Reason for revert:
Seems to be causing flakiness in perf test:
FullStackTest.ScreenshareSlidesVP8_2TL_LossyNet
Original issue's description:
> Reland of Issue 2434073003: Extract bitrate allocation ...
>
> This is a reland of https://codereview.webrtc.org/2434073003/ including
> some fixes for failing test cases.
>
> Original description:
>
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/647bf43dcb2fd16fccf276bd94dc4400728bb405
> Cr-Commit-Position: refs/heads/master@{#15023}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2491393002
Cr-Commit-Position: refs/heads/master@{#15026}
This is a reland of https://codereview.webrtc.org/2434073003/ including
some fixes for failing test cases.
Original description:
Extract bitrate allocation of spatial/temporal layers out of codec impl.
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2488833004
Cr-Commit-Position: refs/heads/master@{#15023}
Reason for revert:
Breaks perf tests.
Original issue's description:
> Extract bitrate allocation of spatial/temporal layers out of codec impl.
>
> This CL makes a number of intervowen changes:
>
> * Add BitrateAllocation struct, that contains a codec independent view
> of how the target bitrate is distributed over spatial and temporal
> layers.
>
> * Adds the BitrateAllocator interface, which takes a bitrate and frame
> rate and produces a BitrateAllocation.
>
> * A default (non layered) implementation is added, and
> SimulcastRateAllocator is extended to fully handle VP8 allocation.
> This includes capturing TemporalLayer instances created by the
> encoder.
>
> * ViEEncoder now owns both the bitrate allocator and the temporal layer
> factories for VP8. This allows allocation to happen fully outside of
> the encoder implementation.
>
> This refactoring will make it possible for ViEEncoder to signal the
> full picture of target bitrates to the RTCP module.
>
> BUG=webrtc:6301
>
> Committed: https://crrev.com/8f46c679d24a05b3f08e02c6d91ec9637f34e24f
> Cr-Commit-Position: refs/heads/master@{#14998}
TBR=stefan@webrtc.org,perkj@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2489843002
Cr-Commit-Position: refs/heads/master@{#15001}
This CL makes a number of intervowen changes:
* Add BitrateAllocation struct, that contains a codec independent view
of how the target bitrate is distributed over spatial and temporal
layers.
* Adds the BitrateAllocator interface, which takes a bitrate and frame
rate and produces a BitrateAllocation.
* A default (non layered) implementation is added, and
SimulcastRateAllocator is extended to fully handle VP8 allocation.
This includes capturing TemporalLayer instances created by the
encoder.
* ViEEncoder now owns both the bitrate allocator and the temporal layer
factories for VP8. This allows allocation to happen fully outside of
the encoder implementation.
This refactoring will make it possible for ViEEncoder to signal the
full picture of target bitrates to the RTCP module.
BUG=webrtc:6301
Review-Url: https://codereview.webrtc.org/2434073003
Cr-Commit-Position: refs/heads/master@{#14998}
Reason for revert:
Fix gyp build
Original issue's description:
> Revert of Add a webrtc{en,de}coderfactory implementation for VideoToolbox (patchset #2 id:20001 of https://codereview.webrtc.org/2463313002/ )
>
> Reason for revert:
> Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes.
>
> Original issue's description:
> > Add a webrtc{en,de}coderfactory implementation for VideoToolbox
> >
> > This CL removes the coupling of the VideoToolbox h264 implementation
> > to the generic h264 code. The files have been moved into sdb/obj/Framework
> > and all dependency on them has been removed from the rest of WebRTC.
> > We now add it as an external encoder via a factory supplied to the
> > CreatePeerConnectionFactory call. This also brings the iOS implementation
> > closer to what we do on Android for MediaCodec.
> >
> > BUG=webrtc:6619
> >
> > Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4
> > Cr-Commit-Position: refs/heads/master@{#14953}
>
> TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
> BUG=webrtc:6619
>
> Committed: https://crrev.com/d69ad84420d9c0e1c11450c352f6c92e7c9583f1
> Cr-Commit-Position: refs/heads/master@{#14985}
R=magjed@webrtc.orgTBR=kwiberg@webrtc.org, 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:6619
Review URL: https://codereview.webrtc.org/2487723004 .
Cr-Commit-Position: refs/heads/master@{#14992}
Reason for revert:
Broke dependent project because the .gn changes weren't accompanied by corresponding .gyp changes.
Original issue's description:
> Add a webrtc{en,de}coderfactory implementation for VideoToolbox
>
> This CL removes the coupling of the VideoToolbox h264 implementation
> to the generic h264 code. The files have been moved into sdb/obj/Framework
> and all dependency on them has been removed from the rest of WebRTC.
> We now add it as an external encoder via a factory supplied to the
> CreatePeerConnectionFactory call. This also brings the iOS implementation
> closer to what we do on Android for MediaCodec.
