Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values
This cl
Revert "Revert of Replace interface VideoCapturerInput with VideoSinkInterface. (patchset #13 id:280001 of https://codereview.webrtc.org/2257413002/ )"
This reverts commit 9fdbda6aa3f66ea872344c22e79b23361047cbab.
and fix the problem in the original cl in video_quality_test.cc
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2348533002
Cr-Commit-Position: refs/heads/master@{#14265}
Reason for revert:
Fails on Mac and Linux webrtc_perf_tests
Original issue's description:
> Replace VideoCapturerInput with VideoSinkInterface.
> Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
>
> This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
>
> BUG=webrtc:5687
> // Android CQ seems broken.
> NOTRY=true
>
> Committed: https://crrev.com/95a226f55ae7e32b83a6ba96232fb105a014dc6c
> Cr-Commit-Position: refs/heads/master@{#14238}
TBR=nisse@webrtc.org,sprang@webrtc.org,mflodman@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:5687
Review-Url: https://codereview.webrtc.org/2344923002
Cr-Commit-Position: refs/heads/master@{#14239}
Adds new method VideoSendStream::SetSource(rtc::VideoSourceInterface* and VieEncoder::SetSource(rtc::VideoSourceInterface*)
This is the first step needed in order for the ViEEncoder to request downscaling using rtc::VideoSinkWants instead of separately reporting CPU overuse and internally doing downscaling due to QP values.
BUG=webrtc:5687
// Android CQ seems broken.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2257413002
Cr-Commit-Position: refs/heads/master@{#14238}
Remove a large number of targets that are no longer built, to reduce maintenance.
Only targets that have a GN version were removed.
BUG=webrtc:6323
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2340773003
Cr-Commit-Position: refs/heads/master@{#14231}
During GN vs GYP auditing it was discovered that some
GN targets that had public_configs were not exposing them
to dependents where the dependent depended on a group, which
in turn included that target as a dependency. Instead of
changing those public_configs to all_dependent_configs
(which would be a change from GYP), it's better to just change
those group targets to use public_deps instead.
BUG=webrtc:6323
NOTRY=True
TESTED=Generated GYP and GN project files on Mac and ran the
tools/gyp_flag_compare.py script before and after this patch was
applied. The file in question used for inspection was the
webrtc/api/webrtcsessiondescriptionfactory.cc
which is a part of the libjingle_peerconnection target.
Review-Url: https://codereview.webrtc.org/2344623002
Cr-Commit-Position: refs/heads/master@{#14222}
Declare resources for GN targets so that they can be isolated
NOTRY=True
BUG=chromium:497757
Review-Url: https://codereview.webrtc.org/2340753002
Cr-Commit-Position: refs/heads/master@{#14210}
Also enforce a minimum inter-frame interval of 1 ms,
fix a bug in the clipping logic, and improve comments.
BUG=webrtc:5740
Review-Url: https://codereview.webrtc.org/2325563002
Cr-Commit-Position: refs/heads/master@{#14206}
behavior of the audio processing module is quite complex and hard to
implement in a threadsafe and efficient manner. Therefore a new
scheme for setting the parameters in the audio processing module is
introduced in this CL.
The idea is to roll this scheme out gradually and as a first functionality
in the audio processing module where this is applied the level controller
was chosen. This CL includes the replacement of the Config-based
level controller scheme with the new scheme.
TBR=henrik.lundin@webrtc.org, solenberg@webrtc.org,
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2338493002
Cr-Commit-Position: refs/heads/master@{#14190}
Reason for revert:
Interface change in the mock breaks downstream code.
Original issue's description:
> The current scheme for setting parameters and specifying the behavior
> of the audio processing module is quite complex and hard to implement
> in a threadsafe and efficient manner. Therefore a new scheme for setting
> the parameters in the audio processing module is introduced in this CL.
>
> The idea is to roll this scheme out gradually and as a first functionality
> in the audio processing module where this is applied the level controller
> was chosen. This CL includes the replacement of the Config-based
> level controller scheme with the new scheme.
>
> BUG=webrtc:5298
>
> Committed: https://crrev.com/c8bbe3fe9aad9e9a1189a42dcaa8f5d6c261ecc8
> Cr-Commit-Position: refs/heads/master@{#14171}
TBR=solenberg@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org
BUG=webrtc:5298
NOTRY=True
Review-Url: https://codereview.webrtc.org/2334583002
Cr-Commit-Position: refs/heads/master@{#14177}
of the audio processing module is quite complex and hard to implement
in a threadsafe and efficient manner. Therefore a new scheme for setting
the parameters in the audio processing module is introduced in this CL.
The idea is to roll this scheme out gradually and as a first functionality
in the audio processing module where this is applied the level controller
was chosen. This CL includes the replacement of the Config-based
level controller scheme with the new scheme.
BUG=webrtc:5298
Review-Url: https://codereview.webrtc.org/2292863002
Cr-Commit-Position: refs/heads/master@{#14171}
Reason for revert:
Chromium build issues have been resolved.
