| 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/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 4fbd145dcefd23169a9b1612d5ca92dace8196d6 |
|
28-Sep-2015 |
stefan <stefan@webrtc.org> |
Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 5e023eb337eed9746ecea7fc6fbb0fca386f1961 |
|
14-Sep-2015 |
sprang <sprang@webrtc.org> |
Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor
When using send-side bandwidth estimation, the inter-packet delay is
reported back to the sender using RTCP TransportFeedback messages.
Theis data needs to be translated into a format which the bandwidth
estimator (now instantiated on the send side) can use, including looking
up the local absolute send time from the send time history.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1329083005
Cr-Commit-Position: refs/heads/master@{#9929}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 11324b95612876e3335b20c4188bc9790d0b8539 |
|
09-Jul-2015 |
Stefan Holmer <stefan@webrtc.org> |
Wait for a longer time (5 seconds) before establishing the first bandwidth estimate.
This reduces the risk of getting a small initial estimate when doing combined a/v BWE, and the audio stream is received earlier than the video stream.
In addition a check is added to make sure a probe can't reduce the BWE.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1219303002 .
Cr-Commit-Position: refs/heads/master@{#9560}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 468e62a97426a8d001e9187f3ca1d1e43f80b970 |
|
06-Jul-2015 |
Erik Språng <sprang@webrtc.org> |
Remove MimdRateControl and factories for RemoteBitrateEstimor.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1208083002.
Cr-Commit-Position: refs/heads/master@{#9541}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| ff4ea9310e981da6292fb044cff9eeefd986cf2b |
|
18-Jun-2015 |
Stefan Holmer <stefan@webrtc.org> |
Only use paced packets for estimating bitrate probes.
BUG=4778
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1188823007.
Cr-Commit-Position: refs/heads/master@{#9463}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| fb609a1f57ef5aec3382bf28d8309154344d191d |
|
06-Feb-2015 |
stefan@webrtc.org <stefan@webrtc.org> |
Wire up new feedback format by introducing a FeedbackPacket type.
The new format instantiates the RemoteBitrateEstimator at the send-side and feeds back all packet arrival timestamps and sequence numbers to the sender, where inter-arrival deltas are calculated.
Next step will be to make feedback packets part of regular packets and send them over the network. This also requires bi-directional simulations.
BUG=4173
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37109004
Cr-Commit-Position: refs/heads/master@{#8264}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8264 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 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/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 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/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 5779ca478dd93193663b25a624749fe31e83de7a |
|
01-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a potential BWE clock mismatch bug.
Since libjingle provides a packet arrival timestamp to webrtc, and the clock in remote bitrate estimator and the clock used for packet arrival timestamp can be different. This can cause the bandwidth estimator to malfunction.
This CL changes the remote bitrate estimator so that packet arrival timestamps never are compared to the time taken from the internal clock.
BUG=3527
R=mflodman@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6571 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| af839b28b073be3c58a76433d7a4d96013e869f3 |
|
24-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| de782180b04adf606708a86567988fd64360aa3b |
|
12-Feb-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change the type of propagation delta from int64 to int.
The delta value never exceeds the range of int. Changing it to int will save memory and copying cost.
BUG=2910
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5537 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 1f64f067840212b3e5b67a0d6a50bcd805b5bc1a |
|
10-Feb-2014 |
jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add stats of incoming frame delays for debugging bandwidth estimation.
BUG=crbug/338380
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5519 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| e9abd591d73218e11a8bd3e7c72d4d7af9a3cea8 |
|
13-Dec-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Making RemoteRateControl::min_configured_bit_rate_ configurable
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
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 821510637147d6e4ea358db8f0415dd756bb14af |
|
21-Oct-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Framework for testing bandwidth estimation.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2317004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5008 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 91811e2b0457e091886508894a771f0e12054d0b |
|
25-Jun-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove unused multi stream bandwidth estimator.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1712004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4264 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| de98478965979e3de7578caae192c5110bc578ef |
|
04-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update the remote bitrate estimator before passing the packet to the RTP module.
This solves the problem of reconstructed packets biasing the bandwidth estimate.
TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1594005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| a6db54d4c9a2bfc703bc208eb5cbc19505e9cef3 |
|
27-May-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
- Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup.
- Changed implementation of SetReceiveAbsoluteSendTimeStatus API so the RBE instance is changed when at least one channel in a group has the extension enabled.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1553005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4113 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 561990fd73786354a1cdc6587029f333d79eb2a7 |
|
22-May-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
- Changed RemoteBitrateEstimator::IncomingPacket() to include a const WebRtcRTPHeader& and remove ssrc, rtp_timestamp.
- Changed RemoteBitrateObserver::OnReceivedBitrateChanged() to use a const & instead of non-const *, to avoid unnecessary copying.
- Refactored RemoteBitrateEstimatorTest so it can be instantiated for both single and multi stream BWE (first using a parameterized test, but then as a standard test fixture and a few helper functions).
- Refactored some tests in RemoteBitrateEstimatorTest into a common function CapacityDropTestHelper().
BUG=
R=andresp@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1521004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4086 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 29b2219914a059fe5164c312e7cc6d1bf0b4e610 |
|
14-May-2013 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding a factory to remote bitrate estimator and allow it to be set via config.
Additionally:
- clean api to set remote bitrate estimator mode.
- clean api to set over use detector options.
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1448006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4027 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| aea96d36e3691de0f95734f6d88bb94474903b34 |
|
19-Feb-2013 |
fischman@webrtc.org <fischman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Rename webrtc::StatsObserver to webrtc::CallStatsObserver
to avoid ODR violations with peerconnectioninterface.h in libjingle.
Conflict introduced in
https://webrtc-codereview.appspot.com/1060005/diff/14010/webrtc/modules/interface/module_common_types.h#newcode326
TEST=none
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1105011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3540 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| b5865079868c4dec49571e7aef0aa52124b50c64 |
|
01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages.
Also make sure RTT is computed independently of whether it's time to send RTCP messages or not.
BUG=1298
Review URL: https://webrtc-codereview.appspot.com/1060005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3455 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 4100b0402eea1fdea52e5899ee12e93c1f84b4db |
|
19-Nov-2012 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move SSRC list to RemoteBitrateEstimator.
BUG=1105
Review URL: https://webrtc-codereview.appspot.com/965027
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3130 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 42aa10eba7c89dc0b089078faa8dfcadc68366c1 |
|
13-Nov-2012 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Clarifies the bandwidth estimation interfaces.
BUG=
Review URL: https://webrtc-codereview.appspot.com/965019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3087 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|
| 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/remote_bitrate_estimator/include/remote_bitrate_estimator.h
|