231 Commits

Author SHA1 Message Date
nisse
9c00f646f0 Delete method cricket::VideoFrame::Copy.
Should be unused in Chrome since cl
https://codereview.chromium.org/2068703002/

TBR=tkchin@webrtc.org,magjed@webrtc.org
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2080253002
Cr-Commit-Position: refs/heads/master@{#13236}
2016-06-21 11:04:30 +00:00
tommi
2e82f3821f Reland of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #1 id:1 of https://codereview.webrtc.org/2084873002/ )
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}
2016-06-21 07:26:48 +00:00
sakal
a536bfe70d 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

Review-Url: https://codereview.webrtc.org/2084873002
Cr-Commit-Position: refs/heads/master@{#13229}
2016-06-21 07:08:58 +00:00
Tommi
884c336c34 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

Review URL: https://codereview.webrtc.org/2078873002 .

Cr-Commit-Position: refs/heads/master@{#13219}
2016-06-20 17:43:10 +00:00
tommi
8e8222d0d2 Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ )
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}
2016-06-16 22:44:11 +00:00
tommi
e03f878737 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

Review-Url: https://codereview.webrtc.org/2071473002
Cr-Commit-Position: refs/heads/master@{#13175}
2016-06-16 20:29:12 +00:00
Niels Möller
b00dc386d3 Delete GetExecutablePath and related unused code.
The function GetExecutablePath is a hack with poor portability. Delete
it and its caller GetTestFilePath. The latter was used in
videoframe_unittest.h, where it is replaced by
webrtc::test::ResourcePath.

Delete unused functions declared in base/testutils.h: ReadFile,
GetSiblingDirectory, GetGoogle3Directory, GetTalkDirectory,
CmpHelperFileEq, EXPECT_FILEEQ, ASSERT_FILEEQ.

Delete unused functions declared in media/base/testutils.h:
GetTestFilePath (see above), LoadPlanarYuvTestImage,
DumpPlanarYuvTestImage, ComputePSNR, ComputeSumSquareError.

The functions LoadPlanarYuvTestImage, DumpPlanarYuvTestImage were used
in yuvscaler_unittests.cc and planarfunctions_unittests.cc, under
webrtc/pc. However, these tests are never compiled or run, and appear
not to have been for the last few years, and are therefore deleted
rather than updated. It might make sense to check if libyuv have
comparable tests, and if not, resurrect them as part of libyuv
unittests.

BUG=
R=perkj@webrtc.org

Review URL: https://codereview.webrtc.org/2058043002 .

Cr-Commit-Position: refs/heads/master@{#13163}
2016-06-16 10:44:44 +00:00
tommi
17c3cddf9d Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #23 id:430001 of https://codereview.webrtc.org/2035173002/ )
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}
2016-06-14 23:04:48 +00:00
solenberg
05b9803c8e Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2059403002
Cr-Commit-Position: refs/heads/master@{#13135}
2016-06-14 15:59:54 +00:00
ossu
dedfd28a52 Support for two audio codec lists down into WebRtcVoiceEngine.
Added the plumbing necessary to get two different lists of codecs from
WebRtcVoiceEngine up to MediaSessionDescriptionFactory.

This should be the last step in this set of CLs. Once
https://codereview.webrtc.org/1991233004/ has landed, it's possible to
implement the ReceiveCodecs getter with the info from the
AudioDecoderFactory. The factory needs to be updated to actually
produce the correct list, as well.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2013053002
Cr-Commit-Position: refs/heads/master@{#13131}
2016-06-14 14:12:46 +00:00
tommi
1c7eef652b 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=

Review-Url: https://codereview.webrtc.org/2035173002
Cr-Commit-Position: refs/heads/master@{#13129}
2016-06-14 11:38:43 +00:00
nisse
7336225505 Delete left-over files.
References from Chrome's build files are gone with
https://codereview.chromium.org/2054763002/ and
https://codereview.chromium.org/2056243003/

BUG=

Review-Url: https://codereview.webrtc.org/2063703002
Cr-Commit-Position: refs/heads/master@{#13123}
2016-06-14 08:54:52 +00:00
ossu
29b1a8d7ec Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
CreatePeerConnectionFactory does not yet expose the ability to set the
factory from the outside.

