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Author SHA1 Message Date
danilchap
f8385aded0 rtcp::Pli moved into own file and got a Parse function 
Created rtcp::Psfb abstract class between rtcp::Pli and rtcp::RtcpPacket to hold common data for Feedback Message.

BUG=webrtc:5260

Review URL: https://codereview.webrtc.org/1446513002

Cr-Commit-Position: refs/heads/master@{#10823}
 2015-11-27 13:36:17 +00:00
danilchap
50c5136cb2 RTCP Bye packet moved to own file 
Bye class got support for Parsing
 Reason field implemented

Review URL: https://codereview.webrtc.org/1430013003

Cr-Commit-Position: refs/heads/master@{#10741}
 2015-11-22 17:03:16 +00:00
danilchap
0219c9b4bf rtcp::App moved into own file and got Parse function 
Review URL: https://codereview.webrtc.org/1437353003

Cr-Commit-Position: refs/heads/master@{#10688}
 2015-11-18 13:56:57 +00:00
danilchap
df948f03b3 rtcp::ReportBlock refactored to contain parsing 
Review URL: https://codereview.webrtc.org/1420283022

Cr-Commit-Position: refs/heads/master@{#10633}
 2015-11-13 11:03:18 +00:00
Henrik Kjellander
98f53510b2 system_wrappers: rename interface -> include 
BUG=webrtc:5095
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1413333002 .

Cr-Commit-Position: refs/heads/master@{#10438}
 2015-10-28 17:17:50 +00:00
Peter Boström
ebc0b4e993 Use webrtc/base/logging.h for rtp_rtcp. 
BUG=webrtc:5118
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1422023002 .

Cr-Commit-Position: refs/heads/master@{#10437}
 2015-10-28 15:39:43 +00:00
tommi
e4f96501fc Remove system_wrappers/interface/trace_event.h 
BUG=

Review URL: https://codereview.webrtc.org/1417773002

Cr-Commit-Position: refs/heads/master@{#10346}
 2015-10-21 06:00:57 +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
sprang
86fd9ed6f9 Set RtcpSender transport at construction. 
BUG=

Review URL: https://codereview.webrtc.org/1365043002

Cr-Commit-Position: refs/heads/master@{#10106}
 2015-09-29 11:45:51 +00:00
pbos
2d566686a2 Unify Transport and newapi::Transport interfaces. 
BUG=webrtc:1695
R=stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1369263002

Cr-Commit-Position: refs/heads/master@{#10096}
 2015-09-28 16:59:36 +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
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros. 
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
 2015-09-17 07:24:51 +00:00
sprang
233bd87d45 Add RemoteEstimatorProxy for capturing receive times 
For use when send-side bandwidth estimation is enabled.

Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1290813008

Cr-Commit-Position: refs/heads/master@{#9898}
 2015-09-08 20:25:20 +00:00
Erik Språng
521875a9a4 Use RtcpPacket to send APP in RtcpSender 
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1311453002 .

Cr-Commit-Position: refs/heads/master@{#9827}
 2015-09-01 08:11:36 +00:00
Erik Språng
ca28fdcf9f Use RtcpPacket to send XR (RTRR, DLRR, VOIP) in RtcpSender 
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1304123003 .

Cr-Commit-Position: refs/heads/master@{#9820}
 2015-08-31 12:01:08 +00:00
sprang
d83df50e95 Use RtcpPacket to send TMMBN in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1302403002

Cr-Commit-Position: refs/heads/master@{#9793}
 2015-08-27 08:05:12 +00:00
sprang
d8ee4f9915 Use RtcpPacket to send BYE in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1306893003

Cr-Commit-Position: refs/heads/master@{#9763}
 2015-08-24 10:25:27 +00:00
sprang
81a3e60c63 Use RtcpPacket to send TMMBR in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1296163004

Cr-Commit-Position: refs/heads/master@{#9755}
 2015-08-21 12:30:17 +00:00
sprang
dd4edc5813 Reland of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1300863002/ ) 
Reason for revert:
This wasn't the cause of the breakage. Re-reverting.
https://code.google.com/p/webrtc/issues/detail?id=4923

Original issue's description:
> Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ )
>
> Reason for revert:
> A few bots started failing rtc_unittests after this was commited. Ex https://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5048
>
> Original issue's description:
> > Use RtcpPacket to send REMB in RtcpSender
> >
> > BUG=webrtc:2450
> > R=asapersson@webrtc.org
> >
> > Committed: 35ab4baa20
>
> TBR=asapersson@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:2450
>
> Committed: https://crrev.com/141c5951f4beda868797c2746002a4b1b267ab2a
> Cr-Commit-Position: refs/heads/master@{#9723}

TBR=asapersson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1309723002

Cr-Commit-Position: refs/heads/master@{#9754}
 2015-08-21 11:21:56 +00:00
sprang
141c5951f4 Revert of Use RtcpPacket to send REMB in RtcpSender (patchset #1 id:1 of https://codereview.webrtc.org/1290573004/ ) 
Reason for revert:
A few bots started failing rtc_unittests after this was commited. Ex https://build.chromium.org/p/client.webrtc/builders/Linux64%20Debug/builds/5048

Original issue's description:
> Use RtcpPacket to send REMB in RtcpSender
>
> BUG=webrtc:2450
> R=asapersson@webrtc.org
>
> Committed: 35ab4baa20

TBR=asapersson@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1300863002

Cr-Commit-Position: refs/heads/master@{#9723}
 2015-08-18 11:37:39 +00:00
Erik Språng
35ab4baa20 Use RtcpPacket to send REMB in RtcpSender 
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1290573004 .

