This involves changing a few verification functions for frames
received so that they always accept the result if there's no stream.
BUG=
Review URL: https://codereview.webrtc.org/1772353002
Cr-Commit-Position: refs/heads/master@{#11937}
Adds setConfiguration back and renames statsId back to reportId.
BUG=
Review URL: https://codereview.webrtc.org/1778033002
Cr-Commit-Position: refs/heads/master@{#11936}
It may be null if the network is unknown.
Also revised the logging to replace network id with network.toString(). They are pretty much the same for logging but network.toString does not need to parse the int value.
BUG=
Review URL: https://codereview.webrtc.org/1774343002
Cr-Commit-Position: refs/heads/master@{#11925}
At the top level, setting a track on an RtpSender is equivalent to
setting a source (previously called a renderer)
on a voice send stream. An RtpSender without a track
is not supposed to send data (not even muted data), so a send stream without
a source shouldn't send data.
Also replacing SendFlags with a boolean and implementing "Start"
and "Stop" methods on AudioSendStream, which was planned anyway
and simplifies this CL.
R=pthatcher@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1741933002 .
Cr-Commit-Position: refs/heads/master@{#11918}
switchCamera() only calls stopCaptureOnCameraThread(), not
stopCapture(), so the stopListening() call must be placed there.
BUG=webrtc:5519,b/27497950
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1770423002 .
Cr-Commit-Position: refs/heads/master@{#11917}
VideoTrackSource will be implemented in an upcoming cl but is needed to be included in libjingle.gyp in Chrome before the cl can be landed.
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1769343003 .
Cr-Commit-Position: refs/heads/master@{#11897}
Moved VideoSourceInterface to MediaStreamInterface.h
Renamed VideoSourceTest to VideoCapturerTrackSourceTest
Renamed VideoSource to VideoCaptureTrackSource and cl lint and cl format.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1770003002 .
Cr-Commit-Position: refs/heads/master@{#11893}
Changing from:
virtual void RequestIdentity(
rtc::KeyParams key_params,
rtc::Optional<uint64_t> expires,
const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
to:
virtual void RequestIdentity(
const rtc::KeyParams& key_params,
const rtc::Optional<uint64_t>& expires_ms,
const rtc::scoped_refptr<DtlsIdentityRequestObserver>& observer);
Making FakeDtlsIdentityStore DCHECK that |expires_ms| is not set, since it does not support that parameterization.
In a follow-up chromium CL the new signature will be used.
BUG=webrtc:5092, chromium:544902
Review URL: https://codereview.webrtc.org/1766673002
Cr-Commit-Position: refs/heads/master@{#11892}
Also change the type of "time interval" to int from uint32.
Fixed a few TODO therein. I think we should have the following convention:
1. All time delay/intervals should have type int although the time instant should have time uint32_t.
2. "interval" is preferred to "delay" if the delay will be repeated (like rescheduling).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1762863002 .
Cr-Commit-Position: refs/heads/master@{#11888}
Soft reset can be used when input frame resolution changes
to avoid re creating MediaCodec instance.
Instead MediaCodec is flushed and some variables are reset.
R=pbos@webrtc.org, perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1732533002 .
Cr-Commit-Position: refs/heads/master@{#11878}
This temporarily fixes a probem where renderers causes VideoSinkWants.rotation_applied=true.
The problem was introduced by https://codereview.webrtc.org/1759473003/ where VideTrackRenderes are registered to the cricket::VideoCapturer with default VideoSinkWants.
BUG=webrtc:5621
Review URL: https://codereview.webrtc.org/1764693004
Cr-Commit-Position: refs/heads/master@{#11871}
all interfaces that formerly took constraints parameters
in name=value form.
This is in preparation for making Chrome only use these
explicit interfaces.
BUG=webrtc:4906
Review URL: https://codereview.webrtc.org/1717583002
Cr-Commit-Position: refs/heads/master@{#11870}
This CL replaces the function SurfaceTextureHelper.setListener() that
could only be called once with the functions startListening() and
stopListening() that can be called multiple times. This is necessary
when the SurfaceTextureHelper will be passed to the VideoCapturerAndroid
in startCapture(). startListening() will be called in startCapture() and
stopListening() in stopCapture().
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1755573002
Cr-Commit-Position: refs/heads/master@{#11855}
This feature is off by default and can be turned on by setting IceConfig. When turned on, we'll choose a Turn/Turn (UDP takes higher priroity) over the other types of connections while no any connection is writable. However, when there is best connection or there is pending triggered check, those will take higher priority.
