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77 Commits

Author SHA1 Message Date
andresp@webrtc.org
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
andresp@webrtc.org
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
stefan@webrtc.org
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
pbos@webrtc.org
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
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
stefan@webrtc.org
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
pbos@webrtc.org
72491b9a90 Count total bytes sent in RTPSender::Bytes(). 
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
andresp@webrtc.org
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
andresp@webrtc.org
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
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
pbos@webrtc.org
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
stefan@webrtc.org
88e0dda475 Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-04 09:20:42 +00:00
henrike@webrtc.org
fe526ff10f fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. 
BUG=N/A
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-25 20:59:51 +00:00
stefan@webrtc.org
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
stefan@webrtc.org
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
stefan@webrtc.org
420b2567f3 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. 
This caused only the first retransmission to be successful.
Introduced with https://code.google.com/p/webrtc/source/detail?r=5728.

BUG=1811
R=asapersson@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-30 12:17:15 +00:00
andrew@webrtc.org
2c3f1abb69 Replace flooding logs in rtp_sender.cc with a comment. 
Started occurring after:
https://webrtc-codereview.appspot.com/11129004

BUG=3153
R=andresp@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 21:26:34 +00:00
stefan@webrtc.org
2f8d5f3302 Check if a header extension is registered before updating it and fail silently if it's not. 
BUG=
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-15 12:28:46 +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
andresp@webrtc.org
d09d074827 Protect write of send_target_bitrate. 
This issue was catch by tsan bot.

BUG=3065
R=stefan@webrtc.org, andrew

Review URL: https://webrtc-codereview.appspot.com/10619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5790 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-26 14:27:34 +00:00
sprang@webrtc.org
efcad39f77 Fix race condition in RTPSEnder. 
In RTPSender::SendPayloadType(), payload_type_ should not be read
without owning send_critsect_.

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

Review URL: https://webrtc-codereview.appspot.com/8199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5778 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-25 16:51:35 +00:00
stefan@webrtc.org
16395228f5 Properly account for retransmitted packets when not using the pacer. 
This regression was introduced in r5728.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5729 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-19 19:34:07 +00:00
stefan@webrtc.org
7c6ff2da26 Fixes RTX related bugs. 
- An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream.
- The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions.

TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-19 18:14:52 +00:00
sprang@webrtc.org
5a320fb06f Race condition in RTPSender 
RTPSender::sending_media_ should be guarded by send_critsect_. Fix this
in GetSendSideDelay, SendPadData and TimeToSendPadding.
Also add appropriate thread annotations.

BUG=3029
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5697 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-13 15:12:37 +00:00
wu@webrtc.org
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
sprang@webrtc.org
346094cb01 Incorrect overhead calculation when using FEC + RTP extension headers. 
When frames are fragmented inte multiple RTP packets in order to not
exceed a maximum packet size, the header overhead calculation must
take into account that FEC redundancy packets may use more than the
12 bytes of the basic RTP header. For example, a csrc list or extension
headers may be present.

BUG=2899
R=phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8769005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 08:40:33 +00:00
sprang@webrtc.org
5314e85926 Race condition in RTPSender::UpdateRtpStats 
The ssrc should not be access directly from the ssrc_ field, without
holding the send_critsect_ lock. A better way is to just use the SSRC()
getter method.

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

Review URL: https://webrtc-codereview.appspot.com/7539006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 13:20:36 +00:00
sprang@webrtc.org
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
sprang@webrtc.org
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
sprang@webrtc.org
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
sprang@webrtc.org
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
stefan@webrtc.org
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
sprang@webrtc.org
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
stefan@webrtc.org
79b63206b9 Fixes a crash in fullstack tests introduced with r5209. 
TBR=mflodman@webrtc.org
BUG=1812

Review URL: https://webrtc-codereview.appspot.com/4689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 13:34:28 +00:00
stefan@webrtc.org
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
henrik.lundin@webrtc.org
6e95d7afab Increment RTP timestamps for padding packets 
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.

A test was implemented to verify that the padding packets do
get their own timestamps.

BUG=2611
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-15 08:59:19 +00:00
stefan@webrtc.org
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
pbos@webrtc.org
e07049f19f Lock RTPSender statistics. 
Suppressing these errors in TSan has become tedious. It's better to just
lock them.

BUG=2349
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2197004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-09-10 11:29:17 +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
d4f607e70a Fixes to padding when driven by encoder. 
- Allow padding to be sent on an ssrc which doesn't produce video, therefore
  never having the last_packet_marker_bit_ set.
- Add the random timestamp offset to all padding packets.
- Store the capture time of padding packets to properly create an offset.

BUG=2258
TEST=trybots and a new test which will be committed separately.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2060005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-19 15:55:01 +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
12dc1a38ca Switch C++-style C headers with their C equivalents. 
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
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
hclam@chromium.org
1a7b9b94be Cleanup WebRTC tracing 
The goal of this change is to:
1. Remove unused tracing events.
2. Organize tracing events to facilitate measurement of end to end latency.

The major change in this CL is to use ASYNC_STEP such that operation
flow can be traced for the same frame.

R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1761004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-08 21:31:18 +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
stefan@webrtc.org
a4c5abb52a Make sure padding packets are sent. 
BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1717006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-25 15:46:16 +00:00
hclam@chromium.org
2e402ce873 Enqueue packet in pacer if sending fails 
If a packet cannot be sent while pacer is in use it should be
queued. This avoid packet loss due to congestion.

BUG=1930
R=pwestin@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1693004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-20 20:18:31 +00:00
stefan@webrtc.org
8ccb9f9716 Fixes some pacer/padding issues found while testing. 
- A bug was introduced in r4234 causing no paced packets to be sent.
- Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss.
- Have all packets go through the pacer if pacing is enabled to avoid reordering.
- Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc.

BUG=1837
TEST=trybots and vie_auto_test --automated
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1682004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-19 14:13:42 +00:00
stefan@webrtc.org
508a84b255 Wire up pacer-based padding. 
This connects the pacer-based padding with the RTP modules, which will
generate padding packets roughly according to what the pacer suggests.
It will only generate padding packets of maximum size to keep the number
off padding packets as small as possible. This also sets a limit of how much
padding + media bitrate which the pacer is allowed to "request" from the
RTP modules.

Padding will for now only be generated by the first sending RTP module.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1612005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-17 12:53:37 +00:00
stefan@webrtc.org
a817962bab Refactor padding and rtp header functionality. 
BUG=1837
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1611004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-04 13:47:36 +00:00