History log of /external/webrtc/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
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/remote_estimator_proxy.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/remote_estimator_proxy.h
233bd87d45bbeeec50d7687e7d98c1cfc7f65562 08-Sep-2015 sprang <sprang@webrtc.org> Add RemoteEstimatorProxy for capturing receive times

For use when send-side bandwidth estimation is enabled.

Receive times need to be captured, buffered and then sent using
TransportFeedback RTCP messaged back to the send side.

BUG=webrtc:4173

Review URL: https://codereview.webrtc.org/1290813008

Cr-Commit-Position: refs/heads/master@{#9898}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h