To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2378103005 .
Cr-Commit-Position: refs/heads/master@{#14452}
Reason for revert:
Caused issues with webrtc_perf_tests on build bots.
Original issue's description:
> Fix race / crash in OnNetworkRouteChanged().
>
> To achieve this some refactoring was done to make it possible to synchronize
> access to the DelayBasedBwe in TransportFeedbackAdapter:
> - The callback was removed from DelayBasedBwe, it now instead returns its
> result.
> - TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
> unnecessary dependencies.
>
> Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
>
> BUG=webrtc:6427, webrtc:6422
>
> Committed: https://crrev.com/fd0d42669204e6dd92a60736bca7ae0196663024
> Cr-Commit-Position: refs/heads/master@{#14430}
TBR=terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6427, webrtc:6422
Review-Url: https://codereview.webrtc.org/2377303002
Cr-Commit-Position: refs/heads/master@{#14433}
To achieve this some refactoring was done to make it possible to synchronize
access to the DelayBasedBwe in TransportFeedbackAdapter:
- The callback was removed from DelayBasedBwe, it now instead returns its
result.
- TransportFeedbackAdapter was moved to modules/congestion_controller to avoid
unnecessary dependencies.
Reenables previously disabled flaky test. Can no longer reproduce flakiness with gtest-parallel and asan/tsan builds.
BUG=webrtc:6427, webrtc:6422
Review-Url: https://codereview.webrtc.org/2366333003
Cr-Commit-Position: refs/heads/master@{#14430}
Reason for revert:
Fix backward compatibility support
Original issue's description:
> Revert of Unify rtcp packet setters (patchset #8 id:130001 of https://codereview.webrtc.org/2348623003/ )
>
> Reason for revert:
> Breaks compilation of internal downstream project.
>
> Original issue's description:
> > Unify rtcp packet setters
> > Renamed setters in rtcp classes
> > from WithField to SetField
> > from WithItem to AddItem or SetItems
> > from From to SetSenderSsrc
> > from To to SetMediaSsrc
> > Some redundant or unsued setters removed.
> > Pass-by-const& replaced with pass-by-value when appropriate.
> >
> > BUG=webrtc:5260
> >
> > Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> > Cr-Commit-Position: refs/heads/master@{#14393}
>
> TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5260
>
> Committed: https://crrev.com/efc6e41866662e0922858fbce1d9ee3bdd0637ed
> Cr-Commit-Position: refs/heads/master@{#14400}
TBR=sprang@webrtc.org,stefan@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2370313002
Cr-Commit-Position: refs/heads/master@{#14402}
Reason for revert:
Breaks compilation of internal downstream project.
Original issue's description:
> Unify rtcp packet setters
> Renamed setters in rtcp classes
> from WithField to SetField
> from WithItem to AddItem or SetItems
> from From to SetSenderSsrc
> from To to SetMediaSsrc
> Some redundant or unsued setters removed.
> Pass-by-const& replaced with pass-by-value when appropriate.
>
> BUG=webrtc:5260
>
> Committed: https://crrev.com/20e77c7b8a9f19942ef3c3c4f1fa3888b2cd54ea
> Cr-Commit-Position: refs/heads/master@{#14393}
TBR=sprang@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2372713005
Cr-Commit-Position: refs/heads/master@{#14400}
Renamed setters in rtcp classes
from WithField to SetField
from WithItem to AddItem or SetItems
from From to SetSenderSsrc
from To to SetMediaSsrc
Some redundant or unsued setters removed.
Pass-by-const& replaced with pass-by-value when appropriate.
BUG=webrtc:5260
Review-Url: https://codereview.webrtc.org/2348623003
Cr-Commit-Position: refs/heads/master@{#14393}
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}
This includes if RTCP is received, but the number of packets received by the
other end hasn't increased.
Further, if no RTCP is received for more than 3 feedback intervals (3 seconds)
we start reducing the estimate by 20%. This is put under an experiment.
BUG=webrtc:6238
R=terelius@webrtc.org
Review URL: https://codereview.webrtc.org/2262213002 .
Cr-Commit-Position: refs/heads/master@{#14306}
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}
This patch enables bwe related variable logging to the command line.
This is useful to test congestion control algorithm over real networks.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2296253002
Cr-Commit-Position: refs/heads/master@{#14209}
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}
Also lowering the min bitrate for simulations to be able to better capture this issue in the BweFeedbackTest.Choke200kbps30kbps200kbps performance test.
