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Author SHA1 Message Date
henrike@webrtc.org
60bdb07a16 Disables ReceivesPliAndRecoversWithNack and NoPacketLoss as they break the bots. 
BUG=2277,2278
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2086004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4586 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 19:55:53 +00:00
henrike@webrtc.org
a0218a84d1 Revert 4582 "Reverts a second set of reverts caused by a bug in ..." 
> Reverts a second set of reverts caused by a bug in a dependency.
> 
> Revert "Revert r4328"
> 
> Revert "Revert r4322 "Support sending multiple report blocks and keeping track
> of statistics on"
> 
> BUG=1811
> R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/2072004

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2087004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4585 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 19:44:13 +00:00
hta@webrtc.org
cc39484770 IP address display from stats. 
This CL demonstrates a couple of methods to extract more complex properties from the stats that are linked via stats IDs.

RISK=P3
TESTED=manual test
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1667005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 17:00:54 +00:00
phoglund@webrtc.org
17018ed39c Added perf summary pages to the dashboard server. 
The purpose is to make the WebRTC performance metrics easier to overview.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2081004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 16:31:12 +00:00
stefan@webrtc.org
1a65d6c36b Reverts a second set of reverts caused by a bug in a dependency. 
Revert "Revert r4328"

Revert "Revert r4322 "Support sending multiple report blocks and keeping track
of statistics on"

BUG=1811
R=henrika@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2072004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 16:22:21 +00:00
pbos@webrtc.org
fbf0f69bf8 Call SetExecutablePath from test_main.cc 
Fixes crash in video_engine_tests on bots, that were unabled to locate
the resource file.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2083004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4581 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 16:00:15 +00:00
pbos@webrtc.org
4c96601aed Make FrameGeneratorCapturer own frame_generator. 
Fixes memleaks where test::FrameGenerator::Create() was used to create
frame_generator, but it was never freed. Since the frame generator
shouldn't be used concurrently it's easiest if FrameGeneratorCapturer
take ownership of the instance.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2047005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4580 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 12:07:37 +00:00
phoglund@webrtc.org
abc1ed37c6 Merging video_full_stack_tests and video_engine_tests. 
The reason is that we want to have as few test targets as possible to simplify bot configuration. It's also more convenient for developers since it will be trivial to introduce more perfing tests.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/2068004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4579 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 12:06:03 +00:00
fischman@webrtc.org
d0f4c2185b iOS: unbreak the build following r4546 
BUG=2255
R=niklas.enbom@webrtc.org, sjlee@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2078004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 22:16:55 +00:00
wu@webrtc.org
ebe68aad44 Fix memory leak in portallocatorsessionproxy_unittest. 
Remove the suppressions that have been fixed.

BUG=1972,2263
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2062005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4576 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 21:14:39 +00:00
kjellander@webrtc.org
cbdb9d1c69 Add comment about updating webrtc.DEPS when rolling gflags 
BUG=none
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2070004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 16:18:35 +00:00
kjellander@webrtc.org
25b39ab1a6 Document updating gflags and remove code duplication. 
When rolling the google-gflags dependency, there might be
a need of updating the generated configuration files. I added
a instructions to the README.webrtc file for doing that.

This CL also removes duplicated configuration headers so we
only separete the ones that differs (Windows and everything
else).

BUG=2251
TEST=none
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2046004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 16:17:10 +00:00
pbos@webrtc.org
119a1ccdca VideoSendStream SSRC test. 
Verifies that the VideoSendStream starts sending the set SSRC over RTP.

BUG=2227
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2074004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 13:14:07 +00:00
pbos@webrtc.org
e6dc38ea9b Lock resources in event_posix.cc. 
Fixes errors reported by Helgrind from event_posix.cc when running video_engine_tests.

BUG=
TEST=helgrind,trybots
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2060004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 09:49:19 +00:00
pbos@webrtc.org
62e5af4425 Use a sourceforge_url for jsoncpp in DEPS. 
BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4571 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-20 09:19:16 +00:00
henrike@webrtc.org
7238e5f708 Fixes broken deps. Jsoncpp has moved from http://jsoncpp.svn.sourceforge.net to http://svn.code.sf.net 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2034005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4570 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 18:55:54 +00:00
pbos@webrtc.org
d5f4c15e8f Added missing static_cast conversion. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2061004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 16:35:36 +00:00
pbos@webrtc.org
e7f056ec45 Implementation and testing of PLI in new API. 
BUG=2174
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2011004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 16:09:34 +00:00
stefan@webrtc.org
d4f607e70a Fixes to padding when driven by encoder. 
- Allow padding to be sent on an ssrc which doesn't produce video, therefore
  never having the last_packet_marker_bit_ set.
- Add the random timestamp offset to all padding packets.
- Store the capture time of padding packets to properly create an offset.

BUG=2258
TEST=trybots and a new test which will be committed separately.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 15:55:01 +00:00
phoglund@webrtc.org
32fe90b3f9 Made all integration tests use consistent naming. 
After decision by pbos@, mflodman@ et. al.