Cr-Commit-Position: refs/heads/master@{#9722}
 2015-08-18 09:54:18 +00:00
sprang
cf7f54d6f4 Use RtcpPacket to send RPSI in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1291013002

Cr-Commit-Position: refs/heads/master@{#9704}
 2015-08-13 11:37:48 +00:00
sprang
0365a27f56 Use RtcpPacket to send SLI in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1268383002

Cr-Commit-Position: refs/heads/master@{#9695}
 2015-08-11 08:02:44 +00:00
sprang
62dae19098 Use RtcpPacket to send FIR in RtcpSender 
BUG=webrtc:2450

Review URL: https://codereview.webrtc.org/1261323003

Cr-Commit-Position: refs/heads/master@{#9677}
 2015-08-05 09:37:21 +00:00
Erik Språng
72aa9a6c6e Use RtcpPacket to send PLI in RtcpSender 
BUG=webrtc:2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1262153003 .

Cr-Commit-Position: refs/heads/master@{#9666}
 2015-07-31 14:16:12 +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
Erik Språng
0ea42d319e Send Sdes using RtcpPacket 
BUG=2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1196863003.

Cr-Commit-Position: refs/heads/master@{#9504}
 2015-06-25 12:46:23 +00:00
Erik Språng
bdc0b0d869 Use RtcpPacket classes for SenderReport/ReceiveReport in RTCPSender 
BUG=2450
R=asapersson@webrtc.org

Review URL: https://codereview.webrtc.org/1170723002.

Cr-Commit-Position: refs/heads/master@{#9483}
 2015-06-22 13:21:40 +00:00
Peter Boström
9ba52f89ac Remove intermediate RTCP CNAME buffers. 
Sets CNAME using a pointer to only perform a copy inside the RTCP
sender.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50169005

Cr-Commit-Position: refs/heads/master@{#9346}
 2015-06-01 12:12:40 +00:00
Erik Språng
11beccd712 Remove external report blocks from RtcpSender and rtp_rtcp interface. 
Feature does not seem to be used and complicates other refactoring of
the rtcp module.

BUG=
R=asapersson@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54569004

Cr-Commit-Position: refs/heads/master@{#9304}
 2015-05-28 09:10:34 +00:00
Erik Språng
242e22b055 Refactor RTCP sender 
The main purpose of this CL is to clean up RTCPSender::PrepareRTCP, but
it has quite a few ramifications. Notable changes:

* Removed the rtcpPacketTypeFlags bit vector and don't assume
  RTCPPacketType values have a single unique bit set. This will allow
  making this an enum class once rtcp_receiver has been overhauled.

* Flags are now stored in a map that is a member of the class. This
  meant we could remove some bool flags (eg send_remb_) which was
  previously masked into rtcpPacketTypeFlags and then masked out again
  when testing if a remb packet should be sent.

* Make all build methods, eg. BuildREMB(), have the same signature.
  An RtcpContext struct was introduced for this purpose. This allowed
  the use of a map from RTCPPacketType to method pointer. Instead of
  18 consecutive if-statements, there is now a single loop.
  The context class also allowed some simplifications in the build
  methods themselves.

* A few minor simplifications and cleanups.

The next step is to gradually replace the builder methods with the
builders from the new RtcpPacket classes.

BUG=2450
R=asapersson@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48329004

Cr-Commit-Position: refs/heads/master@{#9166}
 2015-05-11 08:17:46 +00:00
Erik Språng
61be2a4016 Clean up RTCPSender. 
Reformat to current code style, remove non-const references, use
scoped_ptr, remove empty comments and dead code, etc..

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49019004

Cr-Commit-Position: refs/heads/master@{#9086}
 2015-04-27 11:32:31 +00:00
sprang@webrtc.org
779c3d16b9 Use ByteReader/ByteWriter instead of rtputility and manual shift/add. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41289004

Cr-Commit-Position: refs/heads/master@{#8761}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-17 16:44:54 +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
mflodman@webrtc.org
0abc6011b9 Remove SetCaptureDelay from the RTP module. 
This is a small step in getting rid of the default module, but also to
eventually delete FrameProviderBase completely.

BUG=769
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34229004

Cr-Commit-Position: refs/heads/master@{#8396}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8396 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-17 16:36:48 +00:00
sprang@webrtc.org
0200f70792 Set webrtc_rtp category to be disabled by default. 
Should not affect webrtc standalone. For chromium, disabling helps
mitigate viewing performance problems.

BUG=chromium:441440
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41909004

Cr-Commit-Position: refs/heads/master@{#8375}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-16 12:06:48 +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
d08d389ce8 Add field to counters for when first rtp/rtcp packet is sent/received. 
Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min).

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 12:03:11 +00:00
pbos@webrtc.org
9334ac2d78 Use vector of CSRCs for DeliverFrame & SetCSRCs. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28029004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 08:25:50 +00:00
pkasting@chromium.org
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
pbos@webrtc.org
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
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
pbos@webrtc.org
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
stefan@webrtc.org
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
pbos@webrtc.org
62bafae661 Some refactoring inside rtp_rtcp/. 
Renaming ModuleRTPUtility -> RtpUtility.
Renaming RTPHeaderParser -> RtpHeaderParser.
Making RtpHeaderParser accept size_t instead of int for packet length.
Making RtpUtility::RtpHeaderParser accept size_t for packet length.

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

Review URL: https://webrtc-codereview.appspot.com/19899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-08 12:10:51 +00:00
andresp@webrtc.org
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
stefan@webrtc.org
9d4762e8b6 Have changes to REMB trigger RTCP to be sent immediately. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 17:13:00 +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