WebRTC standalone build may depend on "video_capture_internal_impl"
instead of "video_capture". Including "video_capture" in public_deps
leads to duplicated definition in this case.
BUG=None
NOTRY=True
Review-Url: https://codereview.webrtc.org/2657783002
Cr-Commit-Position: refs/heads/master@{#16279}
Because Voice Engine was the only user.
(We have tried to land this many times before. I'm hoping that this
time all external dependencies on these files will really be gone.)
BUG=none
Review-Url: https://codereview.webrtc.org/2622493002
Cr-Commit-Position: refs/heads/master@{#15978}
Reason for reland:
Added disabling of the BlockProcessor DEATH tests due to false alarms on the trybots.
These errors cannot be reproduced locally and they occur only in the DEATH test code,
which means that the memory leaks do not occur under normal conditions.
Original issue's description:
> Reason for revert:
> Memcheck buildbot detected memory leaks in the death tests. The code looks fine
> to me, but please investigate what causes the error and reland.
> > Original issue's description:
> > Added first layer of the echo canceller 3 functionality.
> >
> > This CL adds the first layer of the echo canceller 3.
> > All of the code is as it should, apart from
> > block_processor.* which only contains placeholder
> > functionality. (Upcoming CLs will add proper
> > functionality into those files.)
> >
> >
> >
> > BUG=webrtc:6018
> >
> > Review-Url: https://codereview.webrtc.org/2584493002
> > Cr-Commit-Position: refs/heads/master@{#15861}
> > Committed:
> 38fd1758e9
>
> TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6018
BUG=webrtc:6018
TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org
Review-Url: https://codereview.webrtc.org/2608233002
Cr-Commit-Position: refs/heads/master@{#15884}
Reason for revert:
Memcheck buildbot detected memory leaks in the death tests. The code looks fine to me, but please investigate what causes the error and reland.
Original issue's description:
> Added first layer of the echo canceller 3 functionality.
>
> This CL adds the first layer of the echo canceller 3.
> All of the code is as it should, apart from
> block_processor.* which only contains placeholder
> functionality. (Upcoming CLs will add proper
> functionality into those files.)
>
>
>
> BUG=webrtc:6018
>
> Review-Url: https://codereview.webrtc.org/2584493002
> Cr-Commit-Position: refs/heads/master@{#15861}
> Committed: 38fd1758e9TBR=ivoc@webrtc.org,aleloi@webrtc.org,henrik.lundin@webrtc.org,peah@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2603293002
Cr-Commit-Position: refs/heads/master@{#15877}
This CL adds the first layer of the echo canceller 3.
All of the code is as it should, apart from
block_processor.* which only contains placeholder
functionality. (Upcoming CLs will add proper
functionality into those files.)
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2584493002
Cr-Commit-Position: refs/heads/master@{#15861}
This moves some GN check configurations out of .gn to individual
targets.
This commit also removes the source file 'mocks/mock_rtp_rtcp.h' from
the static_library 'rtp_rtcp' because it depends on a 'testonly = true'
target. After a check this seems only included in the unitest code:
$ grep -Rn "mocks/mock_rtp_rtcp.h" webrtc/modules/rtp_rtcp/
webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc:18:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc:17:#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
This commit also removes the dependency on
'//webrt/modules/video_coding' because it seems that the following
include can be removed:
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
The now checked target is:
"//webrtc/modules/rtp_rtcp/*"
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2598963002
Cr-Commit-Position: refs/heads/master@{#15760}
Theil and Sen's estimator essentially looks at the line through every pair of points and selects the median slope. This is robust to corruption of up to 29% of the data points.
Wire up new estimator to field trial experiment. Add unit and integration tests. Results are promising.
BUG=webrtc:6728
Review-Url: https://codereview.webrtc.org/2512693002
Cr-Commit-Position: refs/heads/master@{#15508}
This moves some GN check configurations out of .gn to individual targets.
The now checked targets are:
"//webrtc/api/*",
"//webrtc/audio/*",
"//webrtc/modules/audio_coding/*",
Many targets were fixed by adding dependencies, but the ones that
requires more refactorings are left with the check_includes attribute
set to false instead.