Added notry due to android_dbg being broken.

NOTRY=True
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/1991233004
Cr-Commit-Position: refs/heads/master@{#13112}
2016-06-13 14:35:01 +00:00
Niels Möller
718a763d59 Refactor scaling.
Introduce a new method I420Buffer::CropAndScale, and a static
convenience helper I420Buffer::CenterCropAndScale. Use them for almost
all scaling needs.

Delete the Scaler class and the cricket::VideoFrame::Stretch* methods.

BUG=webrtc:5682
R=pbos@webrtc.org, perkj@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2020593002 .

Cr-Commit-Position: refs/heads/master@{#13110}
2016-06-13 11:06:14 +00:00
Taylor Brandstetter
5d97a9a05b Adding more detail to MessageQueue::Dispatch logging.
Every message will now be traced with the location from which it was
posted, including function name, file and line number.

This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).

This logging should help us identify messages that are taking
longer than expected to be dispatched.

R=pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2019423006 .

Cr-Commit-Position: refs/heads/master@{#13104}
2016-06-10 21:17:33 +00:00
nisse
bdce06e460 Delete unused YuvFrameGenerator class.
NOTRY=True # android_arm64_rel bot not cooperating
BUG=

Review-Url: https://codereview.webrtc.org/2044703007
Cr-Commit-Position: refs/heads/master@{#13100}
2016-06-10 11:43:56 +00:00
nisse
efec5902a5 Reland of New method I420Buffer::SetToBlack. (patchset #1 id:1 of https://codereview.webrtc.org/2049023002/ )
Reason for revert:
Plan to reland with InitToBlack kept, to be able to update Chrome to use the new I420Buffer::SetToBlack method.

Original issue's description:
> Revert of New static method I420Buffer::SetToBlack. (patchset #4 id:60001 of https://codereview.webrtc.org/2029273004/ )
>
> Reason for revert:
> Breaks chrome, in particular, the tests in
>
> media_stream_remote_video_source_unittest.cc
>
> use the InitToBlack method which is being deleted.
>
> Original issue's description:
> > New static method I420Buffer::SetToBlack.
> >
> > Replaces cricket::VideoFrame::SetToBlack and
> > cricket::WebRtcVideoFrame::InitToBlack, which are deleted.
> >
> > Refactors the black frame logic in VideoBroadcaster, and a few of the
> > tests.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/663f9e2ddc86e813f6db04ba2cf5ac1ed9e7ef67
> > Cr-Commit-Position: refs/heads/master@{#13063}
>
> TBR=perkj@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/271d74078894bb24f454eb31b77e4ce38097a2fa
> Cr-Commit-Position: refs/heads/master@{#13065}

TBR=perkj@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2049513005
Cr-Commit-Position: refs/heads/master@{#13083}
2016-06-09 07:31:46 +00:00
nisse
271d740788 Revert of New static method I420Buffer::SetToBlack. (patchset #4 id:60001 of https://codereview.webrtc.org/2029273004/ )
Reason for revert:
Breaks chrome, in particular, the tests in

media_stream_remote_video_source_unittest.cc

use the InitToBlack method which is being deleted.

Original issue's description:
> New static method I420Buffer::SetToBlack.
>
> Replaces cricket::VideoFrame::SetToBlack and
> cricket::WebRtcVideoFrame::InitToBlack, which are deleted.
>
> Refactors the black frame logic in VideoBroadcaster, and a few of the
> tests.
>
> BUG=webrtc:5682
>
> Committed: https://crrev.com/663f9e2ddc86e813f6db04ba2cf5ac1ed9e7ef67
> Cr-Commit-Position: refs/heads/master@{#13063}

TBR=perkj@webrtc.org,pbos@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2049023002
Cr-Commit-Position: refs/heads/master@{#13065}
2016-06-08 12:21:02 +00:00
nisse
663f9e2ddc New static method I420Buffer::SetToBlack.
Replaces cricket::VideoFrame::SetToBlack and
cricket::WebRtcVideoFrame::InitToBlack, which are deleted.