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

Review URL: https://webrtc-codereview.appspot.com/2041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 11:40:19 +00:00
kjellander@webrtc.org
f3bf5e02c8 Add suppressions file for TSan v2 
This is needed for our tests to pass when run under TSan v2.
More details on TSan v2 can be found at
http://www.chromium.org/developers/testing/threadsanitizer-tsan-v2

BUG=chromium:274414
TEST=ran tests locally standing in trunk/ using:
GYP_DEFINES=tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only" gclient runhooks
ninja -C out/Release
TSAN_OPTIONS=suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7 out/Release/testname
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2057004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 05:33:20 +00:00
turaj@webrtc.org
f1efc57139 Implementing APIs to set maximum and minimum for latency. 
cpplint warnning fixed

Ready for review

BUG=
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1971004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 23:44:24 +00:00
agalusza@google.com
b655985abd Added choice of decode error mode to loopback test. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1997004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 23:07:14 +00:00
fischman@webrtc.org
28ff3ee6aa Fix invalid cricket::SrtpStat::FailureKey::operator<() implementation. 
If operator<(a, b) returns true, then it must not be the case that
operator<(b, a) is true as well, but the old implementation would do exactly
that if a={1, 0, 0} and b={0, 0, 1}, for example.

Should fix e.g.:
[004:555] Error(unittest_main.cc:40): c:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\xtree(1746) : Assertion failed: invalid operator<
from http://chromegw/i/client.libjingle/builders/Win32%20Debug/builds/245/steps/libjingle_p2p_unittest/logs/stdio

R=juberti@webrtc.org, mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2054005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4561 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 19:12:26 +00:00
wu@webrtc.org
166991fa1f Suppress tsan errors on libjingle_peerconnection_unittest. 
TBR=mallinath
BUG=1205

Review URL: https://webrtc-codereview.appspot.com/2055004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4560 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 17:57:44 +00:00
wu@webrtc.org
a2e0901e54 Suppress tsan errors. 
BUG=1205,2079
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/2054004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4558 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 17:00:01 +00:00
mallinath@webrtc.org
4d3e8b8c1b Update srtp error value in channel unittests. 
TBR=ronghuawu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2053004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4557 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-16 00:31:17 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
Together with Stefan's http://review.webrtc.org/1960004/.

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2048004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:38:54 +00:00
fischman@webrtc.org
dde7d4c6ed Roll chromium_revision 214260:217707 and gflags 45:84 
gflags roll is needed mostly to pick up fixes for warnings triggered by newer
compiler/settings pulled in by the chromium roll.  Had to switch from the old
google-gflags project the current gflags project to pick up this fix (see
https://code.google.com/p/gflags/source/detail?r=74 for details).

Update android build.xml file to reflect tools moves in new SDK pulled in by the chromium_revision roll.

R=niklas.enbom@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2043004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4555 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 23:31:30 +00:00
niklas.enbom@webrtc.org
cc9238e385 Fix OSX keydown detection. I noticed that the OSX implementation differs from Linux and Windows, and it will trigger continuously for a key that is pressed down. It would totally make sense to change this to a callback driven model, but that's a bigger change. 
I need to test this before committing...

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1996004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4550 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 14:19:12 +00:00
henrike@webrtc.org
c92781737c OpenSl bug: not matching playout and record sample rate led to high or low pitch audio (depending on if playout rate was higher or lower than record rate). 
BUG=N/A
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2031004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 14:13:13 +00:00
kjellander@webrtc.org
4298f73031 Revert 4547 "Isolate GYP target and .isolate files for tests" 
As this breaks the FYI bots in 
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)

> Isolate GYP target and .isolate files for tests
> 
> Implemented according to the instructions at
> http://www.chromium.org/developers/testing/isolated-testing
> 
> Workflow has been like this:
> 1. create _run GYP target
> 2. create a stripped down .isolate file
> 3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
>    actually executing it:
>    tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
>    tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
> 
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
> 
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
> 
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
> 
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
> 
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1673004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 11:29:58 +00:00
kjellander@webrtc.org
d7a4d235d2 Isolate GYP target and .isolate files for tests 
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing

Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
   actually executing it:
   tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
   tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated

All tests that run on the bots for WebRTC has got _run target
and .isolate file created.

"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_integrationtests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests

Tests that requires bare-metal and audio/video devices:
* audio_device_integrationtests
* video_capture_integrationtests

I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_integrationtests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above.
BUG=1916
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1673004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-15 10:02:06 +00:00
sjlee@webrtc.org
d690eab54f The video capture module for iOS. 
This CL is from https://webrtc-codereview.appspot.com/1339004.

Patch this CL, then run the trunk/webrtc/build/vie-webrtc.sh.

BUG=2105
R=fischman@webrtc.org, mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1641004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4546 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 22:07:04 +00:00
pbos@webrtc.org
3d0019f09a Remove ViEBase::Init() call from VideoCall. 
ViEBase::Init() is a no-op in the current implementation. Keeping it
there is just confusing.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4544 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 14:27:11 +00:00
pbos@webrtc.org
fd39e13c80 Remove VideoEngine class from new VideoEngine API. 
The VideoEngine class had minimal use, so it makes more sense to bake
its functionality and config into VideoCall for a simpler API. The only
thing the VideoEngine class could do was to create VideoCalls.