>
> BUG=webrtc:6619
>
> Committed: https://crrev.com/6a5047dad31f14f53dd9f8bc1ecde19e1dede2e4
> Cr-Commit-Position: refs/heads/master@{#14953}
TBR=magjed@webrtc.org,stefan@webrtc.org,kthelgason@webrtc.org
BUG=webrtc:6619
Review-Url: https://codereview.webrtc.org/2483273002
Cr-Commit-Position: refs/heads/master@{#14985}
This CL removes the coupling of the VideoToolbox h264 implementation
to the generic h264 code. The files have been moved into sdb/obj/Framework
and all dependency on them has been removed from the rest of WebRTC.
We now add it as an external encoder via a factory supplied to the
CreatePeerConnectionFactory call. This also brings the iOS implementation
closer to what we do on Android for MediaCodec.
BUG=webrtc:6619
Review-Url: https://codereview.webrtc.org/2463313002
Cr-Commit-Position: refs/heads/master@{#14953}
The H264SpsPpsTracker class:
- Keeps track of all received SPS/PPS.
- Decides whether a packet should be inserted into the PacketBuffer or not.
- Don't insert if this packet only contains SPS and/or PPS.
- Don't insert if this is the first packet of and IDR and we have not
received the required SPS/PPS.
- Insert start codes, and in the case of the first packet of an IDR prepend
the bitstream with the given SPS/PPS for this IDR.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2466993003
Cr-Commit-Position: refs/heads/master@{#14906}
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}
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}
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}
When rtc_build_libyuv=false an incorrect code path
is surfaced in GN.
BUG=webrtc:6412
NOTRY=True
TESTED=gn gen out/foo --args='rtc_build_libyuv=false target_os="ios"'
Review-Url: https://codereview.webrtc.org/2375603002
Cr-Commit-Position: refs/heads/master@{#14392}
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
In the migration to GN templates, some targets got the whole
rtc_common_config removed, which can have unpredicted consequences
in terms of different code behavior due to defines not being set
as expected etc.
It's better to enable this config and only disable the warnings
that fails the build.
BUG=webrtc:6306,webrtc:6307,webrtc:6308
NOTRY=True
Review-Url: https://codereview.webrtc.org/2347263002
Cr-Commit-Position: refs/heads/master@{#14280}
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}
Remove common_inherited_config from the targets and add it to the
template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2311843002
Cr-Commit-Position: refs/heads/master@{#14069}
Remove common_config from the targets' config and add
it to the template instead.
BUG=webrtc:6187
NOTRY=True
Review-Url: https://codereview.webrtc.org/2300413002
Cr-Commit-Position: refs/heads/master@{#14063}
Defines the rtc_executable, rtc_source_set, rtc_test and
rtc_static_library templates.
These templates provide no functionality yet, but will enable common
configuration to be introduced, avoiding repetition in every target
Changes summary:
- Prepend rtc_ to test, source_set, executable and static_library targets
- Change "configs -= [" to "suppressed_configs += ["
- Include webrtc/build/webrtc.gni where it wasn't included yet
- Delete import("//testing/test.gni"), since rtc_test makes it unnecessary.
BUG=webrtc:6187
TBR=henrik.lundin@webrtc.org,tommi@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2301053002
Cr-Commit-Position: refs/heads/master@{#14043}
Scaling causes us to work the CPU too much, which very quickly degrades quality. This causes us to at least behave better on good networks.
NOTRY=True
BUG=
Review-Url: https://codereview.webrtc.org/2205763002
Cr-Commit-Position: refs/heads/master@{#13630}
Changes:
* Enabled protobuf for iOS globally.
* Set WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE on a global
scope similar to GYP since tests depend on it.
* Added missing rtc_libvpx_build_vp9 variable.
* Moved out audio_coding defines into .gni file to avoid code duplication
* Renamed files to avoid object naming conflicts that GN disallows:
* webrtc/modules/audio_processing/{echo_cancellation_unittest.cc->echo_cancellation_bit_exact_unittest.cc}
* webrtc/modules/video_coding/codecs/vp9/{screenshare_layers_unittest.cc->vp9_screenshare_layers_unittest.cc}
BUG=webrtc:5949
TESTED=Built and ran the tests on Mac. Also ran:
gn gen out/Default --args="rtc_enable_bwe_test_logging=true"
and verified that more objects are being built (1885 vs 1883)
when compiling modules_unittests.
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2041233006
Cr-Commit-Position: refs/heads/master@{#13108}
In https://codereview.webrtc.org/2034923003 it was discovered
that a test binary rtc_sdk_peerconnection_objc_tests was
a dependency to rtc_unittests. Unfortunately gtest doesn't
include dependent executables into the same test executable;
only libraries (so theses tests weren't run).
This CL incorporates those tests into rtc_unittests and
does the same changes to the GN build.
BUG=webrtc:5949
TESTED=Built and ran rtc_unittests locally on Mac.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2041743003
Cr-Commit-Position: refs/heads/master@{#13060}
This cl split the class MediaOptimization into two parts. One that deals with frame dropping and stats and one new class called ProtectionBitrateCalculator that deals with calculating the needed FEC parameters and how much of the estimated network bitrate that can be used by an encoder
Note that the logic of how FEC and the needed bitrates is not changed.
BUG=webrtc:5687
R=asapersson@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1972083002 .
