16 Commits

Author SHA1 Message Date
kthelgason
b906172e02 Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ )
Reason for revert:
Internal project has been fixed

Original issue's description:
> Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ )
>
> Reason for revert:
> Breaks internal project
>
> Original issue's description:
> > Move current bitstream parser to more appropriate directory.
> >
> > This CL groups together the code that has to do with parsing H264 bitstreams.
> > This code logically belongs together, and having it in the same directory not
> > only simplifies things from a project structure perspective, but also makes it
> > easier to refactor out common parts incrementally.
> > An added benefit is that this simplifies modular compilation, where for example
> > one would like a build of WebRTC without the H264 codec-specific parts.
> >
> > BUG=webrtc:6338
> >
> > Committed: https://crrev.com/cc6817e9ce4a5ffc73efb660cf0368afbc7d9a4f
> > Cr-Commit-Position: refs/heads/master@{#14684}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6338
>
> Committed: https://crrev.com/f04f14e772f803de39f8a6128e5157127cd35103
> Cr-Commit-Position: refs/heads/master@{#14685}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6338

Review-Url: https://codereview.webrtc.org/2434043002
Cr-Commit-Position: refs/heads/master@{#14783}
2016-10-26 09:48:24 +00:00
kthelgason
f04f14e772 Revert of Move bitstream parser to more appropriate directory. (patchset #4 id:60001 of https://codereview.webrtc.org/2370853005/ )
Reason for revert:
Breaks internal project

Original issue's description:
> Move current bitstream parser to more appropriate directory.
>
> This CL groups together the code that has to do with parsing H264 bitstreams.
> This code logically belongs together, and having it in the same directory not
> only simplifies things from a project structure perspective, but also makes it
> easier to refactor out common parts incrementally.
> An added benefit is that this simplifies modular compilation, where for example
> one would like a build of WebRTC without the H264 codec-specific parts.
>
> BUG=webrtc:6338
>
> Committed: https://crrev.com/cc6817e9ce4a5ffc73efb660cf0368afbc7d9a4f
> Cr-Commit-Position: refs/heads/master@{#14684}

TBR=magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6338

Review-Url: https://codereview.webrtc.org/2430353004
Cr-Commit-Position: refs/heads/master@{#14685}
2016-10-19 17:34:39 +00:00
kthelgason
cc6817e9ce Move current bitstream parser to more appropriate directory.
This CL groups together the code that has to do with parsing H264 bitstreams.
This code logically belongs together, and having it in the same directory not
only simplifies things from a project structure perspective, but also makes it
easier to refactor out common parts incrementally.
An added benefit is that this simplifies modular compilation, where for example
one would like a build of WebRTC without the H264 codec-specific parts.

BUG=webrtc:6338

Review-Url: https://codereview.webrtc.org/2370853005
Cr-Commit-Position: refs/heads/master@{#14684}
2016-10-19 16:31:15 +00:00
kthelgason
194f40a2e7 Refactor QualityScaler and MovingAverage
The MovingAverage class was very specific to the QualityScaler. This
commit generalizes the MovingAverage class to be useful in other
situations as well, and adapts the QualityScaler to use the new
MovingAverage.

BUG=webrtc:6304

Review-Url: https://codereview.webrtc.org/2310853002
Cr-Commit-Position: refs/heads/master@{#14207}
2016-09-14 09:15:02 +00:00
Erik Språng
78ce619a0c Extract simulcast rate allocation outside of video encoder.
This is a first step to refactor this code.
I'm deprecating https://codereview.webrtc.org/1913073002 and
implementing this in smaller more isolated steps.

BUG=webrtc:5206
R=asapersson@webrtc.org, kjellander@webrtc.org, noahric@chromium.org

Review URL: https://codereview.webrtc.org/2288223002 .

Cr-Commit-Position: refs/heads/master@{#14186}
2016-09-12 14:04:56 +00:00
perkj
4e417b242a Reland of Switch to use SequencedTaskChecker instead of ThreadChecker where needed.
(patchset #1 id:1 of https://codereview.webrtc.org/2149553002/ )"
This reverts commit efd902cb1d9bbd81247a3e168f2080beae761d78.

Originally reviewed in https://codereview.webrtc.org/2149553002

The uptream problem should be fixed by https://codereview.webrtc.org/2145393003/

BUG=webrtc:5687
TBR=tommi@webrtc.org

Review-Url: https://codereview.webrtc.org/2152013002
Cr-Commit-Position: refs/heads/master@{#13483}
2016-07-15 06:36:00 +00:00
Peter Boström
cc1543abf3 Move H264BitstreamParser to video_coding.
Moves parser, used in video_coding/ from rtp_rtcp where it is unused.

BUG=webrtc:5678
R=asapersson@webrtc.org
TBR=glaznev@webrt.org

Review URL: https://codereview.webrtc.org/2007553003 .

Cr-Commit-Position: refs/heads/master@{#12866}
2016-05-24 10:16:39 +00:00
sprang
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
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
asapersson
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
jackychen
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
jackychen
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
andresp@webrtc.org
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
pbos@webrtc.org
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
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
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