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178 Commits

Author SHA1 Message Date
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
asapersson@webrtc.org
0f2809a5ac Add RTCP packet class. 
Adds packet types: sr, rr, bye, fir.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5592 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-21 08:14:45 +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
mflodman@webrtc.org
c320027d6a Don't print a warning if RTPPacketHistory::SetStorePacketStatus is called 
twice with the same settings.

Without this change, setting up a call with the new video API will
print a trace warning.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-18 14:51:00 +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
stefan@webrtc.org
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
sprang@webrtc.org
a45cac0fb7 Avoid potential dead lock in StreamStatisticianImpl 
Extract callbacks for rtp/rtcp data, from StreamStatisticianImpl to
ReceiveStatisticsImpl, into separate methods with guards agains having
incorrect lock order.

BUG=2856
R=andresp@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5441 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 16:22:08 +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
c00adbed73 Race in StreamStatisticianImpl::GetStatistics vs. ::IncomingPacket 
StreamStatisticianImpl.ssrc_ is protected by stream_lock_, should
  be cached while holding lock to avoid race condition.

  Also, rtp_callback_ do not need to be called in GetStatistics() at all

BUG=2853
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5435 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-27 10:42:48 +00:00
asapersson@webrtc.org
871d949299 Remove loopback setup in RtpRtcpImplTest. Changed to use two separate rtp/rtcp modules. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5426 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-24 13:23:49 +00:00
sprang@webrtc.org
0e93257cee Add callbacks for receive channel RTP statistics 
This allows a listener to receive new statistics (byte/packet counts, etc) as it
is generated - avoiding the need to poll. This also makes handling stats from
multiple RTP streams more tractable. The change is primarily targeted at the new
video engine API.

TEST=Unit test in ReceiveStatisticsTest.
Integration tests to follow as call tests when fully wired up.

BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5416 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-23 10:00:39 +00:00
sprang@webrtc.org
7dba27c740 Potential dead lock in receive statistics 
A dead lock could occur if the following to code paths are called
concurrently:

ReceiveStatisticsImpl::IncomingPacket() ->
  StreamStatisticianImpl::IncomingPacket()

StreamStatisticianImpl::GetStatistics() ->
  ReceiveStatisticsImpl::StatisticsUpdated()

Solution is to release ReceiveStatisticsImpl lock after lookup/lazy-init of StreamStatisticianImpl. Don't need to hold it when doing the call to StreamStatisticianImpl::IncomingPacket().

BUG=2818
R=asapersson@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5406 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-21 16:33:37 +00:00
asapersson@webrtc.org
efaeda0c76 Add configuration and test for extended RTCP reference time reports to new video api. 
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-01-20 08:34:49 +00:00
braveyao@webrtc.org
2fb72cfeec Add include guards to forward_error_correction_internal.h 
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5335 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-24 05:06:12 +00:00
andresp@webrtc.org
7fb75ecbd4 Add thread_annotations for clang targets. 
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.

R=niklas.enbom@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-20 20:20:50 +00:00
sprang@webrtc.org
54ae4ffb9e Add callbacks for receive channel RTCP statistics. 
This allows a listener to receive new statistics as it is generated - avoiding the need to poll. This also makes handling stats from multiple RTP streams more tractable.
The change is primarily targeted at the new video engine API.

TEST=Unit test in ReceiveStatisticsTest. Integration tests to follow as call tests when fully wired up.

BUG=2235
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-19 13:26:02 +00:00
mflodman@webrtc.org
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
asapersson@webrtc.org
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
pbos@webrtc.org
5ab756703e Revert r5294 to re-roll r5293. 
To fix races in test each stream now owns its own encoder/decoder.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-16 12:24:44 +00:00
sprang@webrtc.org
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
turaj@webrtc.org
41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." 
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
> 
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5409004

TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 23:57:54 +00:00
asapersson@webrtc.org
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
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
henrik.lundin@webrtc.org
e9abd591d7 Making RemoteRateControl::min_configured_bit_rate_ configurable 
The minimum bitrate can now be configured from WrappingBitrateEstimator.

BUG=2698
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5279 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-13 08:42:42 +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
fischman@webrtc.org
eb7def234e Fix compilation errors on Fedora 20. 
peerconnection_jni.cc: syscall() comes from <unistd.h>
RTPtimeshift.cc: char* being compared to 0 instead of the atoi() of it
rtp_payload_registry_unittest.cc: avoid narrowing int to uint32.

BUG=2700
R=andrew@webrtc.org, fischman@webrtc.org, henrik.lundin@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5019004

Patch from Victor Costan <costan@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5248 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-09 21:34:30 +00:00
sprang@webrtc.org
88615f0659 Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-06 13:16:44 +00:00
asapersson@webrtc.org
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
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
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
stefan@webrtc.org
c4726d06fa Make RTPSender::SendPadData public. 
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5219 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-05 09:16:33 +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
sprang@webrtc.org
499631c1e4 Utility class for reading/writing network-byte-ordered integers. 
BUG=
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2151008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 13:22:48 +00:00
stefan@webrtc.org
47fadba750 Add include stdlib.h to files using abs. 
abs function is declared in stdlib.h

Committing for alextaran@chromium.org.
Reviewed here: http://review.webrtc.org/4239004/

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 12:03:56 +00:00
pbos@webrtc.org
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
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
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
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
asapersson@webrtc.org
38599510df Parse next RTCP XR report block after an unsupported block type. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-12 08:08:26 +00:00
pbos@webrtc.org
57eb858698 Remove ".." from include_dirs in build/common. 
BUG=1662
TEST=compile on trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2332004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5107 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-11 10:20:27 +00:00
stefan@webrtc.org
48df38114d Fix for making sure that the packet in order checks are done prior to updating the last received packet state. 
Without this fix all packets are considered out-of-order by the rtp receiver, causing the last received state
in the rtp receiver to never get valid.

Also makes sure that only valid timestamps and receive times are used for audio/video sync.

BUG=2608
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-08 15:18:52 +00:00
asapersson@webrtc.org
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
asapersson@webrtc.org
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
asapersson@webrtc.org
042e91c2b2 Fix for incorrect RTT estimation. A too low RTT value could be estimated. 
R=andrew@webrtc.org, holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 13:58:31 +00:00