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43 Commits

Author SHA1 Message Date
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
kwiberg@webrtc.org
00b8f6b364 Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36229004

Cr-Commit-Position: refs/heads/master@{#8517}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-02-26 14:43:50 +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
asapersson@webrtc.org
9ffd8fe96b Indentation changes. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 08:22:50 +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
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
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
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
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
asapersson@webrtc.org
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
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
pbos@webrtc.org
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
stefan@webrtc.org
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
henrike@webrtc.org
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
stefan@webrtc.org
1a65d6c36b 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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-21 16:22:21 +00:00
wu@webrtc.org
822fbd8b68 Update talk to 50918584. 
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
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
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
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
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
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
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
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
2f44673d66 WebRtc_Word32 => int32_t for rtp_rtcp/ 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1279007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-08 11:08:41 +00:00
edjee@google.com
79b0289bfc Adds event traces and counters for WebRTC receive side. 
Review URL: https://webrtc-codereview.appspot.com/1279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-04 19:43:34 +00:00
stefan@webrtc.org
a678a3baee Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. 
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-21 07:42:11 +00:00
stefan@webrtc.org
20ed36dada Break out RtpClock to system_wrappers and make it more generic. 
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.

Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.

TEST=vie_auto_test, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 14:01:20 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/ 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-22 18:19:23 +00:00