BUG=2224
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2020004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4543 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 13:52:52 +00:00
pbos@webrtc.org
d65914360a Disable CanTransmitExtraRtpPacketsWithoutError on Windows. 
Flakily crashes on Windows.

BUG=2240
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2028005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4542 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-14 09:44:19 +00:00
marpan@webrtc.org
62ecc20afb Revert r4539 "Disable racy part of RunsRtpRtcpTestWithoutErrors". 
Bot failures for Win32-Release and Linux64-Release.

TBR=pbos@webrtc.org.

Review URL: https://webrtc-codereview.appspot.com/2026004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4541 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 21:36:48 +00:00
vikasmarwaha@webrtc.org
83ffb0dd5c Added functionality in apprtc demo to close the capture device on hangup. 
BUG=1589
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2018004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 17:53:37 +00:00
pbos@webrtc.org
a05653b2c1 Disable racy part of RunsRtpRtcpTestWithoutErrors. 
Disabled part as suggested in bug 1790, but without breaking it up into
multiple tests. These tests will be made redundant by tests for the new
API, and it would take far too long to clean these up properly.

BUG=1790
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2022004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 14:27:20 +00:00
wuchengli@chromium.org
e1051b0731 Add native_handle.h to gyp. 
BUG=http://crbug.com/170345
TEST=Build all.
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 05:53:38 +00:00
minyue@webrtc.org
db1cefc14e To allow the propagation of under-run in NetEq. 
BUG=
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1974004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4537 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 01:39:21 +00:00
wu@webrtc.org
97d1a988b6 Remove suppressions for the cases that's already fixed. 
Rename some of the suppressions to new issue.
Fix leaks in virtualsocket_unittest.

BUG=1972,1976,2100
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2010005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4536 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 00:13:26 +00:00
wu@webrtc.org
6603736038 PeerConnection::RemoveStream now removes the local stream even when it's closed. Updated the unit test accordingly. 
RISK=P3
TESTED=PeerConnectionInterfaceTest.CloseAndTestMethods
TBR=fischman_webrtc

Review URL: https://webrtc-codereview.appspot.com/2018005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4535 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-13 00:09:35 +00:00
fischman@webrtc.org
32001ef124 PeerConnection shutdown-time fixes 
- TCPPort::~TCPPort() was leaking incoming_ sockets; now they are deleted.
- PeerConnection::RemoveStream() now removes streams even if the
  PeerConnection::IsClosed().  Previously such streams would never get removed.
- Gave MediaStreamTrackInterface a virtual destructor to ensure deletes on base
  pointers are dispatched virtually.
- VideoTrack.dispose() delegates to super.dispose() (instead of leaking)
- PeerConnection.dispose() now removes streams before disposing of them.
- MediaStream.dispose() now removes tracks before disposing of them.
- VideoCapturer.dispose() only unowned VideoCapturers (mirroring C++ API)
- AppRTCDemo.disconnectAndExit() now correctly .dispose()s its
  VideoSource and PeerConnectionFactory.
- CHECK that Release()d objects are deleted when expected to (i.e. no ref-cycles
  or missing .dispose() calls) in the Java API.
- Create & Return webrtc::Traces at factory birth/death to be able to assert
  that _all_ threads started during the test are collected by the end.
- Name threads attached to the JVM more informatively for debugging.
- Removed a bunch of unnecessary scoped_refptr instances in
  peerconnection_jni.cc whose only job was messing with refcounts.

RISK=P2
TESTED=AppRTCDemo can be ended and restarted just fine instead of crashing on camera unavailability.  No more post-app-exit logcat lines.  PCTest.java now asserts that all threads are collected before exit.

BUG=2183
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2005004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 23:26:21 +00:00
mallinath@webrtc.org
a5506690b4 Update libjingle to 50733053. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2017004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 21:18:15 +00:00
pbos@webrtc.org
4ca7d3f9fe Replace MapWrapper with std::map<>. 
MapWrapper was needed on some platforms where STL wasn't supported, we
now use std::map<> directly.

BUG=2164
TEST=trybots
R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 19:51:57 +00:00
fischman@webrtc.org
dd14b2add1 libjingle gyp: signal errors during gyp time to avoid cryptic failures during build time. 
- $JAVA_HOME / java_home missing or not pointing to a JDK
- Multiple or zero mac codesigning identities

BUG=2206
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2012004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4527 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 18:06:29 +00:00
elham@webrtc.org
1928d0ef67 Updated WebRTC version to 3.39 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2014004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4525 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 17:12:44 +00:00
pbos@webrtc.org
468e19aa93 Signal when shutting down DirectTransport. 
Avoids starting the network thread when there are no packets to be read.
This allows the transport to shut down directly, which makes tests using
it able to quit faster, and not have to wait up to 10ms.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2010004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 14:28:00 +00:00