| 13f6b8f7f4afe2316fa853f02732293d604a473f |
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21-Nov-2015 |
stefan <stefan@webrtc.org> |
Increase transport feedback frequency to 20 Hz.
BUG=4173
Review URL: https://codereview.webrtc.org/1466023002
Cr-Commit-Position: refs/heads/master@{#10736}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
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| 0b9e29c87da2d9c1a3792d2c87197b0688b68e4e |
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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/remote_bitrate_estimator/remote_estimator_proxy.cc
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| 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/remote_estimator_proxy.cc
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| 98f53510b222f71fdd8b799b2f33737ceeb28c61 |
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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/remote_bitrate_estimator/remote_estimator_proxy.cc
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| bbe876f0d30ec806c7c4a12629eb1f19ab45fb86 |
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23-Oct-2015 |
stefan <stefan@webrtc.org> |
Set send times in send time history via OnSentPacket.
BUG=webrtc:4173
Review URL: https://codereview.webrtc.org/1419503004
Cr-Commit-Position: refs/heads/master@{#10384}
/external/webrtc/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
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| 4fbd145dcefd23169a9b1612d5ca92dace8196d6 |
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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.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/remote_estimator_proxy.cc
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| 233bd87d45bbeeec50d7687e7d98c1cfc7f65562 |
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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.cc
|