Removes use of global VP8EncoderFactory::use_simulcast_adapter which is
thread-unsafe. Also the code wasn't in use.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1598803005 .
Cr-Commit-Position: refs/heads/master@{#11370}
The FFmpeg video decoder requires up to 8 additional bytes to be allocated for its encoded image buffer input, due to optimized byte readers over-reading on some platforms.
We plan to use FFmpeg for a soon-to-land H.264 enc/dec.
This CL adds support for padding encoded image buffers based on codec type, and makes sure calls to VCMEncodedFrame::VerifyAndAllocate use the padding.
All padding constants are 0 but making H.264 pad with 8 bytes will be a one-line change.
Also, added -framework CoreFoundation to webrtc_h264_video_toolbox which was missing.
BUG=chromium:468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
NOTRY=True
Review URL: https://codereview.webrtc.org/1602523004
Cr-Commit-Position: refs/heads/master@{#11337}
Issue may occur for very small input images (e.g. 4x4) when encoded image length > input image size.
BUG=chromium:578193
Review URL: https://codereview.webrtc.org/1603643006
Cr-Commit-Position: refs/heads/master@{#11329}
Provides a better string (provides names of all implementations), but
also fixes a crash when accessing the ImplementationName() of
SimulcastEncoderAdapter where InitEncode has failed.
BUG=chromium:577932, webrtc:4897
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1599353003 .
Cr-Commit-Position: refs/heads/master@{#11321}
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}
Reason for revert:
Reverting due to problem with roll:
/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
-> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
configs -= [ "//build/config/clang:find_bad_constructs" ]
^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@
Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}
TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1586563003
Cr-Commit-Position: refs/heads/master@{#11239}
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Review URL: https://codereview.webrtc.org/1538643004
Cr-Commit-Position: refs/heads/master@{#11231}
Tests were failing on android with new libvpx.
vp9 speed setting was changed to 8 recently and some recent changes
in libvpx require update for the tests to pass.
TBR=stefan@webrtc.org
BUG=webrtc:5401
Review URL: https://codereview.webrtc.org/1569903002 .
Cr-Commit-Position: refs/heads/master@{#11173}
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.orgTBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
At speed 8, vp9 on ARM is currently ~2x times slower than vp8 on ARM (speed -12).
Update some parameters in videoprocessor_integrationtest.cc
to make tests pass on android (which uses the new speed setting).
TBR=stefan@webrtc.org
BUG=
Review URL: https://codereview.webrtc.org/1526973004 .
Cr-Commit-Position: refs/heads/master@{#11072}
Specify kf_min_dist to get correct key frame interval in svc mode.
Also set QP-max/min per temporal and spatial layer (was previously only allowed to be set per spatial layer).
BUG=chromium:500602
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1492633005 .
Cr-Commit-Position: refs/heads/master@{#10890}
The callback keeps a reference to an object until the callback goes out of scope.
Review URL: https://codereview.webrtc.org/1487493002
Cr-Commit-Position: refs/heads/master@{#10847}
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.orgTBR=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1440873005 .
Cr-Commit-Position: refs/heads/master@{#10699}
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}
To avoid breaking downstream, the "interface" directories were copied
into a new "common_video/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).
The header guards are also identical to avoid mixing them up in the transition.
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
Review URL: https://codereview.webrtc.org/1418913006
Cr-Commit-Position: refs/heads/master@{#10659}
Reason for revert:
Failed test not related to this CL (test fails on
master at an earlier date), re-landing original CL..
(This time from my @webrtc account.)
Original issue's description:
> Revert of Work on flexible mode and screen sharing. (patchset #28 id:520001 of https://codereview.webrtc.org/1328113004/ )
>
> Reason for revert:
> Seems to break VideoSendStreamTest.ReconfigureBitratesSetsEncoderBitratesCorrectly on Linux Memcheck buildbot.
>
> Original issue's description:
> > Work on flexible mode and screen sharing.
> >
> > Implement VP8 style screensharing but with spatial layers.
> > Implement flexible mode.
> >
> > Files from other patches:
> > generic_encoder.cc
> > layer_filtering_transport.cc
> >
> > BUG=webrtc:4914
> >
> > Committed: https://crrev.com/77ccfb4d16c148e61a316746bb5d9705e8b39f4a
> > Cr-Commit-Position: refs/heads/master@{#10572}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,philipel@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:4914
>
> Committed: https://crrev.com/0be8f1d347bdb171462df89c2a4c69b3f3eb7519
> Cr-Commit-Position: refs/heads/master@{#10578}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@google.com,asapersson@webrtc.org,mflodman@webrtc.org,terelius@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4914
Review URL: https://codereview.webrtc.org/1431283002
Cr-Commit-Position: refs/heads/master@{#10581}
On Android, we would like to use MediaCodec output buffers to hold decoded frames until they can be rendered to a texture. There can only be one texture buffer used at the same time and therefore the calculated decode time in VCMTiming will be wrong since that calculation will also include the time where the decoder waited for the upper layers (that depend on network jitter and actual render time) to release the frame.
This new method will be used in
https://codereview.webrtc.org/1422963003/
BUG=webrtc:4993
R=stefan@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1414693006 .
Cr-Commit-Position: refs/heads/master@{#10576}
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}
This is a re-land of https://codereview.webrtc.org/1353263005/
which was reverted because of perf-regressions. Changes since that CL:
* Change LayerFilteringTransport to send a padding packet instead of
dropping it for data that should be filtered out. This prevents
confusion due to changed sequence numbers.
* Changed timing of stats poller thread in VideoAnalyzer. Startup was
racy wrt initializion of send_stream_.
* Minor formatting issues.
PERF NOTE: This change will affect some performance numbers slightly.
In particular, {encode_frame_rate, encode_time_ms,
encode_usage_percent, media_bitrate_bps} will change due to timing
of the measurements.
BUG=
R=pbos@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1412233003
Cr-Commit-Position: refs/heads/master@{#10483}
- "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}
