Constructing default options is racy when initializing multiple VP8
encoders in parallel. This is only used for VP8 temporal layers. Adding
TemporalLayerFactory to VP8 codec specifics instead of generic options.
Removes the last webrtc::Config uses/includes from video code.
Also removes VideoCodec equality operators which are no longer in use.
BUG=webrtc:5410
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1606613003 .
Cr-Commit-Position: refs/heads/master@{#11307}
The main purpose was the interface-> include rename, but other files
were also moved, eliminating the "main" dir.
To avoid breaking downstream, the "interface" directories were copied
into a new "video_coding/include" dir. The old headers got pragma
warnings added about deprecation (a very short deprecation since I plan
to remove them as soon downstream is updated).
Other files also moved:
video_coding/main/source -> video_coding
video_coding/main/test -> video_coding/test
BUG=webrtc: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 --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, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417283007 .
Cr-Commit-Position: refs/heads/master@{#10694}
Removes VP8::Encode trace in favor of VCMGenericEncoder ones and adds
one to InitEncode. Also adds an instant event to ::Encoded since this
can be done on a different thread.
Also adds the corresponding traces to VCMGenericDecoder.
BUG=webrtc:5167
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1412573010
Cr-Commit-Position: refs/heads/master@{#10674}
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}
- "WebRTC.Video.BandwidthLimitedResolutionInPercent"
If the frame is bandwidth limited, the average number of disabled resolutions is logged:
- "WebRTC.Video.BandwidthLimitedResolutionsDisabled"
BUG=
Review URL: https://codereview.webrtc.org/1311533012
Cr-Commit-Position: refs/heads/master@{#10333}
- "WebRTC.Video.QualityLimitedResolutionInPercent"
and if a frame is downscaled, the average number of times the frame is downscaled:
- "WebRTC.Video.QualityLimitedResolutionDownscales"
BUG=
Review URL: https://codereview.webrtc.org/1325153009
Cr-Commit-Position: refs/heads/master@{#10319}
Since padding is no longer sent on Encoded() callbacks, dummy callbacks
aren't required to generate padding. This skip-frame behavior can then
be removed to get rid of dummy callbacks though nothing was encoded. As
frames don't have to be generated for frames that don't have to be sent
we skip encoding frames that aren't intended to be sent either, reducing
CPU load.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1369923005 .
Cr-Commit-Position: refs/heads/master@{#10181}
This CL changes the threshold where we consider a block to be static and
of sufficient quality to not spend bits/CPU encoding it.
Perf note: This change may result in a minor degradation of PSNR/SSIM
and available send bitrate. CPU usage and bitrate sent should however
be greately reduced.
BUG=webrtc:5015
Review URL: https://codereview.webrtc.org/1383533002
Cr-Commit-Position: refs/heads/master@{#10134}
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.
BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1364253002 .
Cr-Commit-Position: refs/heads/master@{#10078}
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}
This change includes several improvements:
* VP8 configured with new rate control
* Detection of frame dropping, with qp bump for next frame
* Increased target and TL0 bitrates
* Reworked rate control (TL allocation) in screenshare_layers
A note on performance: PSNR and SSIM is expected to get slightly worse with this cl. Frame drops and delays should however improve.
BUG=4171
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1193513006.
Cr-Commit-Position: refs/heads/master@{#9495}
Now when we don't use SwapFrame consistently anymore, we need to recycle allocations with a buffer pool instead. This CL adds a buffer pool class, and updates the vp8 decoder to use it. If this CL lands successfully I will update the other video producers as well.
BUG=1128
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41189004
Cr-Commit-Position: refs/heads/master@{#8748}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8748 4adac7df-926f-26a2-2b94-8c16560cd09d
Allow for setting different cpu_speed setting based on resolution, for non-simulcast.
Use the existing low resolution simulcast cpu_speed setting for the non-simulcast case.
No change to simulcast behavior, unless top/highest layer stream is also below CIF resolution,
(in which case all layers will use lower the cpu_speed setting =-4).
BUG=
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37319004
Cr-Commit-Position: refs/heads/master@{#8603}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8603 4adac7df-926f-26a2-2b94-8c16560cd09d