BUG=webrtc:4591
Review URL: https://codereview.webrtc.org/1577233006
Cr-Commit-Position: refs/heads/master@{#11850}
This is a preparation CL. The expires param will be used in
a follow-up CL. Initially it will only be used by the
chromium implementation. Then we will either update the
webrtc implementation (DtlsIdentityStoreImpl) to use it or
we will remove that store completely as part of clean-up
work.
There are currently two versions of RequestIdentity, one
that takes KeyType and one that takes KeyParams.
The KeyType version is removed in favor of the new
KeyParams + expires version. The KeyParams version without
expires is kept as to not break chromium which currently
implements that. This is the version that can be removed in
a follow-up CL.
BUG=webrtc:5092, chromium:544902
Review URL: https://codereview.webrtc.org/1749193002
Cr-Commit-Position: refs/heads/master@{#11846}
Multiple sources with the same names forces ugly GYP hacks in
Chromium's libjingle.gyp. Rename the sources in WebRTC to
enable cleaning this up in Chromium.
To summarize:
webrtc/media/base/constants.{cc,h} -> mediaconstants.{cc,h}
webrtc/p2p/base/constants.{cc,h} -> p2pconstants.{cc,h}
This CL will require coordinating landing a roll in Chromium.
BUG=webrtc:4256
NOTRY=True
Review URL: https://codereview.webrtc.org/1750593002
Cr-Commit-Position: refs/heads/master@{#11842}
This cl copies the value of cricket::VideoCapturer::IsScreencast into
a flag in VideoOptions. It is passed on via the chain
VideortpSender::SetVideoSend
WebRtcVideoChannel2::SetVideoSend
WebRtcVideoChannel2::SetOptions
WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions
Where it's used, in
WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame, we can look it up
in parameters_, instead of calling capturer_->IsScreencast().
Doesn't touch screencast logic related to cpu adaptation, since that
code is in flux in a different cl.
Also drop the is_screencast flag from the Dimensions struct, and drop separate options argument from ConfigureVideoEncoderSettings and SetCodecAndOptions, instead always using the options recorded in VideoSendStreamParameters::options.
In the tests, changed FakeVideoCapturer::is_screencast to be a construction time flag. Generally, unittests of screencast have to both use a capturer configured for screencast, and set the screencast flag using SetSendParameters. Since the automatic connection via VideoSource and VideoRtpSender isn't involved in the unit tests.
Note that using SetSendParameters to set the screencast flag doesn't make sense, since it's not per-stream. SetVideoSend would be more appropriate. That should be fixed if/when we drop VideoOptions from SetSendParameters.
BUG=webrtc:5426
R=pbos@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1711763003 .
Cr-Commit-Position: refs/heads/master@{#11837}
Rename SetCodecAndOptions to SetCodec, it no longer sets or uses the
VideoOptions. In MediaConfig, collect the video-related flags into a
struct.
As a followup, it should be possible to delete VideoOptions from
VideoSendParameters and VideoSendStreamParameters.
TBR=pthatcher@webrtc.org
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1745003002
Cr-Commit-Position: refs/heads/master@{#11828}
The current implementation is unnecessary expensive - we create a local reference frame for creating new Java objects and then create a new local reference. It's cheaper to just do jni->IsSameObject(obj, nullptr).
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1741723002 .
Cr-Commit-Position: refs/heads/master@{#11825}
I readded virtual bool Pause(bool paused) for now with a dummy implementation since Chrome remoting override this method.
Original cl description:
Removed unused cricket::VideoCapturer methods:
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
bool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()
This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.
There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
BUG=webrtc:5426
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1744153002 .
Cr-Commit-Position: refs/heads/master@{#11809}
This change is done to remove abnormally high decode time measurements for H264 decoding. H264 decoding sometimes keeps a few frames as reference before outputting a new decoded frame. This pipeline causes some frames to get stuck when the source stops sending new frames. When the source starts sending frames again, the decode time measurements for the frames that were stuck will include the pause time, which can be arbitrary high. This CL is a simple fix for this problem by constraining the decode time values to a "reasonable" range.
BUG=b/27306053
Review URL: https://codereview.webrtc.org/1725243007
Cr-Commit-Position: refs/heads/master@{#11792}
Currently, VideoCapturerAndroid owns a dedicated tread, and
SurfaceTextureHelper get this thread passed in the ctor. In
VideoCapturerAndroid.dispose(), ownership of the thread is passed to
SurfaceTextureHelper so that we can return directly instead of waiting
for the last frame to return.
This CL makes the SurfaceTextureHelper own the thread the whole time
instead, and VideoCapturerAndroid calls getHandler() to get it instead.