Cr-Commit-Position: refs/heads/master@{#13018}
This makes the GN configurations easier to read.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2020343003
Cr-Commit-Position: refs/heads/master@{#13006}
Compile fixes for GN on iOS that finally gets our bots green.
Changes to system_wrappers:
* Updated to only use inclusive sources for maintainability
* Add a few missing GN headers.
* Cleanup GYP hack for atomic32_mac.cc
* Renamed changes sources to avoid problems with GYP/GN file
suffix rules:
- atomic32_mac.cc -> atomic32_darwin.cc
- atomic32_posix.cc -> atomic32_non_darwin_unix.cc
See https://code.google.com/p/chromium/codesearch#chromium/src/build/config/BUILDCONFIG.gn&l=325
for details on which extensions can/cannot be used.
BUG=webrtc:5586
NOTRY=True
Review-Url: https://codereview.webrtc.org/1999723002
Cr-Commit-Position: refs/heads/master@{#12897}
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}
This CL is expected to lower goog_max_decode_ms and total_delay_incl_network/receiver_time for screenshare.
Reason for revert:
This CL did not cause the unexpected goog_encode_usage_percent and goog_avg_encode_ms perf changes.
Original issue's description:
> Revert of VCMCodecTimer: Change filter from max to 95th percentile (patchset #5 id:180001 of https://codereview.webrtc.org/1742323002/ )
>
> Reason for revert:
> Caused unexpected perf stats changes, see http://crbug/594575.
>
> Original issue's description:
> > VCMCodecTimer: Change filter from max to 95th percentile
> >
> > The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
> >
> > BUG=b/27306053
> >
> > Committed: https://crrev.com/4bf0c717740d1834e810ea5f32b3c4306c64235f
> > Cr-Commit-Position: refs/heads/master@{#11952}
>
> TBR=stefan@webrtc.org,philipel@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=594575,b/27306053
>
> Committed: https://crrev.com/c4a74e95b545f4752d4e72961ac03c1380d4bc1f
> Cr-Commit-Position: refs/heads/master@{#12018}
TBR=stefan@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=594575,b/27306053
Review URL: https://codereview.webrtc.org/1824763003
Cr-Commit-Position: refs/heads/master@{#12087}
Reason for revert:
Caused unexpected perf stats changes, see http://crbug/594575.
Original issue's description:
> VCMCodecTimer: Change filter from max to 95th percentile
>
> The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
>
> BUG=b/27306053
>
> Committed: https://crrev.com/4bf0c717740d1834e810ea5f32b3c4306c64235f
> Cr-Commit-Position: refs/heads/master@{#11952}
TBR=stefan@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=594575,b/27306053
Review URL: https://codereview.webrtc.org/1808693002
Cr-Commit-Position: refs/heads/master@{#12018}
Testing the nack module by implementing it into the current jitter buffer
under the experiment WebRTC-NewVideoJitterBuffer.
BUG=webrtc:5514
Review URL: https://codereview.webrtc.org/1778503002
Cr-Commit-Position: refs/heads/master@{#11969}
The purpose with this change is to make the filter more robust against anomalies. googMaxDecodeMs is expected to drop a litte by this.
BUG=b/27306053
Review URL: https://codereview.webrtc.org/1742323002
Cr-Commit-Position: refs/heads/master@{#11952}
There were a couple of GN and GYP references that were incorrect in Chromium builds:
- GN references between WebRTC targets must be using relative paths, not absolute.
- GYP references between WebRTC targets must be using the <(webrtc_root)v variable
in order to be expanded to the correct path in a Chromium build.
NOTRY=True
TBR=hjon@webrtc.org, hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1681493002
Cr-Commit-Position: refs/heads/master@{#11521}
Renamed the WEBRTC_THIRD_PARTY_H264 macro to WEBRTC_USE_H264 to match flag name.
The idea is to be able to turn off H264 from chromium with this function because...
1) The Chromium trybots will soon use this flag, we want to temporarily disable H264 from chromium even if flag is set in case something is broken. That way when we are ready to flip the switch the trybots will run our test code then and not after it is already enabled.
2) If feature is launched and we discover major problems we can easily disable H264 and merge with beta/stable.
3) Or, if feature is behind a *runtime* flag, this is how we would control if it is used or not.
The idea is to call DisableRtcUseH264 in chromium's PeerConnectionDependencyFactory.
BUG=chromium:500605, chromium:468365
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1657273002
Cr-Commit-Position: refs/heads/master@{#11474}
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.
In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.
BUG=chromium:500605, chromium:468365, webrtc:5427
Review URL: https://codereview.webrtc.org/1639273002
Cr-Commit-Position: refs/heads/master@{#11456}
It works on all platforms except Android and iOS (FFmpeg limitation).
Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.
Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)
Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)
NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
Review URL: https://codereview.webrtc.org/1306813009
Cr-Commit-Position: refs/heads/master@{#11390}
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}
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg).
These will be used in a follow-up CL that adds an encoder and decoder implementation.
The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added.
BUG=468365
Review URL: https://codereview.webrtc.org/1575913003
Cr-Commit-Position: refs/heads/master@{#11204}