Refactors the black frame logic in VideoBroadcaster, and a few of the
tests.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/2029273004
Cr-Commit-Position: refs/heads/master@{#13063}
2016-06-08 11:26:27 +00:00
deadbeef
5a4a75ae48 Combining SetVideoSend and SetSource into one method.
This means there's only one thread hop to the worker thread.

At the video engine level, SetOptions and SetSource
are combined into one method (all within the same critical section)
which ensures that no frame will be encoded while SetVideoSend
is only partially finished.

BUG=webrtc:5691

Review-Url: https://codereview.webrtc.org/1838413002
Cr-Commit-Position: refs/heads/master@{#13022}
2016-06-02 23:23:47 +00:00
terelius
54f9171b3f Minor lint-fixes in MediaChannel and VideoEngine2.
Review-Url: https://codereview.webrtc.org/2020243005
Cr-Commit-Position: refs/heads/master@{#12996}
2016-06-01 18:18:59 +00:00
isheriff
a1c548b9b9 Add RtpHeaderExtension to avoid client breakage
This fixes a client breakage by adding back the RtpHeaderExtension temporarily
so that it can be fixed in the client before being removed in webrtc.

BUG=

CQ_INCLUDE_TRYBOTS=tryserver.chromium.linux:linux_chromium_rel_ng;tryserver.chromium.win:win_chromium_rel_ng

Review-Url: https://codereview.webrtc.org/2024153002
Cr-Commit-Position: refs/heads/master@{#12977}
2016-05-31 23:12:32 +00:00
isheriff
6f8d686d35 Remove use of RtpHeaderExtension and clean up
Currently there are two structs that are identical and track extension details:
webrtc::RtpExtension
cricket::RtpHeaderExtension

The use of the structs is mixed in the code to track the extensions being
supported. This results in duplicate definition of
the URI constants and there is code to convert between the two structs.

Clean up to use a single RtpHeader throughout the codebase. The actual location
of RtpHeader may change in future (perhaps to be located in api/). Additionally,
this CL renames some of the constants to clarify Uri and Id use.

BUG= webrtc:5895

Review-Url: https://codereview.webrtc.org/1984983002
Cr-Commit-Position: refs/heads/master@{#12924}
2016-05-26 18:25:04 +00:00
nisse
d591e3fcf3 Delete IsMutable and IsExclusive methods.
This affects the webrtc::VideoFrameBuffer and cricket::VideoFrame
classes.

To make this work, VideoFrameFactory is changed to use an
I420BufferPool rather than a plain VideoFrame to cache allocated
frames.

The I420BufferPool is reorganized to return an I420Buffer,
rather than a proxy object.

BUG=webrtc:5921, webrtc:5682

Review-Url: https://codereview.webrtc.org/2009193002
Cr-Commit-Position: refs/heads/master@{#12919}
2016-05-26 13:50:00 +00:00
nisse
47ac4620c8 Delete AndroidVideoCapturer::FrameFactory.
Splits VideoCapturer::OnFrameCaptured into helper methods,
which enables use of the VideoAdaptation logic without
using a frame factory.

Refactors AndroidVideoCapturer to make adaptation decision
earlier, so we can crop and rotate using
NV12ToI420Rotate.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1973873003
Cr-Commit-Position: refs/heads/master@{#12895}
2016-05-25 15:47:05 +00:00
nisse
c82d0902e1 Don't do a thread jump for incoming frames.
We're now supposed to accept incoming frames from any thread.

BUG=webrtc:5902

Review-Url: https://codereview.webrtc.org/1987663002
Cr-Commit-Position: refs/heads/master@{#12844}
2016-05-23 07:39:45 +00:00
nisse
04ebea3629 Delete obsolete cricket::VideoFrame methods.
GetWidth and GetHeight (renamed to width and height),

GetNativeHandle (replaced by video_frame_buffer()->native_handle).

TBR=tkchin@webrtc.org (trivial changes to objc RTCVideoFrame and VideoRendererAdapter)

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1990063005
Cr-Commit-Position: refs/heads/master@{#12822}
2016-05-20 08:48:53 +00:00
magjed
604abe09f1 VideoAdapter: Drop frames based on actual fps instead of expected fps
Pass timestamps to VideoAdapter instead of setting expected input frame rate, and use that to calculate when frames should be dropped.

