Reason for revert:
Regressed behavior is actually desirable (go down to 360p instead of producing super-bad 720p).
Original issue's description:
> Revert of Make QualityScaler more responsive to downgrades. (patchset #3 id:40001 of https://codereview.webrtc.org/1830593003/ )
>
> Reason for revert:
> Speculative revert: want to see if this causes the regression in https://crbug.com/602621
>
> Original issue's description:
> > Make QualityScaler more responsive to downgrades.
> >
> > Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
> > windows for going up quickly in the beginning of a call (before any
> > downscaling happens due to bad quality).
> >
> > BUG=webrtc:5678
> > R=glaznev@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/85829fd90cc4e7a91c9857921b19e8fc126aeb60
> > Cr-Commit-Position: refs/heads/master@{#12219}
>
> TBR=glaznev@webrtc.org,stefan@webrtc.org,pbos@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:5678
> NOTRY=true
>
> Committed: https://crrev.com/19b4fecf08e3fe215e431a260fb673553c15e569
> Cr-Commit-Position: refs/heads/master@{#12331}
TBR=glaznev@webrtc.org,stefan@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:602621, webrtc:5678
Review URL: https://codereview.webrtc.org/1887493003
Cr-Commit-Position: refs/heads/master@{#12341}
Reason for revert:
Speculative revert: want to see if this causes the regression in https://crbug.com/602621
Original issue's description:
> Make QualityScaler more responsive to downgrades.
>
> Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
> windows for going up quickly in the beginning of a call (before any
> downscaling happens due to bad quality).
>
> BUG=webrtc:5678
> R=glaznev@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/85829fd90cc4e7a91c9857921b19e8fc126aeb60
> Cr-Commit-Position: refs/heads/master@{#12219}
TBR=glaznev@webrtc.org,stefan@webrtc.org,pbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5678
NOTRY=true
Review URL: https://codereview.webrtc.org/1880103002
Cr-Commit-Position: refs/heads/master@{#12331}
This logic currently prevents loopback calls on Nexus 5X when it's
slightly overloaded to maintain input framerate since encoding at ~25fps
with one framedrop results in >70ms between frames naturally.
With this change applied Nexus 5X can maintain ~25fps both in and out
without building excessive latency (>2 frames) (this is now covered by
CPU adaptation outside the codec wrapper).
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1854413004 .
Cr-Commit-Position: refs/heads/master@{#12317}
Java objects in the API should be allowed to be null in some cases.
Specifically, a null value for maxBitrateBps in RtpParameters.java
has a specific meaning and doesn't imply an error has occurred.
NOTRY=True
Review URL: https://codereview.webrtc.org/1853523002
Cr-Commit-Position: refs/heads/master@{#12221}
Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
windows for going up quickly in the beginning of a call (before any
downscaling happens due to bad quality).
BUG=webrtc:5678
R=glaznev@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1830593003 .
Cr-Commit-Position: refs/heads/master@{#12219}
That won't work when rtc::scoped_ptr becomes a type alias for
std::unique_ptr.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1834103002
Cr-Commit-Position: refs/heads/master@{#12145}
The track state should be implicitly set by the underlying source.
This removes the public method and cleans up how AudioRtpReceiver is created. Further more it cleans up how the RtpReceivers are destroyed.
Note that this cl depend on https://codereview.webrtc.org/1790633002.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1816143002
Cr-Commit-Position: refs/heads/master@{#12115}
Reason for revert:
New attempt. Cl for removing videosourceinterface.h dep in chrome is landed here: https://codereview.chromium.org/1810273003/
Original issue's description:
> Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ )
>
> Reason for revert:
> Breaks Chromium build. Need to remove the references to the obsolete header files from Chromium and reland.
>
> Original issue's description:
> > Delete empty API files and cleaned up includes.
> >
> > TBR=glaznev@webrtc.org
> >
> > BUG=webrtc:5426
> >
> > Committed: https://crrev.com/c9022f508644dc33c01b05cb22ebfc2be145d6b2
> > Cr-Commit-Position: refs/heads/master@{#12039}
>
> TBR=nisse@webrtc.org,glaznev@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
>
> Committed: https://crrev.com/246b5273986d5a5b140b3d1a656baa8d40c36276
> Cr-Commit-Position: refs/heads/master@{#12042}
TBR=nisse@webrtc.org,glaznev@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:5426
Review URL: https://codereview.webrtc.org/1819733002
Cr-Commit-Position: refs/heads/master@{#12065}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1823503002
Cr-Commit-Position: refs/heads/master@{#12062}
Reason for revert:
I'm really sorry for having to revert this but it seems this hit an unexpected compile error downstream:
webrtc/media/sctp/sctpdataengine.cc: In function 'void cricket::VerboseLogPacket(const void*, size_t, int)':
webrtc/media/sctp/sctpdataengine.cc:172:37: error: invalid conversion from 'const void*' to 'void*' [-fpermissive]
data, length, direction)) != NULL) {
^
In file included from webrtc/media/sctp/sctpdataengine.cc:20:0:
third_party/usrsctp/usrsctplib/usrsctp.h:964:1: error: initializing argument 1 of 'char* usrsctp_dumppacket(void*, size_t, int)' [-fpermissive]
usrsctp_dumppacket(void *, size_t, int);
^
I'm sure you can fix this easily and just re-land this CL, while I'm going to look into how to add this warning at the public bots (on Monday).
Original issue's description:
> Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.
