| ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
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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/remote_bitrate_estimator/test/packet_sender.cc
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| e23e737177cf5d131a6d4a4d229aa513c5270a59 |
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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/remote_bitrate_estimator/test/packet_sender.cc
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| 91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
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17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 318673cf5a1b230d445a50fdc4869f4b8f99c85d |
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04-Sep-2015 |
sprang <sprang@webrtc.org> |
Update SendTimeHistory to store complete PacketInfo, not just send time
This will be used for the send side bitrate estimation. Storing various
meta-data about packets that can be retreived when arrival time feeback
arrives.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1288033008
Cr-Commit-Position: refs/heads/master@{#9859}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 2386a45dc710f74174ae8b3ba18b744839b969d9 |
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31-Jul-2015 |
Cesar Magalhaes <magalhaesc@webrtc.org> |
Supporting Pause/Resume, Sending Estimate logging. Corrected plot colors
PacketSender can now log Pause/Resume events into a MetricRecorder. Solved estimate error and optimal bitrate issue for test 5.7 (multiple short TCP flows).
Added Sending Estimate logging and plotting.
Fixed plotting issue on plot_dynamics.py
Now lines with the same color (in different boxes) correspond to the same flow.
Adjusting plot_dynamics.py font size according to number of variables.
R=asapersson@webrtc.org
Review URL: https://codereview.webrtc.org/1270543002 .
Cr-Commit-Position: refs/heads/master@{#9664}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 9c261f2d13793fbb5a0d07b26bec4154bc38342b |
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15-Jul-2015 |
Cesar Magalhaes <magalhaesc@webrtc.org> |
Supports logging for dynamic and histogram plots on Simulation Framework.
---- Dynamic receiving rate.
---- Dynamic packet-loss.
---- Dynamic objective function.
---- Dynamic available capacity.
---- Dynamic available capacity per flow.
---- Average delay Histogram with standard deviation or 5th/95th percentiles.
---- Average bitrate Histogram with error bars.
---- Optimal average bitrate dashed line.
---- Average packet-loss Histogram.
---- Total objective function Histogram.
Added media Pause/Resume methods to Video and TcpSender.
Modified LinkedSet: computing GlobalPacketLossRatio even if packet's sequence_number overflows.
Added small randomization to frame send times, modified bwe_test_framework_unittest accordingly.
Taking offset time into account for plotting.
Added nada_unittests.
Added bwe_unittests.
Added a RateCounter to BweReceiver (replaced ReceivingRate)
Added LossAccount.
Fixed NadaBweReceiver issue: using sender_timestamp instead of creation_time.
Fixed memory leaks.
Fixed int division rounding issues.
Supporting plots on bandwidth Estimators:
Logging received packet information on on SubClassesBweReceiver::ReceivePacket
Updating RateCounter, updating packet loss account and relieving LinkedSet when necessary.
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1202253003 .
Cr-Commit-Position: refs/heads/master@{#9585}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 0908d0dcf227b78c55ba4d59d7f5eadcc95af75b |
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01-Jun-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix issue with RTT computations in simulator.
R=magalhaesc@google.com
Review URL: https://webrtc-codereview.appspot.com/56429005
Cr-Commit-Position: refs/heads/master@{#9343}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 53d0dc3f067d6440c37157db6060d2f2c9fa8446 |
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07-May-2015 |
Stefan Holmer <stefan@webrtc.org> |
Wire up RTT to send-side GCC and TCP.
BUG=4548
R=magalhaesc@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52429004
Cr-Commit-Position: refs/heads/master@{#9151}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| c81591d63f5e441bd26025a5e986bb2ebfd9fdfd |
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06-May-2015 |
Cesar Magalhaes <magalhaesc@webrtc.org> |
NADA's proposal from Cisco.
The implementation of this proposal is in progress.
More unittest will be added.
Sender side is being implemented.
Some constants need to be tuned.
BUG=4550
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43299004
Cr-Commit-Position: refs/heads/master@{#9146}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| bcbcd84888819ca913d809e162cc7c7615bc98c7 |
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28-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Improve TCP implementation by adding ssthresh and make it possible to start it with an offset.
Add a propagation delay to tests and make the run-time configurable for the fairness tests.
Handle losses in-between feedback messages.
BUG=4549
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49819004
Cr-Commit-Position: refs/heads/master@{#9099}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| ac69016b0f5f5e4b9521cd1761ad728afd703d86 |
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22-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Improve TCP by adding a real timeout to in flight packets.
Note that the timeout should depend on the smoothed RTT, but for now is hard coded to 1000 ms.
This solves issues where a full cwnd gets lost.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51739004
Cr-Commit-Position: refs/heads/master@{#9051}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 5f92051f063d5d827a16102d72cf1c23d423d3ec |
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21-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix bug in TCP implementation (simulations).
The problem was that only ACKed packets were subtracted from in_flight_, but lost packets were never removed, which caused TCP to stop sending eventually.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43239004
Cr-Commit-Position: refs/heads/master@{#9041}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| a51e8f490c52fc939cb9a384efbd91031aabac9c |
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17-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix some simulation issues.
Don't default to an infinite queue.
Make sure the computation of missing packets is correct.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49059004
Cr-Commit-Position: refs/heads/master@{#9028}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 1d19893f3aa35bb132053ca546006a0f819c9347 |
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17-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Add TCP fairness test.
BUG=4548
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43199004
Cr-Commit-Position: refs/heads/master@{#9026}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| d4e80146e37c37283c8818a000f1d67f232c8ca6 |
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16-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix build errors in r9022 / 09bdc1e5f5a9.
Implicit casts detected by Win64 Release.
TBR=pbos@webrtc.org
BUG=4548
Review URL: https://webrtc-codereview.appspot.com/44239004
Cr-Commit-Position: refs/heads/master@{#9023}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 379593792082a86f389df9b1b790cc0fe9eb9975 |
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16-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Adds a simplified Reno-type TCP sender.
BUG=4559
R=sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44189004
Cr-Commit-Position: refs/heads/master@{#9021}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| dcbd3acbef881b0e6a5d459c6f6b7c7080eb1a20 |
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10-Apr-2015 |
Stefan Holmer <stefan@webrtc.org> |
Improve BWE plotting and logging to make it possible to use multiple windows/figures.
Also adds plotting of the BWE threshold and offset.
R=solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43119004
Cr-Commit-Position: refs/heads/master@{#8968}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 4346d92578e5acbf3c40c89967c548e8f72e7543 |
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18-Mar-2015 |
stefan@webrtc.org <stefan@webrtc.org> |
Use SendTimeHistory to keep track of send times in simulations.
Use SendTimeHistory to keep track of send times in simulations.
Keep piggybacking send time in PacketInfo for now but use history in
order to be more in line with what we expect to do.
Landing this for sprang@. Original CL: https://review.webrtc.org/43559004/
TBR=sprang@webrtc.org
BUG=4308
Review URL: https://webrtc-codereview.appspot.com/48569004
Cr-Commit-Position: refs/heads/master@{#8778}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8778 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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| 14b0279416c4916534c1e76939b0b8927a208a04 |
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16-Feb-2015 |
stefan@webrtc.org <stefan@webrtc.org> |
Break out code from bloated files in the BWE simulator.
No changes to functionality.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34209004
Cr-Commit-Position: refs/heads/master@{#8374}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8374 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
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