BUG=webrtc:4938
TEST=Enable quality slider and HUD in debug settings. Request low fps with the quality slider and observe dropped frames.

Review-Url: https://codereview.webrtc.org/1982983003
Cr-Commit-Position: refs/heads/master@{#12811}
2016-05-19 13:05:49 +00:00
Alejandro Luebs
c9b0c26e0c Surface the IntelligibilityEnhancer on MediaConstraints
R=henrika@webrtc.org, peah@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1952123003 .

Cr-Commit-Position: refs/heads/master@{#12763}
2016-05-16 22:32:45 +00:00
Taylor Brandstetter
db0cd9e774 Adding getParameters/setParameters APIs to RtpReceiver.
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.

R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1917193008 .

Cr-Commit-Position: refs/heads/master@{#12761}
2016-05-16 18:40:38 +00:00
magjed
709f73c04e VideoAdapter: Add cropping based on OnOutputFormatRequest()
If OnOutputFormatRequest() is called, VideoAdapter will crop to the same
aspect ratio as the requested format. The output from
VideoAdapter.AdaptFrameResolution() now contains both how to crop the
input frame, and how to scale the cropped frame to the final adapted
resolution.

BUG=b/28622232

Review-Url: https://codereview.webrtc.org/1966273002
Cr-Commit-Position: refs/heads/master@{#12732}
2016-05-13 17:26:05 +00:00
ivoc
c1513ee1a3 Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API.
The caller can set a negative or zero file size to avoid using a limit.
BUG=

Review-Url: https://codereview.webrtc.org/1974453002
Cr-Commit-Position: refs/heads/master@{#12730}
2016-05-13 15:30:44 +00:00
Henrik Kjellander
3fe372dbee Fix all -Wnon-virtual-dtor warnings.
This is needed to get the GN build going for several parts
of the code tree.

BUG=webrtc:3307
NOTRY=True
R=henrika@webrtc.org, nisse@webrtc.org

Review URL: https://codereview.webrtc.org/1928653005 .

Cr-Commit-Position: refs/heads/master@{#12693}
2016-05-12 06:11:09 +00:00
Danil Chapovalov
33b01f2162 Adds network thread to rtc::BaseChannel
BaseChannel do calls to transport_channel on network_thread,
while keep calls to media_engine on worker_thread.
It still works when network_thread == worker_thread.

BUG=webrtc:5645
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1903393004 .

Cr-Commit-Position: refs/heads/master@{#12690}
2016-05-11 17:55:41 +00:00
minyuel
b031a2e862 Allow WebRTC to offer receiving capability for 120ms Opus packets.
TEST=Build Chromium for receiving + a special AppRTCDemo built with 120ms Opus sending capability. Call went well.

BUG=
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1957963002 .

Cr-Commit-Position: refs/heads/master@{#12673}
2016-05-10 13:35:30 +00:00
nisse
ba6371ec86 Delete unused video capture support for cropping, non-square pixels, and ARGB screencast scaling.
First two are unused, because the instance variables ratio_w_,
ratio_h_, and square_pixel_aspect_ratio_, are never modified after
initialization to 0 and false.

ARGB is believed to be unused, and the scaling logic
is probably not appropriate in any case.

Also delete corresponding helper functions in
videocommon.cc.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1934503002
Cr-Commit-Position: refs/heads/master@{#12659}
2016-05-09 07:47:59 +00:00
Honghai Zhang
82d7862fe7 Change default timestamp to 64 bits in all webrtc directories.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1835053002 .

Cr-Commit-Position: refs/heads/master@{#12646}
2016-05-06 18:29:27 +00:00
kwiberg
bfefb03ec1 Replace scoped_ptr with unique_ptr everywhere
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review-Url: https://codereview.webrtc.org/1937693002
Cr-Commit-Position: refs/heads/master@{#12581}
2016-05-01 21:53:55 +00:00
kwiberg
a4ac4786a8 Define rtc::BufferT, like rtc::Buffer but for any trivial type
And redefine rtc::Buffer as

  using Buffer = BufferT<uint8_t>;

(In the long run, I'd like to remove the type alias and rename the
template to just rtc::Buffer, but that requires all current users of
Buffer to start saying Buffer<uint8_t> instead, and since Buffer is
used in the API, we can't do that in one step.)

