4591fbd09f
Use size_t more consistently for packet/payload lengths.
...
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.
This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.
BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom
Review URL: https://webrtc-codereview.appspot.com/23129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-20 22:28:14 +00:00
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
49ff40e32e
Make SetREMBData accept vector of SSRCs.
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7697 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-13 14:42:37 +00:00
d42a3adf42
Remove partially defined WebRtcRTPHeader from Parse().
...
It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change.
To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28919004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-07 11:02:12 +00:00
0bae1fab4a
Wire up bandwidth stats to the new API and webrtcvideoengine2.
...
Adds stats to verify bandwidth and pacer stats.
BUG=1788
R=mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-05 14:05:29 +00:00
dcebf2daa7
Reworked paced sender queue
...
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.
Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.
Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.
BUG=
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:27:16 +00:00
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
76960d5f74
For FIR packet, payload length is zero, so SendToNetwork function is failing.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-22 09:47:14 +00:00
eb24b04f16
Add periodic logging of received RTP headers and estimated clock offsets for e2e delay.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7444 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-14 11:40:13 +00:00
3cefbc99f4
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE.
...
This also marks all virtual overrides of other classes in the same files.
This will make a subsequent change I intend to do safer, where I'll change the
argument types of the base Transport functions, by breaking the compile if I
miss any overrides.
This also highlighted a number of unused functions. I've removed some of these.
TBR=mflodman@webrtc.org , pkasting@chromium.org
BUG=none
TEST=none
Review URL: https://webrtc-codereview.appspot.com/28709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 09:42:53 +00:00
2c0cdbce22
Estimating NTP time with a given RTT.
...
RemoteNtpTimeEstimator needed user to give a remote SSRC and it intended to call RtpRtcp module to obtain RTT, to be able to calculate Ntp time.
When RTT cannot be directly obtained from the RtpRtcp module with the specified SSRC, RemoteNtpTimeEstimator would fail.
This change allows RemoteNtpTimeEstimator to calculate NTP with an external RTT estimate.
An immediate benefit is that capture_start_ntp_time_ms_ can be obtained in a Google hangout call.
BUG=
TEST=chromium + hangout call
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7407 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 10:52:43 +00:00
730d270771
Remove callback from RtpDepacketizer::Parse().
...
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30489004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-29 08:00:22 +00:00
315669939a
Fix typo from RtpPacketizerH264.
...
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27609004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7295 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-25 07:31:42 +00:00
38344ed280
Move thread_annotations.h to webrtc/base/.
...
R=andresp@webrtc.org , mflodman@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/27579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-24 06:05:00 +00:00
5a098c51ea
Refactor VP8 de-packetizer.
...
It's duplicated to parse VP8 RTP packet at the moment. We firstly call
RTPPayloadParser functions to save parsed information in RTPPayload
structure, then copy them to RTP header.
This CL removes RTPPayloadParser class and directly saves parsed data in
RTP header.
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7211 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:58:20 +00:00
1fb5d1204b
Initialize restored_packet in nack_rtx_unittest.cc.
...
This is to get the DrMemory Full bots to go green, this was previously
suppressed. This fix is likely hiding a real bug that should be
investigated, but it's not a regression from before. The issue should
not be closed before we figure out why this is the case and revert this
"fix".
TBR=stefan@webrtc.org
BUG=3183
Review URL: https://webrtc-codereview.appspot.com/30369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7169 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 16:16:00 +00:00
b5e6bfc76a
Make RTPSender/RTPReceiver generic.
...
Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26399004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 11:05:55 +00:00
6071b0636d
Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.
...
This will make further changes to these classes safer by ensuring that the
compile breaks if the base class changes and not all overrides are fixed.
This also highlighted a number of unused functions which I've removed.
-- This is was reviewed in https://webrtc-codereview.appspot.com/19309004/ , but
-- a new cl was needed to resolve a small conflict before committing.
BUG=none
TEST=none
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7162 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 07:42:33 +00:00
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
c30e9e2300
Ignore FEC packet in stats, if it is first packet on ssrc.
...
