Commit Graph

9 Commits

Author SHA1 Message Date
3911c26bc0 Add support for writing raw encoder output to .ivf files.
Also refactor GenericEncoder to use these file writers, and remove use
of preprocessor to enable file writing.

BUG=

Review URL: https://codereview.webrtc.org/1853813002

Cr-Commit-Position: refs/heads/master@{#12372}
2016-04-15 08:24:21 +00:00
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