Original issue's description:
> Revert of Remove all reference to carbon api (patchset #2 id:20001 of https://codereview.webrtc.org/2299633002/ )
>
> Reason for revert:
> Breaks chromium build
>
> Original issue's description:
> > Remove all reference to carbon api
> >
> > BUG=webrtc:6282
> >
> > Committed: https://crrev.com/dbd8b6bec4143c940b2f2ca8cd85c25d17327964
> > Cr-Commit-Position: refs/heads/master@{#14080}
>
> TBR=magjed@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:6282
>
> Committed: https://crrev.com/b096aa7fd375a980daab3a986596548ca5de2a1c
> Cr-Commit-Position: refs/heads/master@{#14081}
TBR=magjed@webrtc.org,mflodman@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6282
Review-Url: https://codereview.webrtc.org/2321493002
Cr-Commit-Position: refs/heads/master@{#14125}
RecreateWebRtcStream was checking the "sending_" flag, but wasn't
checking rtp_parameters_.encodings[0].active. As a result, if an
application calls "RtpSender.setParameters(inactive_params)" then later
the stream is recreated due to some change in SDP parameters, the stream
would incorrectly start sending.
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2246313002 .
Cr-Commit-Position: refs/heads/master@{#14116}
This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code.
- For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel().
- Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state.
BUG=webrtc:5879
Review-Url: https://codereview.webrtc.org/2307533004
Cr-Commit-Position: refs/heads/master@{#14109}
Reason for revert:
Breaks chromium build
Original issue's description:
> Remove all reference to carbon api
>
> BUG=webrtc:6282
>
> Committed: https://crrev.com/dbd8b6bec4143c940b2f2ca8cd85c25d17327964
> Cr-Commit-Position: refs/heads/master@{#14080}
TBR=magjed@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:6282
Review-Url: https://codereview.webrtc.org/2316563002
Cr-Commit-Position: refs/heads/master@{#14081}
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}
Replaces render_time_ms_, but old accessors are kept for
compatibility.
Also short-circuit timestamp translation in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame.
BUG=webrtc:5682, webrtc:5740
Review-Url: https://codereview.webrtc.org/2282713002
Cr-Commit-Position: refs/heads/master@{#14062}
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}
cl was originally reviewed here:
https://codereview.webrtc.org/2060403002/
- Add task queue to Call with the intent of replacing the use of one of the process threads.
- Split VideoSendStream in two. VideoSendStreamInternal is created and used on the new task queue.
- BitrateAllocator is now created on libjingle's worker thread but always used on the new task queue instead of both encoder threads and the process thread.
- VideoEncoderConfig and VideoSendStream::Config support move semantics.
- The encoder thread is moved from VideoSendStream to ViEEncoder. Frames are forwarded directly to ViEEncoder which is responsible for timestamping ? and encoding the frames.
TBR=mflodman@webrtc.org
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2250123002
Cr-Commit-Position: refs/heads/master@{#14014}
Users are updated to call libyuv functions directly. Also delete
related unit tests.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2287233002
Cr-Commit-Position: refs/heads/master@{#14013}
This test has been failing on TSan lately:
../../webrtc/media/base/videoengine_unittest.h:519: Failure
Value of: GetReceiverStats(i).frame_width
Actual: 0
Expected: DefaultCodec().width
Which is: 640
The root cause for the failure appears to be that the stats update
(https://cs.chromium.org/chromium/src/third_party/webrtc/video/video_receive_stream.cc?rcl=1472584967&l=353)
happens to be after the frame is passed to the renderer - while the test
is only waiting for the former.
The fix is to give it some extra time using EXPECT_EQ_WAIT instead.
Review-Url: https://codereview.webrtc.org/2299483002
Cr-Commit-Position: refs/heads/master@{#13991}
GetCopyWithRotationApplied is not yet deleted; downstream projects
must be updated first.
BUG=webrtc:5682
Review-Url: https://codereview.webrtc.org/2285693002
Cr-Commit-Position: refs/heads/master@{#13973}
In order to get resource files to be properly packaged into
the .app for a unit test on iOS, the resource files needs
to be listed as sources in a bundle_data target.
BUG=webrtc:5949
NOTRY=True
Review-Url: https://codereview.webrtc.org/2292853002
Cr-Commit-Position: refs/heads/master@{#13968}
Reason for revert:
This caused build breakage due to upstream dependencies.
These dependencies need to be resolved before landing the CL.
Original issue's description:
> This CL adds functionality in the level controller to
> receive a signal level to use initially, instead of the
> default initial signal level.
>
> BUG=
>
> Committed: https://crrev.com/57fec1d828113241186e78710ec5e851cc1a0e81
> Cr-Commit-Position: refs/heads/master@{#13931}
TBR=henrik.lundin@webrtc.org,aleloi@webrtc.org,solenberg@webrtc.org,henrika@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review-Url: https://codereview.webrtc.org/2283793002
Cr-Commit-Position: refs/heads/master@{#13936}
receive a signal level to use initially, instead of the
default initial signal level.
BUG=
Review-Url: https://codereview.webrtc.org/2254973003
Cr-Commit-Position: refs/heads/master@{#13931}
Unit test would fail in default configuration (e.g. rtc_use_h264=0), cause it tests instantiating H264 specifics.
BUG=webrtc:6194, webrtc:6198
Review-Url: https://codereview.webrtc.org/2228733004
Cr-Commit-Position: refs/heads/master@{#13929}
