11 Commits

Author SHA1 Message Date
isheriff
4d7bc240b9 Fix valgrind complaint on uninitialized value
BUG=

Review URL: https://codereview.webrtc.org/1883223002

Cr-Commit-Position: refs/heads/master@{#12460}
2016-04-21 23:37:26 +00:00
noahric
91efeecab8 Remove VERBOSE logs from frame_dropper.cc.
They are way too verbose, ~100 lines of log per second.

BUG=

Review URL: https://codereview.webrtc.org/1888453004

Cr-Commit-Position: refs/heads/master@{#12356}
2016-04-14 04:01:48 +00:00
isheriff
7620be8492 Frame dropper improvements & cleanup
1. Fix the case of key frame accumulation being incorrect due to the chunk
    size being computed at the time of leak based on input frame rate. The issue
    is that the count is computed based on key frame ratio and the actual chunk
    size computed from current input frame rate. These can be wildly different
    especially at the beginning of the stream (key frame ratio defaults based
    on 30 fps) resulting in incorrect key frame accumulation causing large frame
    drops when the input frame rate is low.

    2. Add large delta frame compensation. The current code accounts for key frames
    but not large delta frames. This is a common occurence in some application
    (remote desktop as an example)

    3. Fixes an issue identified by the unit tests. The accumulation of
    key frames had an issue in the scenario of a high key frame ratio where
    the full key frame was not being accounted for.

    3. Removes fast mode and other methods that are mostly dead code.

    4. Cleans up variable names as per chromium style.

Review URL: https://codereview.webrtc.org/1750493002

Cr-Commit-Position: refs/heads/master@{#11884}
2016-03-07 07:22:42 +00:00
philipel
5908c71128 Lint fix for webrtc/modules/video_coding PART 3!
Trying to submit all changes at once proved impossible since there were
too many changes in too many files. The changes to PRESUBMIT.py
will be uploaded in the last CL.
(original CL: https://codereview.webrtc.org/1528503003/)

BUG=webrtc:5309
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1540243002

Cr-Commit-Position: refs/heads/master@{#11105}
2015-12-21 16:23:29 +00:00
kjellander@webrtc.org
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
Henrik Kjellander
98f53510b2 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}
2015-10-28 17:17:50 +00:00
pkasting@chromium.org
4591fbd09f 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
2014-11-20 22:28:14 +00:00
minyue@webrtc.org
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
pbos@webrtc.org
034f004a4f WebRtc_Word32 => int32_t in video_coding/
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1203008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3778 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-08 11:13:29 +00:00
stefan@webrtc.org
84cd8e39cf Disable frame dropper for screenshare mode.
BUG=1466

Review URL: https://webrtc-codereview.appspot.com/1170004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3629 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-07 13:12:32 +00:00
stefan@webrtc.org
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