1094 Commits

Author SHA1 Message Date
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
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
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
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
wuchengli@chromium.org
0d94c2f81c Avoid acquiring VCM::_receiveCritSect during decode callback.
When VideoDecoder::Decode, Reset, or Release is called,
VideoCodingModuleImpl::_receiveCritSect may have been
acquired. Decode callback needs to acquire the same lock
in ViEChannel::FrameToRender. It is not a problem for
SW decode because decode callback is run on the same
WebRTC decoding thread and the lock is re-entrant. But
for HW decode, decode callback is run on a thread different
from WebRTC decoding thread. Decode callback gets the locks
in the opposite order. Deadlock can happen.

BUG=http://crbug.com/170345
TEST=Try apprtc.appspot.com/?debug=loopback on ARM Chromebook Daisy.
     Run libjingle_peerconnection_unittest.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1997005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4523 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 14:20:49 +00:00
pbos@webrtc.org
9668467d87 Run loopback tests with network thread.
Running with a network thread provides a more realistic simulation. Like
a real network, packets are handed off to a socket, or buffer, and then
the call returns. This prevents weird scenarios when both the sending
side and receiving side are on the call stack simultaneously, which can
cause deadlocks as locks could otherwise be taken simultaneously in both
the sender and receiver order by the same thread.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4522 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 12:59:04 +00:00
minyue@webrtc.org
ecbe0aa543 Added Opus stereo support
TESTED=git try
BUG=webrtc:1360
R=tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1868004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4521 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 06:48:09 +00:00
sergeyu@chromium.org
bf853f2732 Fix crash in screen capturer on Mac
BUG=crbug.com/247685
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2006004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4518 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-10 01:30:23 +00:00
pbos@webrtc.org
6cd9341801 Hand over loopback packets to a network thread.
This version of LoopBackTransport hands packets over to a network thread
which will deliver them instead. This allows SendRTP and SendRTCP to
always be able to return, preventing deadlocks in voe_auto_test. The
previous case did not represent actual network usage. Now the send and
receive side can run concurrently with the receiving side. Previously
the sender thread also drove the receiving side, which does not
represent the regular use case where packets are put on a network
socket.

BUG=1568,2081,2178
TEST=Ran VoiceEngine RtpRtcpTest.*, known for deadlocking, 100+ times.
R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1985005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4516 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 21:11:57 +00:00
stefan@webrtc.org
80865fd611 Don't pace out packets or generate padding when the pacer is disabled.
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4513 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 11:31:23 +00:00
pbos@webrtc.org
2ab209ef14 Remove include_dirs from test/test.gyp.
This is a cleanup step for having root-relative includes, include_dirs shouldn't be needed anymore.

BUG=1662
R=phoglund@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1984004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4512 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 08:49:48 +00:00
pbos@webrtc.org
a3b7406219 Remove unused unreferenced code in webrtc/
The code removed here are .c, .cc and .h files that are not referenced
from anywhere else. E.g. if git-grep showed no occurrence of the file
it's removed. This process was repeated until no more unreferenced
files were present.

BUG=
R=andrew@webrtc.org, henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, turaj@webrtc.org, wu@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1945004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4511 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 08:47:51 +00:00
wuchengli@chromium.org
f4081ab8d8 Revert "Avoid acquiring VCM::_receiveCritSect during decode callback."
This reverts commit aa3528a9cd65b176b9d6f9d58cecb1068891dca4.

BUG=http://crbug.com/170345
TEST=libjingle_peerconnection_unittest
TBR=stefan,wu

Review URL: https://webrtc-codereview.appspot.com/1999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4510 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 04:42:51 +00:00
wuchengli@chromium.org
a717ee9962 Avoid acquiring VCM::_receiveCritSect during decode callback.
When VideoDecoder::Decode, Reset, or Release is called,
VideoCodingModuleImpl::_receiveCritSect may have been
acquired. Decode callback needs to acquire the same lock
in ViEChannel::FrameToRender. It is not a problem for
SW decode because decode callback is run on the same
WebRTC decoding thread and the lock is re-entrant. But
for HW decode, decode callback is run on a thread different
from WebRTC decoding thread. Decode callback gets the locks
in the opposite order. Deadlock can happen.

BUG=http://crbug.com/170345
TEST=Try apprtc.appspot.com/?debug=loopback on ARM Chromebook Daisy.
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1977004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4509 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 04:08:38 +00:00
mikhal@webrtc.org
64799da6c6 Allowing decoding with errors, when disabling nack.
BUG=1897
R=stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1982004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4508 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 22:45:33 +00:00
niklas.enbom@webrtc.org
e270331481 Fix duplicate code
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1993004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4507 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 22:23:48 +00:00
pbos@webrtc.org
58d76cb635 Delete Channels without ChannelManager lock.
Triggered Helgrind error, as deleting a Channel will also unregister a
module which has called GetChannel(), resulting in a cyclic lock graph.
This change will also allow other threads to access the ChannelManager
instance while Channels are deleted.