Make //webrtc/test:test_support a public dep of //webrtc/test:test_main
to avoid having to add that to all users of it.
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2556943003
Cr-Commit-Position: refs/heads/master@{#15461}
I decided to make one webrtc/sdk/android/BUILD.gn for both tests and Java/jni src.
External dependencies needs to be updated after this CL.
Future work is required to clean up the Android api and move
implementation details to /webrtc/sdk/android/src.
BUG=webrtc:5882,webrtc:6804
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2547483003
Cr-Commit-Position: refs/heads/master@{#15443}
Reason for revert:
Failures on the Linux Memcheck bot
Original issue's description:
> This approach passes packetization mode to the encoder as part of
> a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
>
> BUG=600254
>
> Committed: https://crrev.com/e59647b991f61cf1cf61b020356705e6c0f81257
> Cr-Commit-Position: refs/heads/master@{#15437}
TBR=hbos@webrtc.org,sprang@webrtc.org,mflodman@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=600254
Review-Url: https://codereview.webrtc.org/2558453002
Cr-Commit-Position: refs/heads/master@{#15441}
a cricket::VideoCodec structure, rather than as part of struct VideoCodecH264 inside webrtc::VideoCodec.
BUG=600254
Review-Url: https://codereview.webrtc.org/2528343002
Cr-Commit-Position: refs/heads/master@{#15437}
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
It is unused since cl https://codereview.webrtc.org/2386573002.
The new denoiser implementation and its tests are kept for now. This
code is also unused, but there are some plans to take this code into
use in the not too distant future.
BUG=None
Review-Url: https://codereview.webrtc.org/2496153002
Cr-Commit-Position: refs/heads/master@{#15338}
Pass the selected cricket::VideoCodec to H264EncoderImpl::H264EncoderImpl. The cricket::VideoCodec contains relevant information for H264 about selected profile and packetization mode.
BUG=chromium:600254,webrtc:6402, webrtc:6337
Review-Url: https://codereview.webrtc.org/2474993002
Cr-Commit-Position: refs/heads/master@{#15270}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
Reason for revert:
Downstream code has been updated.
Original issue's description:
> Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ )
>
> Reason for revert:
> Breaks downstream projects.
>
> Original issue's description:
> > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
> >
> > This CL removes RTPPayloadStrategy that is currently used to handle
> > audio/video specific aspects of payload handling. Instead, the audio and
> > video specific aspects will now have different functions, with linear
> > code flow.
> >
> > This CL does not contain any functional changes, and is just a
> > preparation for future CL:s.
> >
> > The main purpose with this CL is to add this function:
> > bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> > const webrtc::VideoCodec& video_codec);
> > that can easily be extended in a future CL to look at video codec
> > specific information.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> > Cr-Commit-Position: refs/heads/master@{#15232}
>
> TBR=danilchap@webrtc.org,solenberg@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:6743
>
> Committed: https://crrev.com/33c81d05613f45f65ee17224ed381c6cdd1c6c6f
> Cr-Commit-Position: refs/heads/master@{#15234}
TBR=danilchap@webrtc.org,solenberg@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:6743
Review-Url: https://codereview.webrtc.org/2531043002
Cr-Commit-Position: refs/heads/master@{#15245}
Reason for revert:
Breaks downstream projects.
Original issue's description:
> Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
>
> This CL removes RTPPayloadStrategy that is currently used to handle
> audio/video specific aspects of payload handling. Instead, the audio and
> video specific aspects will now have different functions, with linear
> code flow.
>
> This CL does not contain any functional changes, and is just a
> preparation for future CL:s.
