
> 
> 
> 
> > Refactor StatsCollector and associated types.
> > * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> > * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> > * Report ids are now const.
> > * Copying of data has been greatly reduced.
> > * This change includes preparation work for making GetStats fully async.
> > 
> > This is a reland of r6778 which was reverted due to fyi bots failing.
> > I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> > 
> > R=xians@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/15119004
> 
> TBR=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/21169004

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-15 08:38:30 +00:00
buildbot@webrtc.org
c3df61e351 (Auto)update libjingle 73256845-> 73260148 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 23:57:23 +00:00
niklas.enbom@webrtc.org
22fa032f22 Revert 6863 "Refactor StatsCollector and associated types." 
Breaks chrome compilation on Mac:

/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/vector.tcc:252:8:
error: no matching constructor for initialization of
'webrtc::StatsReport'
          _Tp __x_copy = __x;
              ^          ~~~
/Applications/Xcode.app/Contents/Developer/Platforms/MacOSX.platform/Developer/SDKs/MacOSX10.6.sdk/usr/include/c++/4.2.1/bits/stl_vector.h:608:4:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::_M_insert_aux' requested here
          _M_insert_aux(end(), __x);
          ^
../../content/renderer/media/mock_peer_connection_impl.cc:282:11:
note: in instantiation of member function
'std::vector<webrtc::StatsReport, std::allocator<webrtc::StatsReport>
>::push_back' requested here
  reports.push_back(report1);
          ^
../../third_party/libjingle/source/talk/app/webrtc/statstypes.h:49:3:
note: candidate constructor not viable: requires 0 arguments, but 1
was provided
  StatsReport() : timestamp(0) {}



> Refactor StatsCollector and associated types.
> * Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
> * Reports are now managed in a set, not a map, since it's enough to store the id in one place.
> * Report ids are now const.
> * Copying of data has been greatly reduced.
> * This change includes preparation work for making GetStats fully async.
> 
> This is a reland of r6778 which was reverted due to fyi bots failing.
> I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/15119004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6897 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 23:11:04 +00:00
buildbot@webrtc.org
449ad98aeb (Auto)update libjingle 73248599-> 73249894 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 21:55:18 +00:00
pbos@webrtc.org
ef8bb8d9b0 Make sure that muting muted streams succeeds. 
We don't want to report an error here, and PeerConnection relies on
being able to mute already-muted streams (I hit an assert when testing
manually).

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 21:36:18 +00:00
pbos@webrtc.org
432893a100 Remove TODO saying to remove WebRtcVideoFrame. 
Comment was added prematurely, there's no decision to get rid of this
type.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 21:17:22 +00:00
pbos@webrtc.org
b15dddf7ae Remove files from talk/PRESUBMIT.py blacklist. 
Many files can now be submitted here and do not have to be rolled in.

BUG=
R=henrike@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6893 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 20:38:53 +00:00
henrike@webrtc.org
d968dd039a Fixes failure triggered by include order re-ordering. 
BUG=N/A
TBR=marpan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 18:39:43 +00:00
buildbot@webrtc.org
a09a99950e (Auto)update libjingle 73222930-> 73226398 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 17:26:08 +00:00
buildbot@webrtc.org
2c0fb05f16 (Auto)update libjingle 73221069-> 73222930 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6889 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 16:47:12 +00:00
buildbot@webrtc.org
67f849575c (Auto)update libjingle 73215194-> 73221069 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 16:22:04 +00:00
henrike@webrtc.org
4eeeefebb2 (Auto)update libjingle 73072800 -> 73215194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6887 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 14:57:30 +00:00
xians@webrtc.org
38d88816e3 Fix the audio source failure due to unsupported constraints. 
Some constraints, like kEchoCancellation, kMediaStreamAudioDucking are supported in Chrome but not in Libjingle, if the users set it in mandatory, LocalAudioSource::Initialize() will fail the getUserMedia call.

This patch fixes the problem by fully initializing the LocalAudioSource even though some constraints are not supported in libjingle.

BUT=crbug/398080
TEST=manual test:
var constraints = {audio: { mandatory: { googEchoCancellation: true } }};
getUserMedia(constraints, gotStream, gotStreamFailed);
verify you get a gotStream callback

R=henrika@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 13:51:58 +00:00
mallinath@webrtc.org
e999bd087b Removing ASSERT for tcp candidate for port 0 and 9, as Android clients 
may not be called with set_allow_tcp_listen(false).

This CL will also sends tcp candidate in RFC 6544 format.

BUG=https://code.google.com/p/webrtc/issues/detail?id=3677
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-13 06:05:55 +00:00
pbos@webrtc.org
afb554f404 Move default-recv-channels to a separate class. 
BUG=1788,3099
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6879 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 23:17:13 +00:00
pbos@webrtc.org
c3d2bd28a3 Fix GetStats() crash. 
GetStats() can be called before codecs are set and the underlying
webrtc::VideoSendStream is created, leading to a null-pointer
dereference.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 20:55:10 +00:00
buildbot@webrtc.org
8d57f08902 (Auto)update libjingle 73072800-> 73072800 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6873 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 14:41:46 +00:00
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
tommi@webrtc.org
730bf30da7 Refactor StatsCollector and associated types. 
* Due to the type changes, I'm going to update the OnCompleted event in two phases to sync with Chrome. This is the first phase.
* Reports are now managed in a set, not a map, since it's enough to store the id in one place.
* Report ids are now const.
* Copying of data has been greatly reduced.
* This change includes preparation work for making GetStats fully async.

This is a reland of r6778 which was reverted due to fyi bots failing.
I found and fixed the issue which was that in a couple of places I needed to replace a report instead of finding+updating an existing one.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6863 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 14:08:33 +00:00
jiayl@webrtc.org
7ec3f9f838 Fix a bug in parsing IceCandidate with IPV6 address. 
It used to treat ":" as a candidate delimiter and got confused by the ":" in the IPV6 address.
The new logic is to check if the input has multiple lines. If so, returns error.

BUG=3669
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6859 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 23:09:15 +00:00