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984 Commits

Author SHA1 Message Date
pbos@webrtc.org
1795c358fc Add default implementation of Add/RemoveObserver. 
Needed to remove Add/RemoveObserver from RTCVideoEncoderFactory in
Chromium before removing these completely. This is done to keep the
chromium.webrtc.fyi bots happy and to make rolling webrtc revisions
easier.

R=stefan@webrtc.org
BUG=3876

Review URL: https://webrtc-codereview.appspot.com/23839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:45:25 +00:00
kjellander@webrtc.org
8cad9432d5 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" 
Breaks debug compilation (didn't run all trybots when testing this).

> Update isolate.gypi files + link to isolate_driver.py
> 
> This updates the isolate.gypi copies we're forced to
> maintain in our code repo to Chromium revision c264a05.
> 
> Since isolated testing is now using a new launch script
> in tools: isolate_driver.py, that is added to our links
> script.
> 
> BUG=395700
> TESTED=Ran one of our tests with:
> ninja -C out/Release tools_unittests_run
> tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
> 
> R=henrika@webrtc.org, jam@chromium.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26649004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 08:44:00 +00:00
kjellander@webrtc.org
02cd3067d2 Update isolate.gypi files + link to isolate_driver.py 
This updates the isolate.gypi copies we're forced to
maintain in our code repo to Chromium revision c264a05.

Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that is added to our links
script.

BUG=395700
TESTED=Ran one of our tests with:
ninja -C out/Release tools_unittests_run
tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate

R=henrika@webrtc.org, jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/26649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 08:34:57 +00:00
glaznev@webrtc.org
359d720004 Allow Android apps to set video renderer scaling type. 
Also add type check for EGL context object received from apps and
switch to byte buffer video decoding if EGL context is incorrect

BUG=3851
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 23:07:08 +00:00
jiayl@webrtc.org
7dfb7fa189 Reland disallowing blocking calls on the worker thread. 
This fixed the issue that invoking the call when the thread is not started.

BUG=3559
R=juberti@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/24769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 22:45:55 +00:00
asapersson@webrtc.org
626624061e Disable flaky tests: 
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined

BUG=3871
R=henrike@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 14:30:07 +00:00
pbos@webrtc.org
34f2a9ea72 Initialize SSL in unittest_main.cc. 
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 11:36:45 +00:00
jiayl@webrtc.org
bebc75e8bd Fix the duplicated candidate problem when using multiple STUN servers. 
BUG=3723
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 23:01:11 +00:00
thorcarpenter@google.com
a21d071607 Reverting part of 
https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80
because of a major regression hanging the executable on start.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 17:19:14 +00:00
pbos@webrtc.org
05305116d6 Explicitly initialize SSL for tests. 
Adding missing SSL initialization/cleanups in
TransportDescriptionFactoryTest and MediaSessionTest.

These being missing prevent these tests from being run individually
without other tests preceding them that initialize SSL.

BUG=3860
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 15:50:26 +00:00
jiayl@webrtc.org
3987b6de50 Fix a problem in Thread::Send. 
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 17:14:05 +00:00
pbos@webrtc.org
d60d79a145 Thread annotation of rtc::CriticalSection. 
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 07:10:57 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +00:00
glaznev@webrtc.org
8166faeff3 Change Android video renderer to maintain video aspect 
ratio when displaying camera or decoded video frames.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 23:58:52 +00:00
glaznev@webrtc.org
90668b1633 Switch HW video decoder to output byte buffers if video 
renderer EGL context is not provided by app.

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 21:42:15 +00:00
buildbot@webrtc.org
1b7dcc1647 (Auto)update libjingle 76169599-> 76176062 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 17:41:48 +00:00
guoweis@webrtc.org
2c1bcea1bc Enable ipv6 by default for webrtc under a Finch experiment. 
Reapply 23529005 after fixing the build break issue (Chromium:582133002)

Committed: https://code.google.com/p/webrtc/source/detail?r=7253

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 16:23:02 +00:00
henrik.lundin@webrtc.org
3987f10c11 Revert "Remove DTMF status methods from Voice Engine" r7276 
This change caused some trouble.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 13:15:14 +00:00
henrik.lundin@webrtc.org
bf7b9e0081 Remove DTMF status methods from Voice Engine 
These methods are not used.

