This cl clean up unused forward declaration of cricket::AudioFrame and cricket::Renderer.
cricket::VideoFrame is used in the interface and thus the header file is included.
BUG=webrtc:5687
Review-Url: https://codereview.webrtc.org/2145613006
Cr-Commit-Position: refs/heads/master@{#13477}
Merge with the latest webrtc native code.
Remove deprecated function Connect() in QuicTransportChannel.
Fix the compiling issue and broken unit tests by adding the network thread to QUIC related classes.
Review-Url: https://codereview.webrtc.org/2089553002
Cr-Commit-Position: refs/heads/master@{#13472}
Reason for revert:
Reverting because description was inaccurate. Will reland after updating description.
Original issue's description:
> Modify PeerConnection for end-to-end QuicDataChannel usage
>
> To allow end-to-end QuicDataChannel usage with a
> PeerConnection, RTCConfiguration has been modified to
> include a boolean for whether to do QUIC, since negotiation of
> QUIC is not implemented. If one peer does QUIC, then it will be
> assumed that the other peer must do QUIC or the connection
> will fail.
>
> PeerConnection has been modified to create data channels of type
> QuicDataChannel when the peer wants to do QUIC.
>
> WebRtcSession has been modified to use a QuicTransportChannel
> instead of a DtlsTransportChannelWrapper/DataChannel
> when QUIC should be used.
>
> Modification of previous in-flight CL: https://codereview.chromium.org/1844803002/
>
> Committed: https://crrev.com/36c8d69ce188102ae6fd48c371cf1518f08698fb
> Cr-Commit-Position: refs/heads/master@{#13470}
TBR=pthatcher@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2146133002
Cr-Commit-Position: refs/heads/master@{#13471}
To allow end-to-end QuicDataChannel usage with a
PeerConnection, RTCConfiguration has been modified to
include a boolean for whether to do QUIC, since negotiation of
QUIC is not implemented. If one peer does QUIC, then it will be
assumed that the other peer must do QUIC or the connection
will fail.
PeerConnection has been modified to create data channels of type
QuicDataChannel when the peer wants to do QUIC.
WebRtcSession has been modified to use a QuicTransportChannel
instead of a DtlsTransportChannelWrapper/DataChannel
when QUIC should be used.
Modification of previous in-flight CL: https://codereview.chromium.org/1844803002/
Review-Url: https://codereview.webrtc.org/2089553002
Cr-Commit-Position: refs/heads/master@{#13470}
SurfaceViewRendererOnMeasureTest#testFrame1280x720 is currently flaky
because of a race. This issue was introduced here:
https://codereview.webrtc.org/2111933002/. That CL moved the function
updateFrameDimensionsAndReportEvents() from renderFrame() to
renderFrameOnRenderThread(). The OnMeasureTest currently just calls
renderFrame() with a new frame size and immediately starts testing the
measured size, which might be before renderFrameOnRenderThread() is
executed. This CL waits for the RendererEvents.onFrameResolutionChanged()
callback before starting the test to fix this race.
BUG=webrtc:6089
Review-Url: https://codereview.webrtc.org/2147463002
Cr-Commit-Position: refs/heads/master@{#13448}
This interface and its implementations have been replaced by
rtc::RTCCertificateGeneratorInterface.
Removes dtlsidentitystore.h, updates .gyp/gn and removes old #includes.
BUG=webrtc:5707, webrtc:5708
Review-Url: https://codereview.webrtc.org/2034013003
Cr-Commit-Position: refs/heads/master@{#13432}
This CL moves some arguments, e.g. the camera thread, from the startCapture() function to a new initialize() function. These arguments are constant during the lifetime of the VideoCapturer, and are not changed for different startCapture() calls. Setting them once allows for simplifications in the code.
This CL also fixes a bug for camera2 where pendingCameraSwitchSemaphore might not be released when switchEventsHandler is null.
In camera1, the handler lock and 'cameraThreadHandler == null' check is replaced with an atomic boolean to check if the camera is stopped.
BUG=webrtc:5519
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2122693002 .
Cr-Commit-Position: refs/heads/master@{#13404}
Add null checks for eglBase, and detach the egl context in
SurfaceViewRenderer.surfaceDestroyed().