The immediate reason for the new template is that we'd like to use
BufferT<int16_t> in the AudioDecoder interface.

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/1929903002
Cr-Commit-Position: refs/heads/master@{#12564}
2016-04-29 15:00:28 +00:00
nisse
ef8b61e110 Enable -Winconsistent-missing-override flag.
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.

NOPRESUBMIT=True
BUG=webrtc:3970

Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
2016-04-29 13:09:23 +00:00
nisse
0565451820 Reland of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #1 id:1 of https://codereview.webrtc.org/1921493004/ )
Reason for revert:
Chrome has been updated, cl https://codereview.chromium.org/1919283005/

Original issue's description:
> Revert of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #5 id:80001 of https://codereview.webrtc.org/1901973002/ )
>
> Reason for revert:
> GetYPlane, GetYPitch etc is used by Chromium.
>
> Original issue's description:
> > Delete cricket::VideoFrame methods GetYPlane and GetYPitch.
> >
> > (And similarly for U and V). Also change video_frame_buffer method to
> > return a const ref to a scoped_ref_ptr.
> >
> > This cl is analogous to https://codereview.webrtc.org/1900673002/,
> > which delete corresponding methods in webrtc::VideoFrame.
> >
> > BUG=webrtc:5682
> >
> > Committed: https://crrev.com/1c27c6bf4cf0476dd2f09425509afaae4cdfe599
> > Cr-Commit-Position: refs/heads/master@{#12492}
>
> TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5682
>
> Committed: https://crrev.com/b05f994bb6f3055c852891c8acb531aee916a668
> Cr-Commit-Position: refs/heads/master@{#12494}

TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,terelius@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1923903002
Cr-Commit-Position: refs/heads/master@{#12559}
2016-04-29 09:56:06 +00:00
nisse
5b3c443d30 Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ )
Reason for revert:
Breaks chrome FYI bots.

Original issue's description:
> Delete webrtc::VideoFrame methods buffer and stride.
>
> To make the HasOneRef/IsMutable hack work, also had to change the
> video_frame_buffer method to return a const ref to a scoped_ref_ptr,
> to not imply an AddRef.
>
> BUG=webrtc:5682

TBR=perkj@webrtc.org,magjed@webrtc.org,pbos@webrtc.org,pthatcher@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:5682

Review-Url: https://codereview.webrtc.org/1935443002
Cr-Commit-Position: refs/heads/master@{#12558}
2016-04-29 09:39:33 +00:00
nisse
a0591b5473 Delete webrtc::VideoFrame methods buffer and stride.
To make the HasOneRef/IsMutable hack work, also had to change the
video_frame_buffer method to return a const ref to a scoped_ref_ptr,
to not imply an AddRef.

BUG=webrtc:5682

Review-Url: https://codereview.webrtc.org/1900673002
Cr-Commit-Position: refs/heads/master@{#12557}
2016-04-29 09:09:33 +00:00
kwiberg
4485ffb58d #include "webrtc/base/constructormagic.h" where appropriate
Any file that uses the RTC_DISALLOW_* macros should #include
"webrtc/base/constructormagic.h", but a shocking number of them don't.
This causes trouble when we try to wean files off of #including
scoped_ptr.h, since a bunch of files get their constructormagic macros
only from there.

Rather than fixing these errors one by one as they turn up, this CL
simply ensures that every file in the WebRTC tree that uses the
RTC_DISALLOW_* macros #includes "webrtc/base/constructormagic.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1917043005

Cr-Commit-Position: refs/heads/master@{#12509}
2016-04-26 15:14:48 +00:00
terelius
b05f994bb6 Revert of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #5 id:80001 of https://codereview.webrtc.org/1901973002/ )
Reason for revert:
GetYPlane, GetYPitch etc is used by Chromium.