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

Review URL: https://webrtc-codereview.appspot.com/24689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 12:54:04 +00:00
pbos@webrtc.org
0a2087a711 Skeleton for registering external encoders/decoders. 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 09:40:22 +00:00
pbos@webrtc.org
83f95ba9a6 Remove engine-level SetOptions. 
Already removed in WebRtcVideoEngine.

R=andresp@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 16:07:18 +00:00
henrik.lundin@webrtc.org
64a2f10f4b Remove Get/SetNetEQPlayoutMode APIs 
These are not used anymore.

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

Review URL: https://webrtc-codereview.appspot.com/26569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 14:30:10 +00:00
guoweis@webrtc.org
97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002) 
Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 21:06:12 +00:00
buildbot@webrtc.org
ed5ca1f122 (Auto)update libjingle 75925673-> 75926712 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:30:44 +00:00
buildbot@webrtc.org
c98f217c65 (Auto)update libjingle 75924589-> 75925673 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:18:10 +00:00
buildbot@webrtc.org
0c9fe72b21 (Auto)update libjingle 75922684-> 75924589 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:05:02 +00:00
glaznev@webrtc.org
ebf2757339 Fix HW video decoder crash on some Android KK devices. 
Remove direct access to decoder Java output buffer memory
when HW decoder is configured to decode to surface.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 19:36:13 +00:00
thorcarpenter@google.com
c1eebfa107 Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. 
R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/22699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 17:54:00 +00:00
glaznev@webrtc.org
e65812427d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. 
Symbol LogcatTraceContext not defined.
Submitting on behalf of serya@.
Dup of https://webrtc-codereview.appspot.com/29529004/

TEST=Build target libjingle_peerconnection_javalib with applied CL https://codereview.chromium.org/551793003/
BUG=https://crbug.com/383418
R=serya@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 16:53:46 +00:00
pbos@webrtc.org
bbe0a8517d Config struct for VideoEncoder. 
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).

BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 12:30:25 +00:00
buildbot@webrtc.org
6e5c78422d (Auto)update libjingle 75875619-> 75878731 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 06:46:37 +00:00
buildbot@webrtc.org
b5a5c44ef7 (Auto)update libjingle 75865376-> 75875619 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 05:36:18 +00:00
buildbot@webrtc.org
d7acf11e8d (Auto)update libjingle 75854833-> 75865376 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 02:01:09 +00:00
buildbot@webrtc.org
ccb3e3f3db (Auto)update libjingle 75854418-> 75854833 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:31:03 +00:00
buildbot@webrtc.org
dcc1f0426b (Auto)update libjingle 75852725-> 75853560 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:14:12 +00:00
glaznev@webrtc.org
0b435ba597 A few fixes to avoid crash in HW codec on device orientation change. 
- Fix video encoder Reset() function to avoid setting codec
resolution to zero.
- Follow SW codec implementation and do not crash when frame
with the resolution different from the encoder resolution arrives.
Instead wait for at least 3 frames with new resolution and
re-initialize the codec. HW codec reset may take much longer
than SW codec, so these 3 frames threshold avoids resetting
codec when outstanding camera frame captured from previous device
orientation arrives.
- Plus some minor changes to make encoder reset/release
implementation closer to decoder implementation.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:01:03 +00:00
glaznev@webrtc.org
83af77bf3c Revert maximum video codec resolution on Android back to 720p again. 
Some low end Android devices still have problems with 1080p support.

BUG=3757
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 21:11:29 +00:00
buildbot@webrtc.org
933d88af58 (Auto)update libjingle 75818332-> 75837294 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 20:23:05 +00:00
jiayl@webrtc.org
42731bdded Avoid writing a double/float to a string to avoid a crash. 
BUG=crbug/367223
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 16:51:51 +00:00
pbos@webrtc.org
6cd6ba8ae0 Expose VP8/H264 defaults through video_encoder.h. 
Reduces code duplication quite a bit, these identical defaults were set
in quite a few different places.