BUG=webrtc:6105
NOTRY=true
NOPRESUBMIT=true
Review-Url: https://codereview.webrtc.org/2201093006
Cr-Commit-Position: refs/heads/master@{#13639}
The added unittest triggers this CHECK:
433ed06800/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc (146)
It happens because the unwrap of the sequence number fails if the unwrappers last sequence number is small, but the newly added sequence number is large (greater than last seq num + 2^15), and therefore should have been interpreted as a reordering and a backwards wrap. Since that would mean the sequence number returned from the unwrapper would be negative, it simply returns the original sequence number instead. This causes problems later where the wrap is correctly handled, and everything breaks.
The real solution should be to correctly handle wraps, but to prevent the crash this is a reasonable workaround for now.
BUG=
Review-Url: https://codereview.webrtc.org/2157843002
Cr-Commit-Position: refs/heads/master@{#13496}
Also adds a copy of the BWE test suite to the new DelayBasedBwe class.
BUG=webrtc:6079
Review-Url: https://codereview.webrtc.org/2126793002
Cr-Commit-Position: refs/heads/master@{#13428}
A bug in the transpot feedback adapter causes new feedback message to
always start with a received packet. This makes it impossible for the
receiver to distinguish from actual dropped packets and dropped feedback
messages.
BUG=webrtc:6073
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2122863002 .
Cr-Commit-Position: refs/heads/master@{#13381}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
Reason for revert:
Reverting the revert. This change is not related to the failure on the Windows FYI bots. The cause of the failure has been reverted in Chromium:
https://codereview.chromium.org/2081653004/
Original issue's description:
> Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #5 id:340001 of https://codereview.webrtc.org/2078873002/ )
>
> Reason for revert:
> Breaks chromium.webrtc.fyi
>
> https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4719
> https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3120
>
> Original issue's description:
> > Reland of IncomingVideoStream refactoring.
> > This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
> >
> > Original issue's description (with non-smoothing references removed):
> >
> > Split IncomingVideoStream into two implementations, with smoothing and without.
> >
> > * Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
> >
> > * Removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
> >
> > * Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
> >
> > * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
> >
> > * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
> >
> > * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
> >
> > * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
> >
> > * Made the render delay value in VideoRenderFrames, const.
> >
> > BUG=chromium:620232
> > R=mflodman@webrtc.org, nisse@webrtc.org
> >
> > Committed: https://crrev.com/884c336c345d988974c2a69cea402b0fb8b07a63
> > Cr-Commit-Position: refs/heads/master@{#13219}
>
> TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,tommi@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=chromium:620232
>
> Committed: https://crrev.com/a536bfe70de38fe877245317a7f0b00bcf69cbd0
> Cr-Commit-Position: refs/heads/master@{#13229}
TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232
Review-Url: https://codereview.webrtc.org/2089613002
Cr-Commit-Position: refs/heads/master@{#13230}
Reason for revert:
Breaks chromium.webrtc.fyi
https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win7%20Tester/builds/4719https://uberchromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win10%20Tester/builds/3120
Original issue's description:
> Reland of IncomingVideoStream refactoring.
> This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
>
> Original issue's description (with non-smoothing references removed):
>
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> * Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
>
> * Removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=chromium:620232
> R=mflodman@webrtc.org, nisse@webrtc.org
>
> Committed: https://crrev.com/884c336c345d988974c2a69cea402b0fb8b07a63
> Cr-Commit-Position: refs/heads/master@{#13219}
TBR=nisse@webrtc.org,philipel@webrtc.org,mflodman@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232
Review-Url: https://codereview.webrtc.org/2084873002
Cr-Commit-Position: refs/heads/master@{#13229}
This reland does not contain the non-smoothing part of the original implementation. Instead, when smoothing is turned off, frame callbacks run on the decoder thread, as they did before. This code path is used in Chrome. As far as Chrome goes, the difference now is that there won't be an instance of IncomingVideoStream in between the decoder and the callback (i.e. fewer locks). Other than that, no change for Chrome.
Original issue's description (with non-smoothing references removed):
Split IncomingVideoStream into two implementations, with smoothing and without.
* Added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 6 locks.
* Removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
* Changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
* Made the render delay value in VideoRenderFrames, const.
BUG=chromium:620232
R=mflodman@webrtc.org, nisse@webrtc.org
Review URL: https://codereview.webrtc.org/2078873002 .
Cr-Commit-Position: refs/heads/master@{#13219}
Allows detecting large-enough audio packets as part of a probe,
speculative fix for a rampup-time regression in M50. These packets are
accounted on the send side when probing.
BUG=webrtc:5985
R=mflodman@webrtc.org, philipel@webrtc.org
Review URL: https://codereview.webrtc.org/2061193002 .
Cr-Commit-Position: refs/heads/master@{#13210}
Reason for revert:
Reverting again. The perf regression does not seem to be related to dropping frames.
Original issue's description:
> Reland of Split IncomingVideoStream into two implementations, with smoothing and without.
