64 Commits

Author SHA1 Message Date
danilchap
0b4b72797e Use NtpTime in RtcpReceiver instead of pair of uints
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2389703007
Cr-Commit-Position: refs/heads/master@{#14557}
2016-10-06 16:24:51 +00:00
danilchap
28b03eb449 Move RTCPHelp::RTCPReportBlockInformation into RTCPReceiver
removing RTCPHelp namespace and rtcp_receiver_help files,
cleaning style of the ReportBlockInformation usage.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2390643002
Cr-Commit-Position: refs/heads/master@{#14527}
2016-10-05 13:59:51 +00:00
danilchap
7851bda9bc Move RTCPHelp::RTCPReceiveInformation inside RTCPReceiver
move all logic from that class into RTCPReceiver too,
Simplify and fix style on the way.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2373053002
Cr-Commit-Position: refs/heads/master@{#14442}
2016-09-29 22:28:12 +00:00
danilchap
798896a4aa Replace RtcpReceiveTimeInfo with rtcp::ReceiveTimeInfo
structs are exactly the same but last one follow naming style.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2368983002
Cr-Commit-Position: refs/heads/master@{#14415}
2016-09-28 09:54:30 +00:00
danilchap
9532124659 RTCPReceiver store cname as std::string.
simplifying cname management.

Remove RTCPUtility::RTCPCnameInformation
since it was last use of the structure.

BUG=webrtc:5565
NOTRY=true

Review-Url: https://codereview.webrtc.org/2354333004
Cr-Commit-Position: refs/heads/master@{#14399}
2016-09-27 14:05:39 +00:00
danilchap
92ea601e90 Move class RTCPHelp::RTCPPacketInformation into RTCPReceiver
Use it by pointer instead of by reference.
Renamed PacketInformation members to follow style,
Unused members removed.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2366563002
Cr-Commit-Position: refs/heads/master@{#14375}
2016-09-23 17:36:12 +00:00
danilchap
1b1863a11a Replace rtcp packet parsing in the RtcpReceiver.
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2316093002
Cr-Commit-Position: refs/heads/master@{#14301}
2016-09-20 08:40:00 +00:00
Danil Chapovalov
530b3f5d06 Merge RtcpReceiver::Handle<Packet>Item functions into Handle<Packet>
As a preparation to replace parsing implementation.

BUG=webrtc:5260
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2337283003 .

Cr-Commit-Position: refs/heads/master@{#14240}
2016-09-15 16:41:12 +00:00
Danil Chapovalov
91511f13e1 Split RtcpReceiver::HandleSenderReceiverReport into two functions
as a preparation to replace parsing implementation

BUG=webrtc:5260
R=philipel@webrtc.org

Review URL: https://codereview.webrtc.org/2340763002 .

Cr-Commit-Position: refs/heads/master@{#14237}
2016-09-15 14:24:42 +00:00
danilchap
17366bc090 Remove handling unused rtcp packets.
App, ExtendedJitterReport and VoipMetric in ExtenedReports are not
used when received (no callbacks, no state change), so removed.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2320703003
Cr-Commit-Position: refs/heads/master@{#14204}
2016-09-14 06:54:55 +00:00
danilchap
dd12892ede Reland of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2332673003/ )
Reason for revert:
Fuzzer changed not use functions moved to private.

Original issue's description:
> Revert of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2320603002/ )
>
> Reason for revert:
> Breaks fuzzer compilation.
>
> Original issue's description:
> > Make rtcp parsing implementation private in RtcpReceiver:
> > Function just for Parse and for Callbacks moved to private section.
> > All handles moved from protected to private section.
> >
> > BUG=webrtc:5260
> > R=sprang@webrtc.org
> >
> > Committed: https://crrev.com/faf708e238c7b43a732fbebf79ac9298b4b95a95
> > Cr-Commit-Position: refs/heads/master@{#14181}
>
> TBR=sprang@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5260
>
> Committed: https://crrev.com/180e4525ca7c9a23602cdf37a8756df7d23e7143
> Cr-Commit-Position: refs/heads/master@{#14182}

TBR=sprang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2336213002
Cr-Commit-Position: refs/heads/master@{#14200}
2016-09-13 19:23:33 +00:00
danilchap
180e4525ca Revert of Make rtcp parsing implementation private in RtcpReceiver (patchset #1 id:1 of https://codereview.webrtc.org/2320603002/ )
Reason for revert:
Breaks fuzzer compilation.

