45 Commits

Author SHA1 Message Date
Per
28a44564c9 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )"
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca.

This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/

patchset #1 is a pure reland.
patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail.

Original cl description:
Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to  Call (and BitrateAllocator)

R=stefan@webrtc.org
TBR=mflodman@webrtc.org

BUG=webrtc:5687

Review URL: https://codereview.webrtc.org/1947873002 .

Cr-Commit-Position: refs/heads/master@{#12630}
2016-05-04 15:13:06 +00:00
perkj
825eb58d59 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )
Reason for revert:
Fails in the waterfall here:

https://build.chromium.org/p/client.webrtc/builders/Win32%20Debug/builds/7832/steps/rtc_media_unittests/logs/stdio

Original issue's description:
> Remove SendPacer from ViEEncoder
>
> This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to  Call (and BitrateAllocator)
>
> BUG=webrtc:5687
>
> Committed: https://crrev.com/857c5ccdb56e4c94196f7c6227abd5993c95abe2
> Cr-Commit-Position: refs/heads/master@{#12620}

TBR=stefan@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1947853002
Cr-Commit-Position: refs/heads/master@{#12621}
2016-05-04 08:08:15 +00:00
perkj
857c5ccdb5 Remove SendPacer from ViEEncoder
This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to  Call (and BitrateAllocator)

BUG=webrtc:5687

Review-Url: https://codereview.webrtc.org/1917793002
Cr-Commit-Position: refs/heads/master@{#12620}
2016-05-04 07:09:56 +00:00
kwiberg
22feaa3d15 Replace scoped_ptr with unique_ptr in webrtc/modules/
Except in places where this would break out-of-tree code,
such as Chromium.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1785173002

Cr-Commit-Position: refs/heads/master@{#12037}
2016-03-17 16:17:49 +00:00
pbos
a26ac925f7 Reland of move ignored return code from modules. (patchset #1 id:1 of https://codereview.webrtc.org/1736663004/ )
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.

Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}

TBR=tommi@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1737013002

Cr-Commit-Position: refs/heads/master@{#11762}
2016-02-25 12:50:09 +00:00
torbjorng
da33a8a2a2 Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
Reason for revert:
Breaks Chromium.

Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58c

TBR=tommi@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=

Review URL: https://codereview.webrtc.org/1736663004

Cr-Commit-Position: refs/heads/master@{#11761}
2016-02-25 12:34:12 +00:00
Peter Boström
f14c47a58c Remove ignored return code from modules.
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).

BUG=
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1703833002 .

Cr-Commit-Position: refs/heads/master@{#11747}
2016-02-24 15:51:23 +00:00
stefan
0665f0518f Fix OOB read in pacing test.
BUG=webrtc:5490

Review URL: https://codereview.webrtc.org/1727283002

Cr-Commit-Position: refs/heads/master@{#11737}
2016-02-24 11:04:22 +00:00
jbauch
d2a22960c3 Enable cpplint for webrtc/modules/pacing and fix all uncovered cpplint errors.
This CL enableds cpplint for webrtc/modules/pacing.

BUG=webrtc:5460
NOTRY=true

TESTED=Fixed issues reported by:
find webrtc/modules/pacing -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.

Review URL: https://codereview.webrtc.org/1674423002

Cr-Commit-Position: refs/heads/master@{#11531}
2016-02-09 07:18:30 +00:00
Stefan Holmer
c482eb3c84 Don't account for audio in the pacer budget.
We should only account for audio packets in the pacer budget if we also
are allocating bandwidth for the audio streams.

BUG=chromium:567659,webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1524763002 .

Cr-Commit-Position: refs/heads/master@{#11053}
2015-12-16 15:55:09 +00:00
Erik Språng
ad113e50d2 Fix bug in calculation of averge queue time in paced sender.
Also work around a flaw in fake encoder which caused bogus perf
regression in rampup tests.

BUG=560434
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1474533006 .

Cr-Commit-Position: refs/heads/master@{#10811}
2015-11-26 15:26:25 +00:00
sprang
0a43fef6dc Allow pacer to boost bitrate in order to meet time constraints.
Currently we limit the enocder so that frames aren't encoded if the
expected pacer queue is longer than 2s. However, if the queue is full
and the bitrate suddenly drops (or there is a large overshoot), the
queue time can be long than the limit.

This CL allows the pacer to temporarily boost the pacing bitrate over
the 2s window.

BUG=

Review URL: https://codereview.webrtc.org/1412293003

Cr-Commit-Position: refs/heads/master@{#10729}
2015-11-20 17:00:41 +00:00
Henrik Kjellander
0b9e29c87d Remove include dirs from modules/{media_file,pacing}
Also move files out of media_file/source.

BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1435093002 .

Cr-Commit-Position: refs/heads/master@{#10647}
2015-11-16 10:12:32 +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
e23e737177 Disable pacer disabling.
Since the pacer is always enabled, removing enable/disable which makes
all packet queueing succeed. Also renaming one of the ::SendPackets
::InsertPacket to avoid confusion.

BUG=webrtc:1695, webrtc:2629
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1392513002 .

Cr-Commit-Position: refs/heads/master@{#10211}
2015-10-08 09:44:29 +00:00
Stefan Holmer
586b19bdb6 Enable probing with repeated payload packets by default.
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.

In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.

BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1327933003 .