>
> Original issue's description:
>
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
>
> Further work done:
>
> * I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
>
> * I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=chromium:620232
> TBR=mflodman
>
> Committed: https://crrev.com/e03f8787377bbc03a4e00184bb14b7561b108cbb
> Cr-Commit-Position: refs/heads/master@{#13175}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:620232
Review-Url: https://codereview.webrtc.org/2071093002
Cr-Commit-Position: refs/heads/master@{#13176}
Original issue's description:
Split IncomingVideoStream into two implementations, with smoothing and without.
This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
Further work done:
* I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
* I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
* I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
* The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
* Made the render delay value in VideoRenderFrames, const.
BUG=chromium:620232
TBR=mflodman
Review-Url: https://codereview.webrtc.org/2071473002
Cr-Commit-Position: refs/heads/master@{#13175}
Reason for revert:
Reverting while we track down the issue on the Win10 bot.
Original issue's description:
> Split IncomingVideoStream into two implementations, with smoothing and without.
>
> This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
>
> Further work done:
>
> * I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
>
> * I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
>
> * I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
>
> * The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
>
> * The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
>
> * Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
>
> * Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
>
> * Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
>
> * Made the render delay value in VideoRenderFrames, const.
>
> BUG=
>
> Committed: https://crrev.com/1c7eef652b0aa22d8ebb0bfe2b547094a794be22
> Cr-Commit-Position: refs/heads/master@{#13129}
TBR=mflodman@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/2061363002
Cr-Commit-Position: refs/heads/master@{#13146}
This CL fixes an issue with the non-smoothing implementation where frames were delivered on the decoder thread. No-smoothing is now done in a separate class that uses a TaskQueue. The implementation may drop frames if the renderer doesn't keep up and it doesn't block the decoder thread.
Further work done:
* I added TODOs and documentation for VideoReceiveStream::OnFrame, where we today grab 5 locks.
* I removed the Start/Stop methods from the IncomingVideoStream implementations. Now, when an instance is created, it should be considered to be "running" and when it is deleted, it's "not running". This saves on resources and also reduces the amount of locking required and I could remove one critical section altogether.
* I changed the VideoStreamDecoder class to not depend on IncomingVideoStream but rather use the generic rtc::VideoSinkInterface<VideoFrame> interface. This means that any implementation of that interface can be used and the decoder can be made to just use the 'renderer' from the config. Once we do that, we can decouple the IncomingVideoStream implementations from the decoder and VideoReceiveStream implementations and leave it up to the application for how to do smoothing. The app can choose to use the Incoming* classes or roll its own (which may be preferable since applications often have their own scheduling mechanisms).
* The non-smoothing IncomingVideoStream implementation currently allows only 1 outstanding pending frame. If we exceed that, the current frame won't be delivered to the renderer and instead we deliver the next one (since when this happens, the renderer is falling behind).
* The lifetime of the VideoStreamDecoder instance is now bound to Start/Stop in VideoReceiveStream and not all of the lifetime of VideoReceiveStream.
* Fixed VideoStreamDecoder to unregister callbacks in the dtor that were registered in the ctor. (this was open to a use-after-free regression)
* Delay and callback pointers are now passed via the ctors to the IncomingVideoStream classes. The thread primitives in the IncomingVideoStream classes are also constructed/destructed at the same time as the owning object, which allowed me to remove one more lock.
* Removed code in the VideoStreamDecoder that could overwrite the VideoReceiveStream render delay with a fixed value of 10ms on construction. This wasn't a problem with the previous implementation (it would be now though) but seemed to me like the wrong place to be setting that value.
* Made the render delay value in VideoRenderFrames, const.
BUG=
Review-Url: https://codereview.webrtc.org/2035173002
Cr-Commit-Position: refs/heads/master@{#13129}
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}
To avoid the case where a single data point or too short window is used,
causing bad behavior due to bad stats, update RateStatistics to return
an Optional rather than a plain rate.
There was also a strange off by one bug where the rate was slightly
overestimated (N + 1 buckets, N ms time window).
These changes requires updates to a number of places, and may very well
cause seeming perf regressions (but the stats were probablty more wrong
previously).
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2029593002 .
Cr-Commit-Position: refs/heads/master@{#13103}
Sync the GYP and GN targets and update the name of the GN one
to 'remote_bitrate_estimator'.
Move the GYP variable 'enable_bwe_test_logging' into the local scope.
Remove redundant entries in modules.gyp.
These are preparations related to the GN migration.
BUG=webrtc:5949
TESTED=Ran GYP with the default variables and with
-Denable_bwe_test_logging=1. Compiled remote_bitrate_estimator
and verified that bwe_test_logging.cc is compiled only when
set.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2040313004
Cr-Commit-Position: refs/heads/master@{#13087}