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

Review URL: https://webrtc-codereview.appspot.com/19299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 12:42:28 +00:00
andresp@webrtc.org
ab071daab8 Split video_render_module implementation into default and internal implementation. 
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 08:58:15 +00:00
guoweis@webrtc.org
369a637ac8 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Committed: https://code.google.com/p/webrtc/source/detail?r=7201

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 22:37:29 +00:00
glaznev@webrtc.org
3b67f8e0ca Enable HW video decoding on Qualcomm devices. 
Parallel decoding and encoding problem is fixed now
(b/16353967), so it is possible to start using Qualcomm
VP8 HW decoder. Bitrate overshoots should be fixed as well.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 21:25:51 +00:00
henrike@webrtc.org
4a5061fbff talk/p2p/base: removed unused variable "port_" 
BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 12:33:07 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture) 
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:50:19 +00:00
andresp@webrtc.org
85ef770d92 Split video engine android initialization into each internal module initialization. 
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.

BUG=3768,3770
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:44:51 +00:00
pbos@webrtc.org
ab990ae43a Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" 
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.

BUG=3070
TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 09:02:25 +00:00
buildbot@webrtc.org
6a9b155798 (Auto)update libjingle 75683337-> 75695882 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 08:08:38 +00:00
glaznev@webrtc.org
a59c501c99 Java VideoRenderer class may be backed by two different native 
classes depending on type of rendering.
Fix crash in AppRtcDemo by calling correct destructor on exit.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 03:26:59 +00:00
guoweis@webrtc.org
40c2aa36f2 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 20:29:41 +00:00
guoweis@webrtc.org
f8bff762d1 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 20:17:22 +00:00
pbos@webrtc.org
cddd17c0f8 Recreate VideoStreams when setting resolution. 
Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 16:33:13 +00:00
pbos@webrtc.org
88e85ad64d Add pbos@webrtc.org (myself) to talk/media/webrtc/. 
Allows easier reviews of webrtcvideoengine2.cc and landing the new video
API on shorter review cycles.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 16:14:51 +00:00
buildbot@webrtc.org
80132e4d70 (Auto)update libjingle 75610402-> 75610402 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 15:24:15 +00:00
kjellander@webrtc.org
595b23c66f Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." 
Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.

> Enable ipv6 by default for webrtc under a Finch experiment.
> 
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
> 
> Review URL: https://webrtc-codereview.appspot.com/23529005

TBR=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 08:58:22 +00:00
andrew@webrtc.org
6ae5a6d7fe Add a target for the approved subset of rtc_base. 
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)

This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.

The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.

BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 01:03:29 +00:00
glaznev@webrtc.org
996784548d HW video decoding optimization to better support HD resolution: 
- Change hw video decoder wrapper to allow to feed multiple input
and query for an output every 10 ms.
- Add an option to decode video frame into an Android surface object. Create
shared with video renderer EGL context and external texture on
video decoder thread.
- Support external texture rendering in Android renderer.
- Support TextureVideoFrame in Java and use it to pass texture from video decoder
to renderer.
- Fix HW encoder and decoder detection code to avoid query codec capabilities
from sw codecs.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 17:52:42 +00:00
guoweis@webrtc.org
cd309e3168 Enable ipv6 by default for webrtc under a Finch experiment. 
BUG=413437 (chromium)
https://code.google.com/p/chromium/issues/detail?id=413437

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 16:31:13 +00:00
pbos@webrtc.org
000d86792d Make BW checks > 0 in peerconnection_unittest.cc. 
These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.

R=stefan@webrtc.org
TBR=jiayl@webrtc.org, solenberg@webrtc.org
BUG=3817,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/29479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 14:38:07 +00:00
henrike@webrtc.org
7f826350e3 Stop building talk/xmllite since it is no longer used. 
BUG=3379
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 08:13:36 +00:00
buildbot@webrtc.org
a42a3ade54 (Auto)update libjingle 75390072-> 75428737 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-13 01:09:18 +00:00
fbarchard@google.com
7e31197cb2 Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." 
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*

> Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
> 
> Breaks other repos.
> 
> TBR=fbarchard@google.com
> BUG=N/A
> 
> Review URL: https://webrtc-codereview.appspot.com/23639004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-13 00:52:42 +00:00
glaznev@webrtc.org
192a54ff2f Temporary revert maximum video codec resolution back to 1080p. 
BUG=3757, 3738
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:40:35 +00:00
henrike@webrtc.org
3decd9b776 Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that." 
Breaks other repos.

