b7ce96470b
modules/video_coding/utility: Remove include
...
This makes it clearer this code not meant to be used as an API.
I could not find any use of this in downstream code.
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=stefan@webrtc.org
TBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440873005 .
Cr-Commit-Position: refs/heads/master@{#10699}
2015-11-18 22:04:20 +00:00
86b016027d
Add stats for average QP per frame for VP8 (for received video streams):
...
"WebRTC.Video.Decoded.VP8.Qp"
BUG=chromium:512752
Review URL: https://codereview.webrtc.org/1340623002
Cr-Commit-Position: refs/heads/master@{#10349}
2015-10-21 06:55:32 +00:00
98d8cf58ee
Hardware VP8 encoding: Use QP as metric for resize.
...
Add vp8 frame header parser to get QP from vp8 bitstream.
BUG= 4273
R=glaznev@webrtc.org , marpan@google.com , pbos@webrtc.org
TBR=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49259004
Cr-Commit-Position: refs/heads/master@{#9256}
2015-05-21 18:11:53 +00:00
61b4d518af
Dynamic resolution change for VP8 HW encode.
...
Off by default for now.
BUG=
R=glaznev@webrtc.org , stefan@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45849004
Cr-Commit-Position: refs/heads/master@{#9045}
2015-04-21 22:29:53 +00:00
86e1e487e7
Move system_wrappers.gyp files to the proper directory.
...
Build targets should not refer to non-subpaths as was happening before when
source/system_wrappers.gyp refers to ../interface/ files.
R=kjellander@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 09:30:52 +00:00
a0d7827b16
Add ability to downscale content to improve quality.
...
BUG=3712
R=marpan@google.com , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7164 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 11:51:47 +00:00
74aaf29a0f
Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
...
The filter is an exponential filter borrowed from video coding module.
The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.
BUG=
R=henrika@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:28:26 +00:00
eb91792cfd
Refactoring temporal layers implementation and adding VideoCodecMode for easier control of codec settings.
...
Review URL: https://webrtc-codereview.appspot.com/1105007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3528 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-18 14:40:18 +00:00