>
> The main purpose with this CL is to add this function:
> bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> const webrtc::VideoCodec& video_codec);
> that can easily be extended in a future CL to look at video codec
> specific information.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> Cr-Commit-Position: refs/heads/master@{#15232}
TBR=danilchap@webrtc.org,solenberg@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:6743
Review-Url: https://codereview.webrtc.org/2528993002
Cr-Commit-Position: refs/heads/master@{#15234}
This CL removes RTPPayloadStrategy that is currently used to handle
audio/video specific aspects of payload handling. Instead, the audio and
video specific aspects will now have different functions, with linear
code flow.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
The main purpose with this CL is to add this function:
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const webrtc::VideoCodec& video_codec);
that can easily be extended in a future CL to look at video codec
specific information.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2524923002
Cr-Commit-Position: refs/heads/master@{#15232}
This change adds RotateDesktopFrame(), RotateRect(), RotateSize(),
ReverseRotate() functions, so an implementation can use these free functions to
rotate and copy pixels from one DesktopFrame to another at the same time.
This is the first part of the change to support rotation in DirectX capturer. In
a coming change, these functions will be used in DxgiOutputDuplicator to do the
rotation and copying.
Background,
DirectX APIs always return unrotated data buffer, so we need to rotate it to
match the user-selected rotation. What worse is except for the data buffer,
other variables return by these APIs are all rotated, e.g. output size, monitor
position. So we will eventually not be able to capture the rotated monitors,
because we cannot set their position and size correctly. Though
DXGI_OUTDUPL_DESC provides a DXGI_MODE_ROTATION enumeration to indicate the
output rotation, it does not provide a simple way to rotate an IDXGIResource,
which is the only thing we can get from duplication APIs. A typical user case
here is to use a matrix to transform the IDXGIResource and render it to a
surface. But since we do not render the IDXGIResource at all, we need to
manually rotate it.
BUG=314516
Review-Url: https://codereview.webrtc.org/2500883004
Cr-Commit-Position: refs/heads/master@{#15205}
the APM parameters to the high-pass filter.
The introduction will be done in three steps:
1) This CL which introduces the new scheme and
changes the code in webrtcvoiceengine.cc to use it.
2) Introduce the scheme into upstream code.
3) Remove the HighPassFilter interface in APM.
BUG=webrtc::6220, webrtc::6296, webrtc::6297, webrtc::6181, webrtc::5298
Review-Url: https://codereview.webrtc.org/2415403002
Cr-Commit-Position: refs/heads/master@{#15197}
Move the resources to //resources and upload them to Google Storage.
BUG=webrtc:6727
Review-Url: https://codereview.webrtc.org/2508943004
Cr-Commit-Position: refs/heads/master@{#15152}
Reason for revert:
Breaks downstream projects:
error: undefined reference to 'rtc::ExpFilter::kValueUndefined'
error: undefined reference to 'rtc::ExpFilter::Apply(float, float)'
error: undefined reference to 'rtc::ExpFilter::Reset(float)'
rror: undefined reference to 'rtc::ExpFilter::UpdateBase(float)'
Original issue's description:
> Move smoothing filter to common audio.
>
> This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/a82395bf7cd15b7396456df06fe952ede8db0c39
> Cr-Commit-Position: refs/heads/master@{#15146}
TBR=minyue@webrtc.org,solenberg@webrtc.org,perkj@webrtc.org,tommi@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2510373002
Cr-Commit-Position: refs/heads/master@{#15147}
This will make the smoothing filter a basic tool that is going to be used by both voice engine and ANA.
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2484153002
Cr-Commit-Position: refs/heads/master@{#15146}
Parse the estimation parameters from the field trial string.
BUG=webrtc:6690
Review-Url: https://codereview.webrtc.org/2489323002
Cr-Commit-Position: refs/heads/master@{#15126}
To be able to smooth the bandwidth estimation according to the probing interval.
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2380883003
Cr-Commit-Position: refs/heads/master@{#15123}
This is yet another reland of https://codereview.webrtc.org/2434073003/
including two fixes:
1. SimulcastRateAllocator did not handle the screenshare settings properly for numSimulcastStreams = 1. Additional test case was added for that.
2. In VideoSender, when rate allocation is updated after setting a new VideoCodec config, only update the state of the EncoderParameters, but don't actually run SetRateAllocation on the encoder itself. This caused some problems upstreams.
Please review only the changes after patch set 1.
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
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2510583002 .
Cr-Commit-Position: refs/heads/master@{#15105}
