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
a84b0a6dab
Small refactor on ViE to remove redudant conditions and long ifdefs.
...
BUG=3694
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22069004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 16:46:46 +00:00
f9460688a6
Make sure padding is sent on the first sending RTP module.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6774 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-24 16:41:25 +00:00
376b4ea93f
Fix breakage introduced by r6691.
...
ModuleRtpRtcpImpl returned incorrectly on RemoteNTP as the
RTCPReceiver::NTP changed return type.
BUG=
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6693 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-15 15:51:33 +00:00
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
eec6ecdb1e
Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC.
...
---
Fixes for re-enabling more MSVC level 4 warnings: webrtc/ edition
This contains fixes for the following sorts of issues:
* Possibly-uninitialized local variable
* Signedness mismatch
* Assignment inside conditional
This also contains a small number of other cleanups to nearby code. In
particular several warning-disables for MSVC are removed because they don't seem
to be necessary (either that warning is not enabled or the code does not trigger
it).
BUG=crbug.com/81439
TEST=none
R=henrika@webrtc.org , pkasting@chromium.org
Review URL: https://webrtc-codereview.appspot.com/18769004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6667 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:09:59 +00:00
180e516bef
Thread annotate RTCPSender.
...
Also fixes data races in RTCPSender::SetCSRCStatus() and
RTCPSender::SetStartTimestamp().
BUG=
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6666 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 15:36:26 +00:00
168f23faa5
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.
...
This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems.
R=pbos@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21869005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 13:44:02 +00:00
4ef438e2de
Remove the send-side cname getter APIs from voice and video engine.
...
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.
R=henrika@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 09:55:30 +00:00
8f1512140e
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 09:39:23 +00:00
d11bec40b2
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-08 14:32:58 +00:00
2bb1bdab8d
Preserve RTP states for restarted VideoSendStreams.
...
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 13:06:48 +00:00
a15fbfdcde
Add round-robin selection of send stream to pad on.
...
BUG=1812
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 17:32:05 +00:00
ef92755780
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.
...
This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out.
BUG=1811
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15629005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 08:25:29 +00:00
cd70119a10
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
...
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 22:10:24 +00:00
dc80bae2a6
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG.
...
Clean some logs and add asserts in the way.
BUG=3153
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-08 11:06:12 +00:00
ebdb0e3ad0
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005.
...
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.
This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.
BUG=
TBR=solenberg@webrtc.org ,henrikg@webrtc.org ,stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-06 23:49:08 +00:00
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
0e5a2b5de6
Handle the invalid case of setting multiple stream_bitrates if there is only a single send stream registered.
...
This can happen when switching between multiple streams and a single while getting feedback from the receiver.
BUG=2881
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5486 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-04 14:38:25 +00:00
e6b871bb29
Added method for getting default module state and protect agains a
...
read/write race for child_modules_.
BUG=2731
TEST=tsan
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5919005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5306 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 08:30:40 +00:00
e7b1e11283
Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."
...
> Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
>
> > Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
> >
> > R=holmer@google.com
> >
> > Review URL: https://webrtc-codereview.appspot.com/5049004
>
> TBR=asapersson@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/5799004
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5299 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 14:40:36 +00:00
87ad57bc75
Incorrect iterator++ in ModuleRtpRtcpImpl::RegisterVideoBitrateObserver
...
The iterator is incremented both in loop header and loop body. Should
only be incremented in header.
BUG=2727
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5899004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5295 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 07:43:51 +00:00
86bb56a7f5
Revert 5228 "Use the RTT from RtcpRttStats class if provided whe..."
...
> Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
>
> R=holmer@google.com
>
> Review URL: https://webrtc-codereview.appspot.com/5049004
TBR=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5285 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 16:16:45 +00:00
6811b6e308
Callback for send bitrate estimates - new roll
...
Issue https://webrtc-codereview.appspot.com/4459004/ was commited as
r5259, after which flakiness was detected and a rollback was performed
at r5261.