TBR=fbarchard@google.com
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/23639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:31:29 +00:00
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
buildbot@webrtc.org
9eabe5e912 (Auto)update libjingle 72931377-> 72931377 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6858 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 22:48:28 +00:00
mallinath@webrtc.org
2d60c5e8bc Encoding and Decoding of TCP candidates as defined in RFC 6544. 
R=juberti@chromium.org, jiayl@webrtc.org, juberti@webrtc.org
BUG=2204

Review URL: https://webrtc-codereview.appspot.com/21479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6857 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-08 22:29:20 +00:00
buildbot@webrtc.org
53df88c1bc (Auto)update libjingle 72847605-> 72850595 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6855 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:46:01 +00:00
buildbot@webrtc.org
65b98d12c3 (Auto)update libjingle 72839629-> 72847605 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 22:09:08 +00:00
tkchin@webrtc.org
c8554be6dd Support for TURN/TLS. 
Wrap the socket in an SSL adapter, then simply call StartSSL() on the
SSLAdapter instance.

Cloned from: https://webrtc-codereview.appspot.com/21799004/

R=juberti@chromium.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/14059004

Patch from Manish Jethani <manish.jethani@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:39:08 +00:00
tkchin@webrtc.org
cb46de24fb Add new OWNERS file to talk/examples. 
R=juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/15039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 20:01:34 +00:00
buildbot@webrtc.org
5b1ebacca2 (Auto)update libjingle 72820109-> 72822008 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 17:18:00 +00:00
buildbot@webrtc.org
d509678a4e (Auto)update libjingle 72819313-> 72820109 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:57:07 +00:00
buildbot@webrtc.org
94b996cc18 (Auto)update libjingle 72785516-> 72819313 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6848 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 16:47:14 +00:00
buildbot@webrtc.org
476efa2031 (Auto)update libjingle 72785180-> 72785516 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 04:55:21 +00:00
buildbot@webrtc.org
4f0d401fae (Auto)update libjingle 72682155-> 72785180 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6841 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 04:47:36 +00:00
jiayl@webrtc.org
56d8e05238 A followup to r6828 to fix a condition check in mediasession.cc. 
BUG=2395
R=juberti@chromium.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6832 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 23:52:36 +00:00
buildbot@webrtc.org
624a504f5b (Auto)update libjingle 72659510-> 72673987 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 22:13:05 +00:00
jiayl@webrtc.org
e7d47a1473 Maintain the order of the m-lines in CreateOffer and CreateAnswer. 
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.

BUG=2395
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-05 19:19:05 +00:00
buildbot@webrtc.org
8e885990ae (Auto)update libjingle 72566057-> 72591796 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6824 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 23:56:14 +00:00
jiayl@webrtc.org
b18bf5e47d Adds the support of RTCOfferOptions for PeerConnectionInterface::CreateOffer. 
Constraints are still supported for CreateOffer, but converted to RTCOfferOptions internally.

BUG=3282
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6822 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 18:34:16 +00:00
buildbot@webrtc.org
a27342b7af (Auto)update libjingle 72446860-> 72550257 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6818 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-04 15:22:32 +00:00
buildbot@webrtc.org
e0d03f13e4 (Auto)update libjingle 72443101-> 72446860 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 03:04:01 +00:00
buildbot@webrtc.org
6e203d50a3 (Auto)update libjingle 72442050-> 72443101 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 01:13:04 +00:00
buildbot@webrtc.org
52148c2f74 (Auto)update libjingle 72430895-> 72442050 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-02 00:56:56 +00:00
buildbot@webrtc.org
7cb60ccae1 (Auto)update libjingle 72407428-> 72430895 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6812 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 22:03:36 +00:00
buildbot@webrtc.org
3bc48247b7 (Auto)update libjingle 72403605-> 72407428 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6811 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 16:53:32 +00:00
buildbot@webrtc.org
6955213eca (Auto)update libjingle 72389720-> 72403605 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 15:52:45 +00:00
solenberg@webrtc.org
42d65ce8d7 Fix memory leak in FakeSSLCertificate::GetChain(), discovered by Linux Memcheck build/try bots. 
TBR=hellner
BUG=