Cr-Commit-Position: refs/heads/master@{#9984}
2015-09-18 09:14:42 +00:00
Stefan Holmer
01b488831b Use padding to achieve bitrate probing if the initial key frame has too few packets.
BUG=4350
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44879004

Cr-Commit-Position: refs/heads/master@{#9134}
2015-05-05 08:21:32 +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
stefan@webrtc.org
e9f0f591b5 Enable bitrate probing by default in PacedSender.
BUG=crbug:425925
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33359004

Cr-Commit-Position: refs/heads/master@{#8379}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8379 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-16 15:48:29 +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
pkasting@chromium.org
2656bf813f Fix ExpectedQueueTimeMs() to avoid truncation or overflow.
BUG=none
TEST=none
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7714 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-17 22:21:14 +00:00
stefan@webrtc.org
d839e0ab52 Reduce to 2 probes when probing for initial bandwidth.
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23359005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 19:33:55 +00:00
sprang@webrtc.org
dcebf2daa7 Reworked paced sender queue
Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage.

Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these.

Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long.

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

Review URL: https://webrtc-codereview.appspot.com/27869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-04 16:27:16 +00:00
stefan@webrtc.org
82462aade0 Adds support for sending first set of packets at increasingly higher bitrates to probe the link and faster ramp up to a high bitrate.
Also wires up a finch experiment to control this.

R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7505 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-23 11:57:05 +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
03c817e405 Fix pacer to accept duplicate sequence numbers on different SSRCs.
BUG=3550
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6610 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-07 10:20:35 +00:00
stefan@webrtc.org
bee164a214 Fix test issues and a win compile error introduced with r6605.
Also changes the name of a variable which has been hijacked by windef.h (included by windows.h), which forces #define near and #define far upon us. This issue was introduced via the following inclusion chain:
bwe_test_framework_unittest.cc includes
  paced_sender.h
    tick_util.h
      windows.h
        windef.h

And causes EXPECT_NEAR(foo, bar, near); to expand to EXPECT_NEAR(foo, bar,); generating a very confusing compile error.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6606 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-04 17:00:06 +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
stefan@webrtc.org
cb254aac3b Enable pacing by default and remove the option to disable it from the new API.
BUG=1672
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6416 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-12 15:12:25 +00:00
pbos@webrtc.org
709e29742e Simplify pacer interface.
New interface uses two bitrates (max/min). The pace multiplier is also
removed from the interface and instead utilized outside. Min bitrate
will be filled with padding if there's not enough media to transmit.

Also fixes a bug in minimum transmission bitrate that made it ignore
REMBs. A regression test has been added to catch it.

BUG=3014
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5723 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-19 10:59:52 +00:00
jiayl@webrtc.org
9fd8d87ff5 Adds APIs for reporting pacer queuing delay.
BUG=2775
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8959005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5621 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-27 22:32:40 +00:00
stefan@webrtc.org
dd393e7b9d Measure pacer queue size based on when packets are inserted rather than captured.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5291 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 22:03:27 +00:00
stefan@webrtc.org
19a40ff05b Ensure that no packet stays in the pacer queue for longer than 2 seconds.
BUG=2682
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5182 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-27 14:16:20 +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
stefan@webrtc.org
b2c8a952a7 Improving padding rules and breaking out bw allocation to ViEEncoder.
BUG=1837
TESTS=vie_auto_test --automated, trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4693 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-06 13:58:01 +00:00
stefan@webrtc.org
80865fd611 Don't pace out packets or generate padding when the pacer is disabled.
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4513 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-09 11:31:23 +00:00
pbos@webrtc.org
db6e3f8bc5 Fix root-relative includes for pacing/.
BUG=1662
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4329 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-11 09:50:05 +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
c3cc375499 Add support for padding in pacer.
This improves pacer-based padding by making sure it limits padding according to:
- Never pad more than 800 kbps.
- Padding + media should not go above a given target bitrate.

Also adds appropriate unittests to make sure we reach the given targets.

BUG=1837
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1582005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4168 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-04 09:36:56 +00:00
pwestin@webrtc.org
52b4e8871a Adding trace and changing pacing constants
BUG=1721,1722
R=mikhal@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1380005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 19:02:17 +00:00
pwestin@webrtc.org
52aa019e98 Avoid adding duplicates in pacer lists.
Review URL: https://webrtc-codereview.appspot.com/1329007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3899 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-25 17:35:56 +00:00
stefan@webrtc.org
bfacda60be Add interface to signal a network down event.
- In real-time mode encoding will be paused until the network is back up.
- In buffering mode the encoder will keep encoding, and packets will be
  buffered at the sender. When the buffer grows above the target delay
  encoding will be paused.
- Fixes a couple of issues related to pacing which was found with the new test.
- Introduces different max bitrates for pacing and for encoding. This allows
  the pacer to faster get rid of the queue after a network down event.

(Work based on issue 1237004)

BUG=1524
TESTS=trybots,vie_auto_test

Review URL: https://webrtc-codereview.appspot.com/1258004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-27 16:36:01 +00:00
pwestin@webrtc.org
db4185664c Introduced pause and resume to the pacer
Review URL: https://webrtc-codereview.appspot.com/1217007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3717 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22 23:39:29 +00:00
pwestin@webrtc.org
b518017e71 Adding pacing module, will replace the transmission_bucket in the RTP module.
TESTED=unittest
Review URL: https://webrtc-codereview.appspot.com/930015

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3073 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-09 20:56:23 +00:00