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1738123002
Cr-Commit-Position: refs/heads/master@{#11790}
Reason for revert:
Breaks remoting::protocol::WebrtcVideoCapturerAdapter::Pause'
See https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/3689/steps/compile/logs/stdio
Original issue's description:
> Removed unused cricket::VideoCapturer methods:
>
> void UpdateAspectRatio(int ratio_w, int ratio_h);
> void ClearAspectRatio();
> ool Pause(bool paused);
> Restart(const VideoFormat& capture_format);
> MuteToBlackThenPause(bool muted);
> IsMuted() const
> set_square_pixel_aspect_ratio
> bool square_pixel_aspect_ratio()
>
> This cl also remove the use of messages and posting of state change.
> Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
> It does not add restrictions on what thread frames are delivered on though.
>
> There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/e9c0cdff2dad2553b6ff6820c0c7429cb2854861
> Cr-Commit-Position: refs/heads/master@{#11773}
TBR=magjed@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1740963002
Cr-Commit-Position: refs/heads/master@{#11777}
This patch tries to only change the interface to VideoTrack, with
minimal changes to the implementation. Some points worth noting:
VideoTrackRenderers should ultimately be deleted, but it is kept for
now since we need an object implementing webrtc::VideoRenderer, and
that shouldn't be VideoTrack.
BUG=webrtc:5426
TBR=glaznev@webrtc.org // please look at examples
Review URL: https://codereview.webrtc.org/1684423002
Cr-Commit-Position: refs/heads/master@{#11775}
void UpdateAspectRatio(int ratio_w, int ratio_h);
void ClearAspectRatio();
ool Pause(bool paused);
Restart(const VideoFormat& capture_format);
MuteToBlackThenPause(bool muted);
IsMuted() const
set_square_pixel_aspect_ratio
bool square_pixel_aspect_ratio()
This cl also remove the use of messages and posting of state change.
Further more - a thread checker is added to make sure methods are called on only one thread. Construction can happen on a separate thred.
It does not add restrictions on what thread frames are delivered on though.
There is more features in VideoCapturer::Onframe related to screen share in ARGB that probably can be cleaned up in a follow up cl.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1733673002
Cr-Commit-Position: refs/heads/master@{#11773}
Reason for revert:
Breaks GN in chromium.
Original issue's description:
> Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.
>
> webrtc/audio/audio_sink.h is used by voice engine, but webrtc/audio is
> depending on voice engine, resulting in a cyclic dependency (which we
> don't detect since we have that check turned off, see webrtc:4243).
>
> BUG=webrtc:4243, webrtc:5589
> R=pbos@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org
> TBR=tommi@webrtc.org
>
> Committed: https://crrev.com/99b345c4e50c59a776c56949c17da3f50992f1a2
> Cr-Commit-Position: refs/heads/master@{#11766}
TBR=solenberg@webrtc.org,pbos@webrtc.org,perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4243, webrtc:5589
Review URL: https://codereview.webrtc.org/1739783002
Cr-Commit-Position: refs/heads/master@{#11769}
Reason for revert:
Revert breaks other uses, a fix will be rolled into Chromium instead.
Original issue's description:
> Revert of Remove ignored return code from modules. (patchset #3 id:40001 of https://codereview.webrtc.org/1703833002/ )
>
> Reason for revert:
> Breaks Chromium.
>
> Original issue's description:
> > Remove ignored return code from modules.
> >
> > ModuleProcessImpl doesn't act on return codes and having them around is
> > confusing (it's unclear what an error return code here would do even).
> >
> > BUG=
> > R=tommi@webrtc.org
> >
> > Committed: f14c47a58c
>
> TBR=tommi@webrtc.org,pbos@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=
>
> Committed: https://crrev.com/da33a8a2a22f6d19ba2a8cce963beafbdbaa8fd8
> Cr-Commit-Position: refs/heads/master@{#11761}
TBR=tommi@webrtc.org,torbjorng@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/1737013002
Cr-Commit-Position: refs/heads/master@{#11762}
Reason for revert:
Breaks Chromium.
Original issue's description:
> Remove ignored return code from modules.
>
> ModuleProcessImpl doesn't act on return codes and having them around is
> confusing (it's unclear what an error return code here would do even).
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: f14c47a58cTBR=tommi@webrtc.org,pbos@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/1736663004
Cr-Commit-Position: refs/heads/master@{#11761}
ModuleProcessImpl doesn't act on return codes and having them around is
confusing (it's unclear what an error return code here would do even).
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1703833002 .
Cr-Commit-Position: refs/heads/master@{#11747}
RFC 5245 allows an ICE restart to occur on only one media section.
However, before this CL, if an endpoint attempted to do this, we would
change our local ICE ufrag/pwd in every media section.
Also did some refactoring, turning the transport options from
mediasesion.h into a map.
Review URL: https://codereview.webrtc.org/1671173002
Cr-Commit-Position: refs/heads/master@{#11728}