Original issue's description:
> Make rtcp parsing implementation private in RtcpReceiver:
> Function just for Parse and for Callbacks moved to private section.
> All handles moved from protected to private section.
>
> BUG=webrtc:5260
> R=sprang@webrtc.org
>
> Committed: https://crrev.com/faf708e238c7b43a732fbebf79ac9298b4b95a95
> Cr-Commit-Position: refs/heads/master@{#14181}

TBR=sprang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2332673003
Cr-Commit-Position: refs/heads/master@{#14182}
2016-09-12 08:40:44 +00:00
Danil Chapovalov
faf708e238 Make rtcp parsing implementation private in RtcpReceiver:
Function just for Parse and for Callbacks moved to private section.
All handles moved from protected to private section.

BUG=webrtc:5260
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/2320603002 .

Cr-Commit-Position: refs/heads/master@{#14181}
2016-09-12 08:31:23 +00:00
danilchap
59cb2bd20e Adjust RtcpReceiver to be testable with callbacks:
Instead of full RtpRtcpImpl takes interface of all functions it needs from it.
Added single function for parsing packets and sending feedback, moving that
logic from RtpRtcpImpl to RtcpReceiver.

BUG=webrtc:5260

Review-Url: https://codereview.webrtc.org/2274573002
Cr-Commit-Position: refs/heads/master@{#13960}
2016-08-29 18:08:53 +00:00
danilchap
853ecb21f7 Style cleanup in UpdateTmmbr:
function names style updated,
unused return type removed.
Comment style fixed, redundant comments removed.
pass-by-pointer parameter changed to pass-by-value because can't be nullptr any more.

NOTRY=true
BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2258523005
Cr-Commit-Position: refs/heads/master@{#13848}
2016-08-22 15:26:22 +00:00
danilchap
da161d795c Reformat rtcp_receiver
git cl format --full

BUG=webrtc:5565
NOTRY=true

Review-Url: https://codereview.webrtc.org/2259213002
Cr-Commit-Position: refs/heads/master@{#13832}
2016-08-19 14:29:51 +00:00
danilchap
2b616397de Remove TMMBRSet class
by cleaning RTCPReceiveInfo class
and following cleaning of RTCPReceiver::BoundingSet function.

BUG=webrtc:5565

Review-Url: https://codereview.webrtc.org/2254703003
Cr-Commit-Position: refs/heads/master@{#13817}
2016-08-18 13:17:48 +00:00
danilchap
287e54820b Cleanup RtcpReceiver::TMMBRReceived function
BUG=webrtc:951

Review-Url: https://codereview.webrtc.org/2250633002
Cr-Commit-Position: refs/heads/master@{#13786}
2016-08-16 22:15:46 +00:00
danilchap
13deaad1bd TMMBRHelp moved from member object/base class to stack object,
indicating the usage of this helper is local.
With local usage critical section become obvisously useless and removed.

BUG=webrtc:5565
R=åsapersson

Review-Url: https://codereview.webrtc.org/1959013003
Cr-Commit-Position: refs/heads/master@{#12881}
2016-05-24 20:25:35 +00:00
danilchap
7c9426cf38 Replaced CriticalSectionWrapper with rtc::CriticalSection in rtp_rtcp module
Review URL: https://codereview.webrtc.org/1877253002

Cr-Commit-Position: refs/heads/master@{#12359}
2016-04-14 10:05:37 +00:00
Taylor Brandstetter
5f0b83b7fb Enabling rtcp-rsize negotiation and fixing some issues with it.
Sending of reduced size RTCP packets should be enabled only if it's
enabled in the send parameters (which corresponds to the remote description).