BUG=chrome:410456
R=mflodman@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 08:20:18 +00:00
0a214ffa8a
Setting marker bit on DTMF correctly
...
BUG=1157
R=braveyao@webrtc.org , pbos@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 11:46:54 +00:00
f8723d666a
Add unit tests to rtcp_receiver_test.
...
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6994 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 07:35:06 +00:00
c3c29113d1
Expose setPayloadType on the rtp_sender. Thus allowing other users of this module
...
to set the payload type to be used without having to call SendOutgoingData.
BUG=3694
R=asapersson@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18289004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-27 09:39:43 +00:00
047a46f8b4
Remove Android.mk build files.
...
These files are generally not maintained and break, some contain files
that don't exist anymore and do not build anymore. If we need to add
some of these back we should really set up a bot for them.
R=andrew@webrtc.org , glaznev@webrtc.org , henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/15249004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6974 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-26 08:48:51 +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
58e2d262fc
Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics().
...
Fixes issues where statistics only was reported for the first stream if
configured with simulcast, and in case of RTX the reported statistics was
depending on the order of the report blocks.
Also fixes issues with multiple report blocks in the SendStatisticsProxy and the
RtcpStatisticsCallback. SendStatisticsProxy is now aware of RTX ssrcs, and the
RTCPReceiver is calling the RtcpStatisticsCallback with the correct SSRCs, and
not only the primary stream SSRC.
R=mflodman@webrtc.org , sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 15:10:49 +00:00
817a034cf2
Fix TimeToSendPadding return to be 0 if no padding bytes are sent.
...
BUG=3694
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15149005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-14 08:24:47 +00:00
c27543d297
Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field.
...
BUG=3679
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6881 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13 07:40:45 +00:00
c891fee7ab
Make a int64 constant use ULL suffix so it wont get truncated.
...
BUG=3690
TESTED=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6878 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-12 22:39:06 +00:00
59a2f9f584
Remove the old H264 code now that a new H.264 packetizer has been implemented.
...
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6847 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 15:09:24 +00:00
9d74f7ce8c
Fix single nalu packetization bug.
...
Nalus which had the same size as the max payload size would cause the payload size accounting to wrap.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6846 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-07 15:02:16 +00:00
8b033adb19
Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant.
...
R=kjellander@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21109004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6833 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-06 08:06:53 +00:00
84b9e1e9d9
Fix for retransmission. Base layer packets were not retransmitted.
...
Issue introduced in r6669.
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6816 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-04 11:59:42 +00:00
2ec560606b
Add H.264 packetization.
...
This also includes:
- Creating new packetizer and depacketizer interfaces.
- Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition.
- Created a Create() factory method for packetizers and depacketizers.
R=niklas.enbom@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/21009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-31 14:59:24 +00:00
e75d78d32d
Integrate rtcp packet class to rtcp receiver tests.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6795 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-29 08:21:50 +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
5ab7616983
Remove remains of WEBRTC_NO_STL.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6752 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-22 06:48:58 +00:00
8b94e3da0f
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled.
...
This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was.
BUG=
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16929004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-17 16:10:14 +00:00
63c60ed224
Remove old padding path in RTPSender.
...
Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 09:37:29 +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
9e1acc8728
Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 .
...
A few places were relying on temporalIdx being signed. Fix to explicitly check
for kNoTemporalIdx.
TBR=pbos,stefan
Review URL: https://webrtc-codereview.appspot.com/13939005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6669 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 20:33:39 +00:00
dd6780d85d
Remove always-true expression.
...
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/16059004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6668 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-11 19:34:54 +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.
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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.
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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.
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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
72491b9a90
Count total bytes sent in RTPSender::Bytes().
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Previously only media bytes were included, this adds header bytes and
padding bytes to the calculation.
BUG=
R=stefan@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 16:24:54 +00:00
8f1512140e
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel.
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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
5ac876bae0
Revert "Remove remains of WEBRTC_NO_STL." (rev 6641).
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Reason breaks linux_memcheck.
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16869004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6645 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-10 07:41:59 +00:00
