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
4436b4436a
Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE.
...
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16619004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6324 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 09:05:30 +00:00
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
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
...
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
a826006132
Add NACK and RPSI packet types to RTCP packet builder.
...
Fixes bug found when parsing received RPSI packet.
BUG=2450
R=mflodman@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 09:53:51 +00:00
88abf11cad
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
...
BUG=3111
TEST=try bots
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 16:53:51 +00:00
8f69330310
Replace scoped_array<T> with scoped_ptr<T[]>.
...
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar ...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
cd70119a10
Calculate local/remote clock delta and capture ntp timestamp in receiver's timebase.
...
BUG=3111
TEST=new performance tests
R=niklas.enbom@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5976 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 22:10:24 +00:00
93fd25c20c
* Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus.
...
* Cast rtp header extension to int in log in rtp_utility.cc.
BUG=3237
TEST=try bots
R=stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-24 20:33:08 +00:00
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
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
2c89b5cb27
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
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
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
440fa23553
Make RTPHeaderParser skip over unknown RTP header extensions rather than bail out.
...
BUG=2954
R=mflodman@webrtc.org , stefan@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5786 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-25 19:57:07 +00:00
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
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
b1f5010075
VoE changes to allow forwarding of packets from VoE to ViE BWE.
...
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 10:38:25 +00:00
af839b28b0
Add AIMD option to BWE API.
...
TEST=trybots
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10319005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5755 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-24 09:42:08 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
6811b6e308
Callback for send bitrate estimates - new roll
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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
e9abd591d7
Making RemoteRateControl::min_configured_bit_rate_ configurable
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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
096e8d9f94
Revert 5259 "Callback for send bitrate estimates"
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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
