| c482eb3c84e958118451fbc443a0b2ba296e7441 |
|
16-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| b86d4e4a8dec1eb1b801244a2a97cda66f561d8e |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Prepare the AudioSendStream to be hooked up to send-side BWE.
This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 7c704b82893bbe7fc206b004fb9dfe6e69a986ef |
|
04-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h in stefan@'s ownership.
Replaces system_wrappers' logging in call/, bitrate_controller/, pacing/
and remote_bitrate_estimator/.
BUG=webrtc:5118
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1484503002 .
Cr-Commit-Position: refs/heads/master@{#10896}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| ad113e50d251c95adcf501ed29f8312ad1193a35 |
|
26-Nov-2015 |
Erik Språng <sprang@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 0a43fef6dc8ce95a3ec52921870e08799ee9a250 |
|
20-Nov-2015 |
sprang <sprang@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 0b9e29c87da2d9c1a3792d2c87197b0688b68e4e |
|
16-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| e23e737177cf5d131a6d4a4d229aa513c5270a59 |
|
08-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| ebbf8a805b45613b4cb118e4eb0cebe7eeee69ac |
|
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it.
BUG=
Review URL: https://codereview.webrtc.org/1350163005
Cr-Commit-Position: refs/heads/master@{#10005}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 586b19bdb615dde34cdcf107272d8857fe2f5631 |
|
18-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| c62642c7a662a2a88293b82192e2240049f0cbb9 |
|
07-Jul-2015 |
stefan <stefan@webrtc.org> |
Make the BWE threshold adaptive.
This improves self-fairness and competing for resources with TCP flows.
BUG=4711
Review URL: https://codereview.webrtc.org/1151603008
Cr-Commit-Position: refs/heads/master@{#9545}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 01b488831bf7cb3276d8bdfbe0204dfbdbbba725 |
|
05-May-2015 |
Stefan Holmer <stefan@webrtc.org> |
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}
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| e9f0f591b5d7af63cbd7ad8b9c3b1058de601b92 |
|
16-Feb-2015 |
stefan@webrtc.org <stefan@webrtc.org> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 0200f70792982c4b5987cf4364dcd53f8aa94779 |
|
16-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Set webrtc_rtp category to be disabled by default.
Should not affect webrtc standalone. For chromium, disabling helps
mitigate viewing performance problems.
BUG=chromium:441440
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41909004
Cr-Commit-Position: refs/heads/master@{#8375}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 0b1534c52eab79372557a6d81aaf4dd9407f55d3 |
|
15-Dec-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 2656bf813ff77121f235e241b6057a04cce6da03 |
|
17-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| dcebf2daa76aebd021dbb778f3908375b819e59a |
|
04-Nov-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 82462aade0ad3fbe76284ac294b41fb500a1d2f8 |
|
23-Oct-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 89fd1e8e99b47544f5fa36bc7fbde1d089536b0b |
|
15-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Improvements to the pacer where it lost some budget due to truncation errors.
With this CL the resolution is increased to microseconds and proper rounding
is done in the Process() function. This means that we will be allowed to send
more than prior to r6664 as we previously truncated away parts of our budget.
We will also not lose budget due to inaccurate calculations in
TimeUntilNextProcess(), which was a regression in r6664.
BUG=cr/393950
TEST=out/Debug/webrtc_perf_tests --gtest_filter=RampUpTest.Simulcast
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6694 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 168f23faa5b8a49d4dd709c6649e77d5fecf36bf |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 03c817e4059f3199f72c37b1df463b03ac9cc9f4 |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 875ad49dee49e95d212a91eb9bb1af327a80ee85 |
|
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert conversion from TickTime to int64_t in paced sender.
Introduced with r6600, causing flakes in SuspendBelowMinBitrate. The reason for this flake is currently unknown.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6605 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 88e0dda475e1f6a5fa5855eec0be111bddbf00ac |
|
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| cb254aac3b18ac41ff175c816190390589182965 |
|
12-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 709e29742eb44a26bca3998d4c19797d6558775d |
|
19-Mar-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 32c3247418aa0c4c6644f0c98a51fe33660b79ea |
|
21-Jan-2014 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for libtalkmobile build error
bug=b/12549061
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5404 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 7fb75ecbd4226ca3fccdb7e64ce19850059c8c13 |
|
20-Dec-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| dd393e7b9d6a44e668ffcf1f1ff526343a385cf6 |
|
13-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| b627f676b3be77e8d9da55104d6553d6972cd2a1 |
|
28-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large.
BUG=2682
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4599005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5190 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|
| 19a40ff05b1ca43a3b4169f311de7b0139269c22 |
|
27-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| ef2d55461be949de6d6265f52665289e573c6b1b |
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21-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Increase size of pacer window to 500 ms as that better matches the encoder.
BUG=1812
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4129006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5154 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 9b82f5a6ed2ceb04f72b66c1d3cca67a2bbcec3a |
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13-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| b2c8a952a7a996b89c6ff2ecdc1364641f2571f6 |
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06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 80865fd61152a105bab87796937ed436883957d9 |
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09-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 6eb53f71d6930f0f92c69d128cb01d108c9de1d6 |
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21-Jun-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix memory bot failure
Exit the method with critical setting held. This should make
the memory bot happy.
TBR=pwestin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1704005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4251 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 2e402ce873f48e0848468345d848bd3fff75dd3e |
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20-Jun-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 8ccb9f9716f306dd1ec284b4f61f0b6c82c08c3c |
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19-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 8ad3ec9722f73f22a5574c0d6fc4cf99e910afa4 |
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04-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix build error introduced with r4168.
TBR=mflodman@webrtc.org
BUG=1837
Review URL: https://webrtc-codereview.appspot.com/1610004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4169 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| c3cc375499b16d463346408dc62f73493b2ee4e5 |
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04-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 0f29810288a2ae53c8e8c9c18bbb3656ac3744d5 |
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06-May-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix crash in pacer.
BUG=1731
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1410006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3964 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 52b4e8871a7c43a12177cb9a717baff3fb2680c0 |
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02-May-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| 52aa019e98bde9f65ac577fbb3f94cb6cb9749be |
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25-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| bfacda60be5f816a04bd278d4aa4cd3d8fd01e9f |
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27-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| db4185664c83e83432e9c11823f81df35bb0f8e6 |
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23-Mar-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
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| b518017e71d7cc0eab031f6259e4d87aaeb5c9c5 |
|
09-Nov-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/pacing/paced_sender.cc
|