Since the RTCPReceiver's RtcpMode isn't used at all, I removed it to ease
confusion.

BUG=webrtc:4868
R=pbos@webrtc.org, pthatcher@google.com, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1713493003 .

Cr-Commit-Position: refs/heads/master@{#12057}
2016-03-18 22:02:13 +00:00
Danil Chapovalov
c1e55c7136 rtt calculation handles time go backwards
CompactNtpIntervalToMs renamed to CompactNtpRttToMs and handle special cases:
large values consider negative/invalid and result in value of 1.
0 result consider too small and increases to 1.

BUG=590996
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1763823003 .

Cr-Commit-Position: refs/heads/master@{#11928}
2016-03-09 14:14:45 +00:00
Danil Chapovalov
b65f3e39d7 [cleanup] Remove unused fields/functions from rtcp module.
Removed fields are initialized but unused.
Removed functions are not called, sometimes are not defined.

BUG=webrtc:5565
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1726403003 .

Cr-Commit-Position: refs/heads/master@{#11839}
2016-03-02 12:26:19 +00:00
danilchap
6db6cdc604 [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1513303003

Cr-Commit-Position: refs/heads/master@{#11025}
2015-12-15 10:54:50 +00:00
danilchap
b8b6fbb7a5 lint build/include errors fixed in rtp_rtcp module
BUG=webrtc:5277
R=mflodman

Review URL: https://codereview.webrtc.org/1505993003

Cr-Commit-Position: refs/heads/master@{#10971}
2015-12-10 13:05:35 +00:00
Henrik Kjellander
ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

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

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00
sprang
7dc39f331a Avoid data race in RtcpReceiver.
See eg https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/3930/steps/video_engine_tests/logs/stdio

Also some cleanup, lock annotations.

BUG=

Review URL: https://codereview.webrtc.org/1401463003

Cr-Commit-Position: refs/heads/master@{#10266}
2015-10-13 16:17:56 +00:00
pbos
da903eaabb Unify newapi::RtcpMode and RTCPMethod.
BUG=webrtc:1695
R=solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1373903003

Cr-Commit-Position: refs/heads/master@{#10143}
2015-10-02 09:37:18 +00:00
Erik Språng
6b8d355168 Reland "Wire up send-side bandwidth estimation."
Revert was patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/

The culprit was RTC_DCHECK(poller_thread_->Start()); in rampup_test.cc

BUG=webrtc:4173
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1362303002 .

Cr-Commit-Position: refs/heads/master@{#10052}
2015-09-24 13:07:17 +00:00
Erik Språng
c9bbeb0354 Revert of Wire up send-side bandwidth estimation. (patchset #8 id:140001 of https://codereview.webrtc.org/1338203003/ )
Reason for revert:
Breaking some Android bots.
https://chromegw.corp.google.com/i/client.webrtc/builders/Android32%20Tests%20%28L%20Nexus5%29

Original issue's description:
> Wire up send-side bandwidth estimation.
>
> BUG=webrtc:4173
>
> Committed: https://crrev.com/ef165eefc79cf28bb67779afe303cc2365885547
> Cr-Commit-Position: refs/heads/master@{#10012}

TBR=stefan@webrtc.org, kjellander@webrtc.org
NOPRESUBMIT=false
NOTREECHECKS=false
NOTRY=false
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1362923002 .

Cr-Commit-Position: refs/heads/master@{#10029}
2015-09-23 11:52:01 +00:00
sprang
ef165eefc7 Wire up send-side bandwidth estimation.
BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1338203003

Cr-Commit-Position: refs/heads/master@{#10012}
2015-09-22 12:10:58 +00:00
Peter Boström
ac547a6538 Remove channel ids from various interfaces.
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.

IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately

BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1335353005 .

Cr-Commit-Position: refs/heads/master@{#9978}
2015-09-17 21:06:02 +00:00
Erik Språng
a38233a586 Removed extended jitter report from RtcpSender.
This was never used (value always 0, when sent)

BUG=2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1208843003 .

Cr-Commit-Position: refs/heads/master@{#9631}
2015-07-24 07:58:29 +00:00
Peter Boström
fe7a80c38c Prevent sender RTCP signals for receive-only channels.
Since RTCP packets are delivered to both senders and receivers that
correspond the receivers currently log that NACKed packets are missing,
since they have no direct connection to the sending side or the RTP
packet history. Also preventing triggering on SR requests and PLI/FIR.

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

Review URL: https://webrtc-codereview.appspot.com/45249004

Cr-Commit-Position: refs/heads/master@{#9071}
2015-04-23 15:52:58 +00:00
mflodman@webrtc.org
96abda0316 Removing FEC functionality from the default RTP module.
This CL removes the last default module methods used from ViEEncoder and
the default module itself will be removed in a separate CL.

BUG=769
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35309004

Cr-Commit-Position: refs/heads/master@{#8505}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8505 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-25 13:50:51 +00:00
pbos@webrtc.org
1d0fa5d352 Add RtcpPacketTypeCounter stats to new API.
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/37489004

Cr-Commit-Position: refs/heads/master@{#8429}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-19 12:47:45 +00:00
pbos@webrtc.org
a28a91d2f0 Fix data race for RTCPReceiver stats callback.
Annotates the callback which identifies the bug, then fixes it.

R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/40009004

Cr-Commit-Position: refs/heads/master@{#8390}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8390 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 14:45:44 +00:00
tommi@webrtc.org
4161715e3f Remove ChangeUniqueID.
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.

It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.

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

Review URL: https://webrtc-codereview.appspot.com/37849004

Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:14:13 +00:00
asapersson@webrtc.org
df7b65ba01 Change CreateOrGetReportBlockInformation to have one return path.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8113 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-21 13:07:04 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
pbos@webrtc.org
d16e839c6d Rtp-Rtcp sender cleanup.
Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.

Also removed const on non-pointer/reference types for related files.

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

Review URL: https://webrtc-codereview.appspot.com/34469004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-19 13:49:55 +00:00
asapersson@webrtc.org
cb79141eab Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc.
When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.

Removed unused function ResetRTT.

BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33659005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 14:30:32 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats.
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-14 11:52:04 +00:00
asapersson@webrtc.org
2dd3134e50 Add stats for duplicate sent and received NACK requests.
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7559 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-29 12:42:30 +00:00
pbos@webrtc.org
2f4b14e3f3 Make RTCP sender report send media bytes.
r6654 changed RtpSender::Bytes() to return the number of bytes sent
instead of number of media bytes. This is used by VideoEngine for stats.
This change broke RTCP which sends this same count as the number of
payload bytes sent (excluding headers and padding).

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 15:25:39 +00:00
asapersson@webrtc.org
8098e07478 Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
Add counter to RTCP sender and RTCP receiver.
Add video api GetRtcpPacketTypes().

BUG=2638
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5575 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-19 11:59:02 +00:00
sprang@webrtc.org
a6ad6e5b58 Add callbacks for send channel rtcp statistics
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5220 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 09:48:44 +00:00
asapersson@webrtc.org
766154aa1d Removed unused code.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 08:35:50 +00:00
asapersson@webrtc.org
7d6bd22019 Propagate estimated RTT from receivers to rtt observer.
BUG=1613
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-31 12:14:34 +00:00
asapersson@webrtc.org
8469f7b328 Added support for sending and receiving RTCP XR packets:
- Receiver reference time report block
- DLRR report block (RFC3611).

BUG=1613
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2196010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4898 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 13:15:34 +00:00