Original issue's description:
> Delete cricket::VideoFrame methods GetYPlane and GetYPitch.
>
> (And similarly for U and V). Also change video_frame_buffer method to
> return a const ref to a scoped_ref_ptr.
>
> This cl is analogous to https://codereview.webrtc.org/1900673002/,
> which delete corresponding methods in webrtc::VideoFrame.
>
> BUG=webrtc:5682
>
> Committed: https://crrev.com/1c27c6bf4cf0476dd2f09425509afaae4cdfe599
> Cr-Commit-Position: refs/heads/master@{#12492}

TBR=magjed@webrtc.org,perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,nisse@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5682

Review URL: https://codereview.webrtc.org/1921493004

Cr-Commit-Position: refs/heads/master@{#12494}
2016-04-25 18:41:55 +00:00
nisse
1c27c6bf4c Delete cricket::VideoFrame methods GetYPlane and GetYPitch.
(And similarly for U and V). Also change video_frame_buffer method to
return a const ref to a scoped_ref_ptr.

This cl is analogous to https://codereview.webrtc.org/1900673002/,
which delete corresponding methods in webrtc::VideoFrame.

BUG=webrtc:5682

Review URL: https://codereview.webrtc.org/1901973002

Cr-Commit-Position: refs/heads/master@{#12492}
2016-04-25 16:45:38 +00:00
Taylor Brandstetter
0cd086b70e Adding codecs to the RtpParameters returned by an RtpSender.
Contains every field except for sdpFmtpLine.
Setting a reordered list of codecs is not yet supported.

R=glaznev@webrtc.org, pthatcher@webrtc.org, skvlad@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1885473004 .

Cr-Commit-Position: refs/heads/master@{#12453}
2016-04-20 23:23:22 +00:00
Peter Boström
d1f584bb06 Fix flake in TwoStreamsSendAndReceive.
Whether two streams get 300k or 150k as initial bitrate is flaky, since
InitEncode may happen asynchronously either before or after two streams
have shared the 300k, meaning that the first sender either thinks it
should start at 300k or at 150k.

This should ideally be fixed by reconfiguring encoders to use QVGA if a
lower estimate arrives before the first frame is encoded, but right now
that would require reconfigure logic in all VideoEncoder wrappers, which
is also less than ideal. It would be good to revisit this once
QualityScaler moves outside the VideoEncoder implementations (into
GenericEncoder).

BUG=webrtc:5678
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1902413002 .

Cr-Commit-Position: refs/heads/master@{#12448}
2016-04-20 14:32:01 +00:00
pbos
9b2119be47 Reland of Use initial bitrates for software VP8. (patchset #1 id:1 of https://codereview.webrtc.org/1898183002/ )
Reason for revert:
Chromium test updated to handle this change.

Original issue's description:
> Revert of Use initial bitrates for software VP8. (patchset #3 id:40001 of https://codereview.webrtc.org/1893313002/ )
>
> Reason for revert:
> Likely broke Chromium:
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Linux%20Tester/builds/26838
> https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win10%20Tester/builds/2224
>
> Original issue's description:
> > Use initial bitrates for software VP8.
> >
> > Makes the software encoder start at VGA as well, since ~300k isn't good
> > enough to produce a good HD stream.
> >
> > BUG=webrtc:5678
> > R=glaznev@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/e1da27e543bdb1983638118172a4efd599ca51b5
> > Cr-Commit-Position: refs/heads/master@{#12428}
>
> TBR=stefan@webrtc.org,glaznev@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5678
>
> Committed: https://crrev.com/5aa2d344d7e0b8940794d3c4422f81ac81249022
> Cr-Commit-Position: refs/heads/master@{#12430}

TBR=stefan@webrtc.org,glaznev@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:5678

Review URL: https://codereview.webrtc.org/1906513002

Cr-Commit-Position: refs/heads/master@{#12447}
2016-04-20 13:37:44 +00:00
Honghai Zhang
0e533ef487 Update the call when the network route changes
so that BWE can be updated promptly.

BUG=webrtc:5726
R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@google.com, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1844773002 .

Cr-Commit-Position: refs/heads/master@{#12432}
2016-04-19 22:41:53 +00:00