>
> This CL removes copy and assign support from Buffer and changes various
> parameters from Buffer to CopyOnWriteBuffer so they can be passed along
> and copied without actually copying the underlying data.
>
> With this changed some parameters to be "const" and fixed an issue when
> creating a CopyOnWriteBuffer with empty data.
>
> BUG=webrtc:5155
>
> Committed: https://crrev.com/944c39006f1c52aee20919676002dac7a42b1c05
> Cr-Commit-Position: refs/heads/master@{#12058}
TBR=kwiberg@webrtc.org,tkchin@webrtc.org,tommi@webrtc.org,pthatcher@webrtc.org,jbauch@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1817753003
Cr-Commit-Position: refs/heads/master@{#12060}
This CL removes copy and assign support from Buffer and changes various
parameters from Buffer to CopyOnWriteBuffer so they can be passed along
and copied without actually copying the underlying data.
With this changed some parameters to be "const" and fixed an issue when
creating a CopyOnWriteBuffer with empty data.
BUG=webrtc:5155
Review URL: https://codereview.webrtc.org/1785713005
Cr-Commit-Position: refs/heads/master@{#12058}
Reason for revert:
Breaks Chromium build. Need to remove the references to the obsolete header files from Chromium and reland.
Original issue's description:
> Delete empty API files and cleaned up includes.
>
> TBR=glaznev@webrtc.org
>
> BUG=webrtc:5426
>
> Committed: https://crrev.com/c9022f508644dc33c01b05cb22ebfc2be145d6b2
> Cr-Commit-Position: refs/heads/master@{#12039}
TBR=nisse@webrtc.org,glaznev@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/1813083002
Cr-Commit-Position: refs/heads/master@{#12042}
This CL changes the interface by adding a SurfaceTextureHelper argument
to VideoCapturer.startCapture(). This removes the need for the
VideoCapturer to create the SurfaceTextureHelper itself. This also means
that it is no longer necessary to send an EGLContext to the
VideoCapturerAndroid.create() function.
The SurfaceTextureHelper is now created in AndroidVideoCapturerJni, and
the EGLContext is passed from PeerConnectionFactory in
nativeCreateVideoSource().
Another change in this CL is that the C++ SurfaceTextureHelper creates
the Java SurfaceTextureHelper instead of getting it passed as an
argument in the ctor.
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1783793002
Cr-Commit-Position: refs/heads/master@{#11977}
and signaling the remote side to remove its remote candidate by setting the candidate priority to 0.
BUG=
Review URL: https://codereview.webrtc.org/1648813004
Cr-Commit-Position: refs/heads/master@{#11958}
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}
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}
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}
Boost low QP threashold to 21, otherwise VGA encoding never
scales up even at 2.5 Mbps.
Also reduce high QP threshold to scale down faster.
BUG=b/26504665
R=jackychen@google.com
Review URL: https://codereview.webrtc.org/1717763003 .
Cr-Commit-Position: refs/heads/master@{#11712}
This CL simplifies the VideoCapturer interface from 'String getSupportedFormatsAsJson() throws JSONException' to 'List<CaptureFormat> getSupportedFormats()'. The intermediate conversion to/from a JSON string is removed, and AndroidVideoCapturerJni converts the Java list to a C++ vector directly instead.
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1702603002 .
Cr-Commit-Position: refs/heads/master@{#11669}
For now, the network cost is purely based on the network type (cellular has cost 0xFFFF and everything else has cost 0).
Add cost to the candidate signaling and the stun request signaling (which is needed for peer reflexive candidates).
BUG=webrtc:4325
Review URL: https://codereview.webrtc.org/1668073002
Cr-Commit-Position: refs/heads/master@{#11642}
This CL factors out the interface that AndroidVideoCapturerJni is using to communicate with the Java counterpart. This interface is moved into VideoCapturer. The interface is not touched in this CL, and a follow-up CL is planned to simplify and improve it.
Another change is that the native part of VideoCapturer is created in PeerConnectionFactory.createVideoSource() instead of doing it immediately in the ctor.
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1696553003 .
Cr-Commit-Position: refs/heads/master@{#11606}
Reason for revert:
Breaks downstream compilation. Please reland in a non-breaking fashion.
Original issue's description:
> Android: Remove VideoCapturer
>
> This CL makes PeerConnectionFactory.createVideoSource() and nativeCreateVideoSource work directly with VideoCapturerAndroid instead of going via VideoCapturer. The native part is now created in nativeCreateVideoSource() instead of doing it immediately in VideoCapturerAndroid.create().
>
> BUG=webrtc:5519
> R=perkj@webrtc.org
>
> Committed: https://crrev.com/09eab315fddc3432c19d8f662f4b9360f2a58010
> Cr-Commit-Position: refs/heads/master@{#11582}
TBR=perkj@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5519
Review URL: https://codereview.webrtc.org/1690073002
Cr-Commit-Position: refs/heads/master@{#11586}
This CL makes PeerConnectionFactory.createVideoSource() and nativeCreateVideoSource work directly with VideoCapturerAndroid instead of going via VideoCapturer. The native part is now created in nativeCreateVideoSource() instead of doing it immediately in VideoCapturerAndroid.create().
BUG=webrtc:5519
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1684403002 .
Cr-Commit-Position: refs/heads/master@{#11582}
In addition to the code moved from talk/app/webrtc
there were some files in webrtc/api/objctests that still
had the libjingle license header.
BUG=webrtc:5418
TBR=tkchin@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1680293005
Cr-Commit-Position: refs/heads/master@{#11552}
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.
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
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc 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/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
