This enabled us to be able to remove VideoTrack::GetSink and RemoteVideoCapturer.
Since video frames from the decoder is delivered on a media engine internal thread, VideoBroadCaster must be made thread safe.
BUG=webrtc:5426
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1765423005 .
Cr-Commit-Position: refs/heads/master@{#11944}
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}
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}
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}
The purose of this cl is to remove dependency on cricket::VideoCapturer from WebRtcVideoChannel2.
This cl change CPU adaptation to use a new VideoSinkWants.Resolution
Cl is WIP and uploaded to start the discussion.
Tested on a N5 with hw acceleration turned off.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1695263002
Cr-Commit-Position: refs/heads/master@{#11804}
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}
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}
For audio, the flag is apparently unused. For video, the flag is moved to
VideoSendParameters, with the intention to keep only per-stream flags in
VideoOptions. The flag is used for the webrtcvideoengine2 logic commented like
// Conference mode screencast uses 2 temporal layers split at 100kbit.
// For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
// on the VideoCodec struct as target and max bitrates, respectively.
// See eg. webrtc::VP8EncoderImpl::SetRates().
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1697163002
Cr-Commit-Position: refs/heads/master@{#11651}
Reason for revert:
There appears there were some uses left of cricket::VideoFrame::GetRotation (to be replaced by GetVideoRotation). Investigating.
Original issue's description:
> Initial cleanup of cricket::VideoFrame.
>
> Deleted GetRotation (old alias for GetVideoRotation).
> Deleted CopyToBuffer.
> Deleted Sizeof.
> Deleted Write.
> Demote CopyToPlanes to protected status.
>
> BUG=webrtc:5426
TBR=perkj@webrtc.org,pbos@webrtc.org,pthatcher@webrtc.org,pthatcher@google.com
# 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/1688953003
Cr-Commit-Position: refs/heads/master@{#11566}
Further more, it adds a VideoBroadcaster than is used for delivering frames to multiple sinks.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1655793003
Cr-Commit-Position: refs/heads/master@{#11563}
More work remains, but is less urgent.
webrtc/media/base/mediacommon.h could not be deleted since
the constants are used in multiple places.
BUG=webrtc:5420
TBR=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1688753002 .
Cr-Commit-Position: refs/heads/master@{#11551}
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
And the list goes on, there's a dozen of different classes which act as video frame sinks.
At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
BUG=webrtc:5426
R=perkj@webrtc.org, pthatcher@webrtc.org
Committed: https://crrev.com/a862d4563fbc26e52bed442de784094ae1dfe5ee
Cr-Commit-Position: refs/heads/master@{#11396}
Review URL: https://codereview.webrtc.org/1594973006
Cr-Commit-Position: refs/heads/master@{#11414}
Reason for revert:
Broke chrome build. Investigating.
First error relating to AddSink method in mock_peer_connection_dependency_factory.h
Original issue's description:
> New rtc::VideoSinkInterface.
>
> The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
>
> And the list goes on, there's a dozen of different classes which act as video frame sinks.
>
> At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
>
> BUG=webrtc:5426
> R=perkj@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/a862d4563fbc26e52bed442de784094ae1dfe5ee
> Cr-Commit-Position: refs/heads/master@{#11396}
TBR=pthatcher@webrtc.org,pbos@webrtc.org,perkj@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/1646463002
Cr-Commit-Position: refs/heads/master@{#11397}
The plan is that this interface should be used by all classes which receive a stream of video frames, and replace the two generic classes webrtc::VideoRendererInterface and cricket::VideoRenderer.
And the list goes on, there's a dozen of different classes which act as video frame sinks.
At some point, we will likely add some method to handle sink properties like, e.g, maximum useful width and height. But hopefully this can be done while keeping the interface very simple.
BUG=webrtc:5426
R=perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1594973006 .
Cr-Commit-Position: refs/heads/master@{#11396}