Review URL: https://webrtc-codereview.appspot.com/18969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6809 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 10:07:46 +00:00
buildbot@webrtc.org
1a678c61f1 (Auto)update libjingle 72320533-> 72380285 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6808 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-01 06:21:50 +00:00
buildbot@webrtc.org
6b21b71068 (Auto)update libjingle 72205295-> 72320533 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6806 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-31 15:08:53 +00:00
henrike@webrtc.org
d9843da9ee libjingle: stop building files in talk/base as they are no longer used as of r6799 
BUG=3379
R=thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-30 04:00:52 +00:00
fbarchard@google.com
48305f5f4c Disable warning 4702 which affects map, xlist and others on vs2012 and vs2013. 
BUG=3584
TESTED=python webrtc\build\gyp_webrtc -G msvs_version=2013 & ninja -C out\Release
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6801 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-30 00:16:20 +00:00
buildbot@webrtc.org
d4e598d57a (Auto)update libjingle 72097588-> 72159069 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-29 17:36:52 +00:00
buildbot@webrtc.org
51c5508bf1 (Auto)update libjingle 72016417-> 72097588 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6792 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-28 22:26:15 +00:00
pbos@webrtc.org
8aed945842 Remove a disabled test. 
ConstrainsSetCodecsAccordingToEncoderConfig has been removed from
webrtcvideoengine_unittest.cc, removing this one as well.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6789 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-26 10:16:49 +00:00
sergeyu@chromium.org
af9e7943d1 Fix compilation on windows with clang, indentation cleanups 
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6779

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6786 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 19:42:19 +00:00
pbos@webrtc.org
257e130a16 Set NACK/REMB when setting receive codecs. 
Enabling an additional test to ensure that REMB can be both enabled and
disabled by setting VideoCodecs.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6785 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 19:01:32 +00:00
henrike@webrtc.org
185636cf70 Revert of 6778 "Refactor StatsCollector and associated types." 
Breakes FYI bots.

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

Review URL: https://webrtc-codereview.appspot.com/21969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6783 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 18:44:42 +00:00
sergeyu@chromium.org
2386882266 Revert "Fix compilation on windows with clang, indentation cleanups" 
This reverts commit f628eaedfeea97e13c63c78dd42f2b1c76723619.

TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/13069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6780 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:37:12 +00:00
sergeyu@chromium.org
a44fce5920 Fix compilation on windows with clang, indentation cleanups 
R=henrike@webrtc.org, thakis@chromium.org
TBR=hellner@chromium.org

Review URL: https://webrtc-codereview.appspot.com/18849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6779 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 17:28:25 +00:00
tommi@webrtc.org
190d269c0f 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 the original attempt in r6747)

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-25 10:32:30 +00:00
jiayl@webrtc.org
06b04ec4ab Fix a crash in statscollector.cc caused by invoking methods on the worker thread which destroys the Transport. 
BUG=3579
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-24 20:41:20 +00:00
buildbot@webrtc.org
45304ff0a7 (Auto)update libjingle 71829282-> 71834788 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6773 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-24 16:06:35 +00:00
henrike@webrtc.org
39f831fbb0 Re-revert of 6747 "Refactor StatsCollector and associated types." 
Breakes FYI bots.

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

Review URL: https://webrtc-codereview.appspot.com/13979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-24 14:20:52 +00:00
buildbot@webrtc.org
437d57db5b (Auto)update libjingle 71775619-> 71778545 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6771 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 21:40:28 +00:00
henrike@webrtc.org
8c7e3291a9 Revert 6747 "Refactor StatsCollector and associated types." 
Breakes FYI bots.

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

Review URL: https://webrtc-codereview.appspot.com/18889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 21:38:58 +00:00
henrike@webrtc.org
8721f989bf Revert 6766 "Temporarily add a default ctor to StatsReport and make |id| non const. As soon as I've updated the chrome side, I'll revert this cl." 
BUG=N/A
TBR=ajm@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6769 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 21:38:09 +00:00
buildbot@webrtc.org
e2da234e27 (Auto)update libjingle 71766184-> 71775619 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6768 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 21:09:01 +00:00
buildbot@webrtc.org
21b4da8ebd (Auto)update libjingle 71753329-> 71766184 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6767 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 19:07:53 +00:00
tommi@webrtc.org
0f7328cd6b Temporarily add a default ctor to StatsReport and make |id| non const. 
As soon as I've updated the chrome side, I'll revert this cl.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/16149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 16:31:57 +00:00
pbos@webrtc.org
9359cb3e75 Enable SendAndReceive tests. 
Also fixes a crash in ::SetCapturer which wasn't exposed by tests
before.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18019005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6765 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 15:44:48 +00:00
pbos@webrtc.org
5ff71ab4b3 Revert "(Auto)update libjingle 71675033-> 71726409" 
This reverts commit r6761 which looks like an accidental auto-revert of
r6760.