R=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1946005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4505 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 17:32:21 +00:00
tina.legrand@webrtc.org
bd21fb5f8d Adding call to Opus PLC
NetEq will call the PLC function in Opus only to set the decoder state. The actual PLC data will not be used.

BUG=https://code.google.com/p/webrtc/issues/detail?id=1181
R=tterribe@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1727004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4504 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 11:01:07 +00:00
agalusza@google.com
d177c10e2d Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests.
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1943004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4503 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-08 01:12:33 +00:00
pbos@webrtc.org
676ff1ed89 Ref-counted rewrite of ChannelManager.
The complexity of the last ChannelManager and potentially usage of it as well caused race conditions and deadlocks in loopback voe_auto_test. This ref-counted solution takes no long-term locks, uses less locks overall and is significantly easier to understand.

ScopedChannel has been split up into a ChannelOwner with a reference to a channel and an Iterator over ChannelManager. Previous code was really used for both things. ChannelOwner is used as a shared pointer to a channel object, while an Iterator should work as expected.

BUG=2081
R=tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1802004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4502 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-07 17:57:36 +00:00
pbos@webrtc.org
a165d9c0a4 Code formatting on files touched in r4447.
BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4500 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-07 14:17:05 +00:00
pwestin@webrtc.org
401ef361ac Added configuration of max delay to ACM and NetEq
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1964004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4499 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 21:01:36 +00:00
agalusza@google.com
c4e1ab515b Added Decoding with errors API to video_coding.h and removed unused DecodeError enum.
R=marpan@google.com, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1937004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4497 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 18:27:41 +00:00
turaj@webrtc.org
0fc2558503 Add turaj@webrtc.org to NetEq owners.
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1980004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4496 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 17:07:18 +00:00
phoglund@webrtc.org
94aca5c7de Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer.
TBR=xians@webrtc.org
BUG=2179

Review URL: https://webrtc-codereview.appspot.com/1955005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4495 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 08:20:47 +00:00
phoglund@webrtc.org
bd69d1beaf Disabled SsrcPropagatesCorrectly on Linux.
BUG=2178
TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1975004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4494 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 08:03:16 +00:00
minyue@webrtc.org
7bb5436e5d Better error treatment in NetEqImpl::InsertPacketInternal()
BUG=webrtc:1364
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1844004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4493 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 05:40:57 +00:00
minyue@webrtc.org
9721db799c removed NetEq::EnableDtmf()
BUG=webrtc:1373
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1822005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4492 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 05:36:26 +00:00
wu@webrtc.org
9dba525627 * Update libjingle to 50389769.
* Together with "Add texture support for i420 video frame." from
wuchengli@chromium.org.
https://webrtc-codereview.appspot.com/1413004

RISK=P1
TESTED=try bots
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1967004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 20:36:57 +00:00
fischman@webrtc.org
f696f253b2 Invert dependency between webrtc_utility and media_file targets to reflect reality.
BUG=2166
R=henrike@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1953004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4488 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 18:45:19 +00:00
elham@webrtc.org
9b8861c358 Updated WebRTC version number to 3.38
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1965004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4487 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 17:19:16 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +00:00
pbos@webrtc.org
ccdcbae177 Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1963004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4484 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 13:25:51 +00:00
pbos@webrtc.org
4052370e89 Use RtpHeaderParser in VideoCall implementation.
BUG=1827
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1962004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4483 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 12:49:22 +00:00
pbos@webrtc.org
bbb07e69e5 Glue code and tests for NACK in new VideoEngine API.
The test works by randomly dropping small bursts of packets until enough
NACKs have been sent back by the receiver. Retransmitted packets are
never dropped in order to assure that all packets are eventually
delivered. When enough NACK packets have been received and all dropped
packets retransmitted, the test waits for the receiving side to send a
number of RTCP packets without NACK lists to assure that the receiving
side stops sending NACKs once packets have been retransmitted.

BUG=2043
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1934004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4482 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 12:01:36 +00:00
pbos@webrtc.org
7fb9ce0cf5 Fix send times in video_full_stack.
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1947004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4481 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 09:29:50 +00:00
pbos@webrtc.org
735a7c8b93 Add back is.FrameProvider() call lost in r4194.
BUG=2119
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1946004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4480 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 09:03:03 +00:00
andrew@webrtc.org
2cbb429323 Remove redundant conditions key.
Gives an error when gyp is run with CHROMIUM_GYP_SYNTAX_CHECK=1.

TBR=henrike

Review URL: https://webrtc-codereview.appspot.com/1952004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4478 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-02 20:52:54 +00:00
turaj@webrtc.org
7df9706a01 Add one API for implementing Initial delay.
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4475 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-02 18:07:13 +00:00