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Author SHA1 Message Date
pbos@webrtc.org
f3e4ceee47 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ 
BUG=163
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1904005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4444 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-31 15:17:19 +00:00
tnakamura@webrtc.org
64e2cbf184 clean up incomplete revert in r4357 
Also revert r4319, will follow up with pbos

Reason for recent series of reverts: video freezes when testing with packet loss

R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1817004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4359 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 21:52:59 +00:00
tnakamura@webrtc.org
aa4d96a134 Revert r4301 
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 19:25:04 +00:00
pbos@webrtc.org
9b82dced8d Make sure first RTP packet counts as in-order. 
BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1811004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4350 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-16 13:03:35 +00:00
elham@webrtc.org
4a44ea21d7 Revert r4320 "Fix three uninitialized members in rtp_receiver_impl.cc" 
TBR=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1803004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4346 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:46:06 +00:00
elham@webrtc.org
4888fd4827 Revert r4321 "Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered" 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4345 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:21:48 +00:00
elham@webrtc.org
b7eda43810 Revert r4322 "Support sending multiple report blocks and keeping track of statistics on 
several SSRCs"

R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4344 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 21:08:27 +00:00
elham@webrtc.org
6f5707e184 Revert r4328 
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1774005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4343 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-15 20:59:52 +00:00
stefan@webrtc.org
e4736eee20 Fixes a crash when sending SR reports from a sender only module. 
BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 08:28:35 +00:00
braveyao@webrtc.org
aeba6e8740 ModuleRTPRTCP call rtcp_sender_.TMMBR() directly instead of calling its own API. 
BUG=2051
TEST=autotest
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1790005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-11 08:06:37 +00:00
pbos@webrtc.org
96edd56170 Sorted headers under rtp_rtcp/. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 15:40:42 +00:00
stefan@webrtc.org
717d147ebb Support sending multiple report blocks and keeping track of statistics on several SSRCs. 
BUG=1811
TEST=vie_auto_test --automated, voe_auto_test --automated, trybots
R=andresp@webrtc.org, tommi@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1768004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4322 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 13:39:27 +00:00
stefan@webrtc.org
9de89a6f6b Fix uninitialized value warning in rtp_payload_registry and make sure we return an error if the payload type isn't registered. 
R=pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1782004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4321 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 12:42:15 +00:00
stefan@webrtc.org
452d853c43 Fix three uninitialized members in rtp_receiver_impl.cc. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1781004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4320 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 10:54:56 +00:00
pbos@webrtc.org
08933a5dfb Initialize payload-type frequency in channel.cc. 
Uninitialized values triggered divide-by-zero crashes in voe_auto_test.

BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1780004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4319 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-10 10:06:29 +00:00
stefan@webrtc.org
f56d612c70 Create gyp target for bwe components. 
R=henrikg@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1775004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4311 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-09 12:32:35 +00:00
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing 
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 21:31:18 +00:00
stefan@webrtc.org
66b2e5c05a Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the 
rtp_rtcp implementation.

This refactoring significantly reduces the receive-side RTP parser and receiver
complexity, and makes it possible to implement RTX correctly by having two
instances of receive-statistics.

With this change the dead-or-alive and packet timeout APIs are removed.

TEST=trybots, vie_auto_test, voe_auto_test
BUG=1811
R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1745004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-05 14:30:48 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
solenberg@webrtc.org
a5fd2f1348 Do basic parsing of RTCP headers in PcapFileReader to enable log filtering. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1697004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-26 08:36:07 +00:00
solenberg@webrtc.org
91811e2b04 Remove unused multi stream bandwidth estimator. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1712004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 20:36:14 +00:00
stefan@webrtc.org
a4c5abb52a Make sure padding packets are sent. 
BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 15:46:16 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing. 
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 14:13:42 +00:00
stefan@webrtc.org
508a84b255 Wire up pacer-based padding. 
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.

Padding will for now only be generated by the first sending RTP module.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 12:53:37 +00:00
kjellander@webrtc.org
63e988856e Merge more tests into modules_{unit,integration}tests. 
A new test target named 'modules_integrationtests' is created
and the following test targets were merged into it:
* audio_coding_module_test
* test_fec
* video_coding_integrationtests
* vp8_integrationtests

A couple of other targets were merged into modules_unittests:
* audio_coding_unittests
* audioproc_unittest
* common_unittests
* video_coding_unittests
* video_processing_unittests
* vp8_unittests

I wasn't able to merge audio_decoder_unittests and neteq_unittests due to
conflicts with different defines in these tests.

Some tests that have special requirements aren't merged into
modules_integrationtests yet. I took the opportunity to rename them
since the bot configs will need to be update anyway:
* audio_device_test_api -> audio_device_integrationtests
* video_capture_module_test -> video_capture_integrationtests
* video_render_module_test -> video_render_integrationtests

Exclude files were added for modules_integrationtests to make sure
the memcheck and tsan bots doesn't tests that are too slow
(audio_coding_module_test and vp8_integrationtests were previously
disabled on those bots).

Suppressions for AudioCodingModuleTest needed to be added to get
modules_integrationtests to pass memcheck (even if the test is
excluded from execution).