BUG=1788
TBR=wu@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:28:56 +00:00
buildbot@webrtc.org
89c833cd9d (Auto)update libjingle 71726409-> 71726772 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6762 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:11:58 +00:00
buildbot@webrtc.org
f67f6aa741 (Auto)update libjingle 71675033-> 71726409 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:04:22 +00:00
pbos@webrtc.org
8120353342 Implement suspend-below-min-bitrate option. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6760 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:04:08 +00:00
pbos@webrtc.org
543e589205 Wire up VideoOptions for payload-based padding. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6759 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-23 07:01:31 +00:00
glaznev@webrtc.org
efe4b9af49 Add VP8 video decoding hw acceleration support to Java Peerconnection library. 
For now NVidia decoder is supported only,
Qualcomm will be added once b/16353967 is fixed.

TODO:
- Support queuing 2-3 decoder input buffers.
- Add average decoding time statistics.
- Add Qualcomm hw decoder support.

BUG=3030
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6758 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 17:44:53 +00:00
pbos@webrtc.org
6f48f1bf68 Implement encoder options in WebRtcVideoEngine2. 
Implementing default options to enable denoising by default and wiring
up encoder settings to propagate VP8 settings.

BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 16:29:54 +00:00
pbos@webrtc.org
cadd078cf9 Remove unused config.h and math.h includes. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6756 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 15:26:09 +00:00
pbos@webrtc.org
85f42949d6 Enable ReceiveStreamReceivingByDefault test. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6754 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-22 09:14:58 +00:00
buildbot@webrtc.org
fa5fcd671d (Auto)update libjingle 71599033-> 71605904 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6751 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 21:55:43 +00:00
buildbot@webrtc.org
e69b061926 (Auto)update libjingle 71575585-> 71599033 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 20:38:58 +00:00
tommi@webrtc.org
908f57ed94 Disable GetStatsForInvalidTrack while I rewrite it. 
TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 11:44:39 +00:00
tommi@webrtc.org
756b8462eb 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.

R=xians@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=6745

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 11:24:17 +00:00
tommi@webrtc.org
fd61a1d693 Revert 6745 "Refactor StatsCollector and associated types." 
Broke build on android.

> 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.
> 
> R=xians@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/18819004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 11:05:28 +00:00
tommi@webrtc.org
647e05cfcd 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.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-21 10:55:11 +00:00
pbos@webrtc.org
3c10758b3b Check before send/receive rtp header extensions. 
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13949004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-20 15:27:35 +00:00
pbos@webrtc.org
8fdeee6abf Implement Base::ConstrainNewCodec2. 
BUG=1788
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6743 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-20 14:40:23 +00:00
jiayl@webrtc.org
3edbaaf337 Ignore empty data in DataChannel::Send to match FF's behavior. 
BUG=crbug/395205
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6742 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 23:57:50 +00:00
buildbot@webrtc.org
99f6308a2d (Auto)update libjingle 71460499-> 71464449 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 23:31:30 +00:00
jiayl@webrtc.org
a0b929b63c Revert "Reland r6707 with the fix for callclient.cc." 
Breaking pulse build again.
This reverts commit 3e0bb9b5bf7f616000399e24f1d9622ad6b612f9.

TBR=wu@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/17979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6740 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 22:28:36 +00:00
buildbot@webrtc.org
196ae6d667 (Auto)update libjingle 71456344-> 71456420 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:41:41 +00:00
buildbot@webrtc.org
3dec81a736 (Auto)update libjingle 71456173-> 71456344 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6738 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:39:56 +00:00
jiayl@webrtc.org
a6e8cf8fb7 Reland r6707 with the fix for callclient.cc. 
TBR=mallinath@webrtc.org
BUG=3310

Review URL: https://webrtc-codereview.appspot.com/13039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:34:11 +00:00
buildbot@webrtc.org
60e65b11c1 (Auto)update libjingle 71452608-> 71453580 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6735 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 21:07:50 +00:00
jiayl@webrtc.org
8636fc852e Creates the default track if the remote media content is send-only and there is no stream in the SDP. 
BUG=2628
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 20:54:27 +00:00
pbos@webrtc.org
e6f84ae8a6 Initial WebRtcVideoEngine2::GetStats(). 
Also forward-declaring and moving WebRtcVideoRenderer out of header.

BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 11:11:55 +00:00
pbos@webrtc.org
d1ea06b3d5 Restart VideoReceiveStreams in WebRtcVideoEngine2. 
Puts VideoReceiveStreams in a wrapper, WebRtcVideoReceiveStream that
contain their state (configs). WebRtcVideoRenderer (the wrapper between
webrtc::VideoRenderer and cricket::VideoRenderer) has also been merged
into WebRtcVideoReceiveStream.

Implements and tests setting codecs with new FEC settings as well as RTP
header extensions on already existing receive streams.

BUG=1788
R=pthatcher@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6727 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 09:35:58 +00:00
buildbot@webrtc.org
c31651d847 (Auto)update libjingle 71378257-> 71410012 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-18 08:22:39 +00:00
mallinath@webrtc.org
aa93611375 Connect to the turn server if address cannot be resolved by the browser by using 
unresolved address. This case is only considered for TCP sockets. P2P layer will
assume socket will do the resolve by using a proxy.

BUG=3384
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6722 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 21:55:04 +00:00
mallinath@webrtc.org
e5995aadd5 Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth. 
This priority will be used in calculating the candidate priority generated from the server. This will allow candidate generated from server to have unique priority.

BUG=3223
R=jiayl@webrtc.org, juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 18:23:52 +00:00
jiayl@webrtc.org
e10d28cf14 fix 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6720 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 17:07:49 +00:00
pbos@webrtc.org
5301b0f1fc Move additional state into WebRtcVideoSendStream. 
Prevents having two places where codecs etc. are set up and allows us to
avoid creating the underlying VideoSendStream before send codecs are
set up.

BUG=1788
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 08:51:46 +00:00
wu@webrtc.org
52eddec71b Revert 6707 "Add support of multiple STUN servers in UDPPort." 
Reason:
Breaks the build on callclient.cc.

> Add support of multiple STUN servers in UDPPort.
> Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.
> 
> I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.
> 
> BUG=3310
> R=mallinath@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/13879004

TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-17 00:03:24 +00:00
wu@webrtc.org
4c3e9917e7 Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be: 
m=  (media name and transport address)
  i=* (media title)
  c=* (connection information -- optional if included at
       session level)
  b=* (zero or more bandwidth information lines)
  k=* (encryption key)
  a=* (zero or more media attribute lines)

BUG=2260
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 21:03:13 +00:00
jiayl@webrtc.org
46fb331bc5 Add support of multiple STUN servers in UDPPort. 
Now UDPPort signals PortComplete or PortError when the Bind requests for all STUN servers are responded or failed. If any STUN bind is successful, PortComplete is signaled; otherwise, PortError is signaled.

I discovered a bug in SocketAddress while working on this. It didn't consider two addresses unequal if they have unresolved IP and different hosts. It's fixed now.

BUG=3310
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 20:55:31 +00:00
buildbot@webrtc.org
a8d8ad2be6 (Auto)update libjingle 71240799-> 71250251 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6705 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 14:23:08 +00:00
pbos@webrtc.org
38ce7d03d8 Implement unittest for SetSendCodecsChangesExistingStreams. 
BUG=1788
R=pbos@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19869004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 08:01:38 +00:00
tommi@webrtc.org
47218956fc Minor refactoring of StatsCollector. 
* Make GetTimeNow a static method in the cc file.
* Make GetTransportIdFromProxy a static method as well and not a class method.

The second change is in preparation of removing the proxy_to_transport_ member variable which isn't needed and is just a copy from the session stats.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6696 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 19:22:37 +00:00
tkchin@webrtc.org
42fe4350fe Remove Thread::RunningForChannelManager(). 
I haven't heard of this failing, so it should be safe to remove. Let me know if this isn't the case.

BUG=3388
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6695 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-15 17:52:43 +00:00