| 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/bitrate_controller/bitrate_controller_unittest.cc
|
| e59041672283a28bde0b043c0c2bc198272f82e1 |
|
26-Mar-2015 |
Stefan Holmer <holmer@google.com> |
Moving the pacer and the pacer thread to ChannelGroup.
This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45549004
Cr-Commit-Position: refs/heads/master@{#8864}
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 792f1a14e2b62382c5c6080d0b8fdc5c89d27bc6 |
|
04-Mar-2015 |
stefan@webrtc.org <stefan@webrtc.org> |
Break out allocation from BitrateController into a BitrateAllocator.
This also refactors some of the padding and allocation code in ViEEncoder, and
makes ChannelGroup a simple forwarder from BitrateController to
BitrateAllocator.
This CL is part of a bigger picture, see https://review.webrtc.org/35319004/ for
details.
BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44399004
Cr-Commit-Position: refs/heads/master@{#8595}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8595 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values.
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| edeea91803eac26f6a229e88a7045797d2af61b7 |
|
08-Dec-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Change all system clock types to int64_t in bitrate_controller.
They are both compared to int64_t types inside the class, and is being called
with int64_t types. Could possibly cause bugs.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7832 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 83d4804a50a9b8ee5e029ef1cb62659611a03d7b |
|
10-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Put send-side bwe probing under finch experiment.
BUG=crbug/425925
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7668 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.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/bitrate_controller/bitrate_controller_unittest.cc
|
| 077593b8058b90ee4a3cc760ba04d0c026978aa5 |
|
19-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Ensure that the start bitrate can be set multiple times.
If the start bitrate is set twice, it will be set to the sum of the start
bitrates of the currently registered bitrate observers, or left unchanged
if the current estimate actually is greater than the sum.
BUG=3503
R=henrik.lundin@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6491 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 44caf01c34d4fddec039f917c83fed7e0ce977b2 |
|
26-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Re-submit: rev5775
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.
Additionally:
Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
Did not touch decrease logic, however since it can be triggered more often it
may decrease much faster and closer to the original written cap of once every
300ms + rtt.
Note:
rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5794 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 4e65602886830e19d2d76d4c07dc22c5c2106e8d |
|
26-Mar-2014 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add API to allow deducting bitrate from incoming estimates before the capacity is distributed among outgoing video streams. For example, this can be used to reserve space for audio streams.
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5791 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 6cd201cf31dc8e50bf815139b0c9fdc83d3ba2bf |
|
25-Mar-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5775 "Modify bitrate controller to update bitrate based o..."
This triggered an occasional TSAN failure in
CallTest.ReceivesPliAndRecoversWithNack e.g.:
http://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan/builds/1444/steps/memory%20test%3A%20video_engine_tests/logs/stdio
I managed to reproduce this locally and verified that reverting this CL
corrected it.
> Modify bitrate controller to update bitrate based on process call and not
> only whenever a RTCP receiver block is received.
>
> Additionally:
> Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
>
> Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
>
> Did not touch decrease logic, however since it can be triggered more often it
> may decrease much faster and closer to the original written cap of once every
> 300ms + rtt.
>
> Note:
> rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
> bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
>
> BUG=3065
> R=stefan@webrtc.org, mflodman@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/10529004
TBR=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10079005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5785 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| da07737e68e23e283466ae21965e43edfe621a12 |
|
25-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Modify bitrate controller to update bitrate based on process call and not
only whenever a RTCP receiver block is received.
Additionally:
Add condition to only start rampup after a receiver block is received. This was same as old behaviour but now an explicit check is needed to verify process does not ramps up before the first block.
Fix logic around capping max bitrate increase at 8% per second. Before it was only increasing once every 1 second and each increase would be as high as 8%. If receiver blocks had a different interval before it would lose an update or waste an update slot and not ramp up as much as a 8% (e.g. if RTCP received < 1 second).
Did not touch decrease logic, however since it can be triggered more often it
may decrease much faster and closer to the original written cap of once every
300ms + rtt.
Note:
rampup_tests.cc don't seem to be affected by this since there is no packet loss or REMB that go higher than expected cap.
bitrate_controller_unittests.cc are don't really simulate a clock and the process thread, but trigger update by inserting an rtcp block.
BUG=3065
R=stefan@webrtc.org, mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5775 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 07bc7344595e9cb3a3039eaffcb7a3ec2ca72928 |
|
21-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor in BitrateController module.
- Move condition of 0 bps as max meaning 1gbps from SendSideBandwidthEstimation to BitrateController.
- Remove condition on bitrate=0 meaning bandwidth estimation off as that could only happen when no observers existed
and in which case the estimation would be ignored.
- Add MaybeTriggerOnNetworkChanged which only runs rate allocation if any of the dependent variables has changed
thus allowing to remove many of the bool returns that try to indicate if the estimation has changed which would not
be aware if the observers have changed.
- SendSideBandwidthEstimation now has a UpdateBitrate and has clear code paths to which calls update bitrate.
- Changes in enforce_min_bitrate so the 10kbps min is set from the BitrateController and not from the outside this keep valid as observers are changed.
R=henrik.lundin@webrtc.org, stefan@webrtc.org
BUG=3065
Review URL: https://webrtc-codereview.appspot.com/10189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5752 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 29dd0de5b35f757289895b7af7c9c5905878270f |
|
21-Oct-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Changing the bitrate clamping in BitrateControllerImpl
This CL implements an alternative to the bitrate clamping that is done
in BitrateControllerImpl. The default behavior is unchanged, but if
the new algorithm is enabled the behavior is as follows:
When the new bitrate is lower than the sum of min bitrates, the
algorithm will give each observer up to its min bitrate, one
observer at a time, until the bitrate budget is depleted. Thus,
with this change, some observers may get less than their min bitrate,
or even zero.
Unit tests are implemented.
Also fixing two old lint warnings in the affected files.
This change is related to the auto-muter feature.
BUG=2436
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2439005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5007 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 28a331eededf17dc3a0860bb1bdf5b2dc3f9e763 |
|
17-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add support for multiple report blocks.
Use a weighted average of fraction loss for bandwidth estimation.
TEST=trybots and vie_auto_test --automated
BUG=1811
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2198004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4762 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 2e10b8e4a05c041a3c39ddc7499bdffdb67999fa |
|
16-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in bitrate_controller/.
BUG=1662
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1787004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4349 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
|