Patch Set 1 of this issue is the code submitted in r5259. Subsequent
patch sets fixes a race condition which caused the seen problems.
The root cause was a dead lock between a thread sending rtp packets and
and a timed module processing thread:
webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock
webrtc::Bitrate::Process() // Get Bitrate lock
webrtc::RTPSender::ProcessBitrate()
webrtc::ModuleRtpRtcpImpl::Process()
...
webrtc::Bitrate::Update() // Get Bitrate lock
webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock
webrtc::RTPSender::SendToNetwork()
...
This is fixed in Bitrate::Process() by releasing the lock before
calling the callback.
BUG=2235
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 09:46:59 +00:00
096e8d9f94
Revert 5259 "Callback for send bitrate estimates"
...
CL is causing flakiness in RampUpTest.WithoutPacing.
> Callback for send bitrate estimates
>
> BUG=2235
> R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/4459004
R=mflodman@webrtc.org , pbos@webrtc.org
TBR=mflodman
Review URL: https://webrtc-codereview.appspot.com/5579005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 14:07:33 +00:00
2656cf9f4c
Callback for send bitrate estimates
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 12:53:03 +00:00
96a9b2dcdc
Use the RTT from RtcpRttStats class if provided when sending/receiving NACK.
...
R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/5049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5228 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 15:06:56 +00:00
ebad765ee0
Add callbacks for send channel rtp statistics
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:29:02 +00:00
0a3c1471b8
Add API to query video engine for the send-side delay.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4559005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 14:05:07 +00:00
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
71f055fb41
Add send frame rate statistics callback
...
BUG=2235
R=mflodman@webrtc.org , pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 15:09:27 +00:00
7e9315b42e
Adds support for sending redundant payloads over RTX.
...
TEST=trybots
BUG=1812
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-04 10:24:26 +00:00
ffe1b17b57
Lock access to ModuleRtpRtcpImpl::simulcast_.
...
Fixes race between RegisterSendPayload and SendOutgoingData.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4099006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5152 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:53:13 +00:00
8d02f5dc71
Added API for enabling/disabling RTCP Receiver Reference Time extension.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3419005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 08:57:04 +00:00
1ae1d0c471
Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module).
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2383004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 12:46:11 +00:00
9b82f5a6ed
Fix for RTX in combination with pacing.
...
Retransmissions didn't get sent over RTX when pacing was enabled since
the pacer didn't keep track of whether a packet was a retransmit or not.
BUG=1811
TEST=trybots
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-13 15:29:21 +00:00
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
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
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
3e7703640f
Remove unused constants, so chrome can enable a warning for that. Patch from thakis@
...
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2296006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4844 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 22:05:05 +00:00
28a331eede
Add support for multiple report blocks.
...
Use a weighted average of fraction loss for bandwidth estimation.
TEST=trybots and vie_auto_test --automated
BUG=1811
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2198004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 07:49:56 +00:00
59f20bb735
Break out RTCPSender dependency on ModuleRtpRtcpImpl.
...
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2191004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-09 16:02:19 +00:00
b2c8a952a7
Improving padding rules and breaking out bw allocation to ViEEncoder.
...
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2170004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:58:01 +00:00
cac7325b84
Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule.
...
Found with tsan.
TEST=try job and tsan
R=holmer@google.com
Review URL: https://webrtc-codereview.appspot.com/2156004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4661 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-03 12:11:12 +00:00
286fe0b04d
Revert 4585 "Revert "Revert 4582 "Reverts a second set of reverts caused by a bug in ..."""
...
...and fixes the RTCP bug.
BUG=2277
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2089004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4588 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-21 20:58:21 +00:00
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
1a65d6c36b
Reverts a second set of reverts caused by a bug in a dependency.
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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
822fbd8b68
Update talk to 50918584.
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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
e270331481
Fix duplicate code
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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
12dc1a38ca
Switch C++-style C headers with their C equivalents.
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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