BUG=1843
TEST=local execution on Linux and trybots (passing except the merged tests of course)
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1656004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-14 20:09:44 +00:00
kjellander@webrtc.org
04996cd5e5 Fix breakage due to test_fec conversion to gtest. 
In my attempt to commit a subset of http://review.webrtc.org/1647005/
instead of all of it, I forgot to add the gtest dependency to the
test_fec.gypi. This CL fixes that.

TEST=local compile + win_rel,mac_rel,linux_rel trybots
BUG=1916
R=marpan
TBR=marpan

Review URL: https://webrtc-codereview.appspot.com/1655004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 12:15:33 +00:00
kjellander@webrtc.org
22bbbdfa68 Convert test_fec to gtest 
All tests needs to be gtest tests in order to be executed
with the upcoming isolate/swarm framework.

TEST=trybots passing
BUG=1916
R=andrew@webrtc.org, marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/1647005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4218 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-12 11:55:05 +00:00
kjellander@webrtc.org
6c35e0b0f7 Reorganize test targets in WebRTC 
This CL will lower the number of test targets in WebRTC by:

Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006):
* resampler_unittests
* signal_processing_unittests
* vad_unittests

Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests

Merge into test_support_unittests:
* channel_transport_unittests

channel_transport.gyp was also removed in favor for test.gyp.

I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.

Buildbot configuration update will be synced with the commit of this CL.

TEST=trybots
BUG=1843
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-11 08:29:17 +00:00
stefan@webrtc.org
a817962bab Refactor padding and rtp header functionality. 
BUG=1837
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1611004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 13:47:36 +00:00
wu@webrtc.org
fa64a595ad Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. 
This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later.

BUG=1828
TEST=unit tests
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1598005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 21:27:57 +00:00
pbos@webrtc.org
a048d7cb0a Include files from webrtc/.. paths in rtp_rtcp/ 
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1557004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 14:27:38 +00:00
stefan@webrtc.org
a5cb98cbbd Breaking out RTP header parsing from the RTP module. 
This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video.

Moving bandwidth estimation before the RTP module is also required for RTX.

TEST=vie_auto_test, voe_auto_test, trybots.
BUG=1811
R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1545004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-29 12:12:51 +00:00
marpan@webrtc.org
a6ae644e52 Add comment about test_packet_masks_metrics. 
R=andrew@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1577004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-28 17:42:12 +00:00
pbos@webrtc.org
8665da8926 Remove dead testRateControl.cc 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1556004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4101 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 13:29:29 +00:00
pbos@webrtc.org
a01f7f6509 Removed dead testH263Parser.cc 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1555004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 13:01:57 +00:00
pbos@webrtc.org
c1f0eb2c03 Remove dead bitstreamTest.cc. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1553004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-24 12:46:08 +00:00
stefan@webrtc.org
c74c3c2447 Adds integration test for RTX and fixes bugs found. 
BUG=1811
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 13:48:22 +00:00
stefan@webrtc.org
5c58f63d3f Fix regression where retransmission bitrate is no longer estimated. 
BUG=1813
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1530004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-23 13:36:55 +00:00
solenberg@webrtc.org
c0352d566a Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. 
BUG=
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1510004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-20 20:55:07 +00:00
phoglund@webrtc.org
9919ad5caf Formatted FEC stuff. 
Unfortunately I had to pull in quite a bit of stuff due to use of unencapsulated public member variables.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1401004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 15:06:28 +00:00
solenberg@webrtc.org
7ebbea14a9 Add handling of the absolute send time header extension to the rtp_rtcp module. 
BUG=
R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1480004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-16 11:10:31 +00:00
mikhal@webrtc.org
6cfa3907c8 Updating NACK RTX test 
BUG=1513
R=holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1274006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-15 20:17:43 +00:00
andresp@webrtc.org
29b2219914 Adding a factory to remote bitrate estimator and allow it to be set via config. 
Additionally:
 - clean api to set remote bitrate estimator mode.
 - clean api to set over use detector options.

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

Review URL: https://webrtc-codereview.appspot.com/1448006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 12:10:58 +00:00
phoglund@webrtc.org
527f6c62fc Reformatted FEC tables. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1400004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 09:25:01 +00:00
justinlin@chromium.org
7bfb3a3227 Add more tracing for key frames. 
R=mallinath@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1428004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-13 22:59:00 +00:00
phoglund@webrtc.org
315d39866e Formatted dtmf_queue. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1398004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-08 10:04:06 +00:00
pbos@webrtc.org
3004c79c6a Fix clang errors in non-GYP_DEFINES=clang=1 build 
BUG=1623
R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1368004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 12:36:21 +00:00
pbos@webrtc.org
77f6b2175e Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." 
> Revert 3933 "Remove traces of deprecated WebRtc_Word types."
> 
> > Remove traces of deprecated WebRtc_Word types.
> > 
> > BUG=314
> > R=tommi@webrtc.org
> > 
> > Review URL: https://webrtc-codereview.appspot.com/1385004
> 
> TBR=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1386004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1397004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-03 12:02:11 +00:00
pwestin@webrtc.org
52b4e8871a Adding trace and changing pacing constants 
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-02 19:02:17 +00:00