BUG=webrtc:6032
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2124983002 .
Cr-Commit-Position: refs/heads/master@{#13396}
we will periodically check if any network does not have any connection on it and if yes, attempt to re-gather on those networks.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2025573002 .
Cr-Commit-Position: refs/heads/master@{#13367}
Deadlock caused by two methods grabbing two locks in the opposite order:
renderFrame():
handlerLock
layoutLock
onMeasure():
layoutLock
handlerLock
This CL removs the nested locking to fix the deadlock and make it less
error prone for the future.
BUG=webrtc:6003
R=sakal@webrtc.org
Review URL: https://codereview.webrtc.org/2111933002 .
Cr-Commit-Position: refs/heads/master@{#13364}
Reason for revert:
Issues fixed
Original issue's description:
> Revert of Combine webrtc/api/java/android and webrtc/api/java/src. (patchset #1 id:1 of https://codereview.webrtc.org/2111823002/ )
>
> Reason for revert:
> Breaks downstream dependencies
>
> Original issue's description:
> > Combine webrtc/api/java/android and webrtc/api/java/src.
> >
> > It used to be that there was a Java api for devices not running Android
> > but that is no longer the case. I combined the directories and made
> > the folder structure chromium style.
> >
> > BUG=webrtc:6067
> > R=magjed@webrtc.org, tommi@webrtc.org
> >
> > Committed: ceefe20dd6
>
> TBR=magjed@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:6067
>
> Committed: 9b0dc622d4TBR=magjed@webrtc.org,tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6067
Review-Url: https://codereview.webrtc.org/2111923003
Cr-Commit-Position: refs/heads/master@{#13363}
Reason for revert:
Breaks downstream dependencies
Original issue's description:
> Combine webrtc/api/java/android and webrtc/api/java/src.
>
> It used to be that there was a Java api for devices not running Android
> but that is no longer the case. I combined the directories and made
> the folder structure chromium style.
>
> BUG=webrtc:6067
> R=magjed@webrtc.org, tommi@webrtc.org
>
> Committed: ceefe20dd6TBR=magjed@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:6067
Review URL: https://codereview.webrtc.org/2106333005 .
Cr-Commit-Position: refs/heads/master@{#13357}
It used to be that there was a Java api for devices not running Android
but that is no longer the case. I combined the directories and made
the folder structure chromium style.
BUG=webrtc:6067
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2111823002 .
Cr-Commit-Position: refs/heads/master@{#13356}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
Cr-Original-Commit-Position: refs/heads/master@{#13335}
Cr-Commit-Position: refs/heads/master@{#13354}
and even if the "unknown" network type is not helpful for identifying the network type, it helps bind sockets to the network.
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/2112963002 .
Cr-Commit-Position: refs/heads/master@{#13351}
Reason for revert:
Breaks Win32/Win64 Debug bots in client.webrtc waterfall
Original issue's description:
> Add config to prune low-priority TURN ports for creating connections
> When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
>
> This effectively reduces the number of TURN candidates and connections created by TURN ports.
>
> BUG=
> R=deadbeef@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/17aac053f585e892114974d2eb248e05ad37f973
> Cr-Commit-Position: refs/heads/master@{#13335}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org,honghaiz@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/2111663003
Cr-Commit-Position: refs/heads/master@{#13342}
The member variable |current_state_| in AndroidVideoCapturer is
unnecessary. All state changes are reported to the base class
cricket::VideoCapturer that already holds the capture state.
R=sakal@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2104813003 .
Cr-Commit-Position: refs/heads/master@{#13341}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}
TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749
Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.
Original issue's description:
> Fix to make the start/stop functions for the Rtc Eventlog non-virtual.
>
> This is needed to prevent the Chromium import bot from breaking.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/df6ecea8ac7c4c3bddeda089d5fb9eccdf38a0a6
> Cr-Commit-Position: refs/heads/master@{#13324}
TBR=tommi@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/2111803002
Cr-Commit-Position: refs/heads/master@{#13339}
functionality and exposes the functionality using the
MediaConstraints.
The exposing of the feature through the MediaConstraints
was done similarly to what was done for the intelligibility
enhancer in the CL
https://codereview.webrtc.org/1952123003
This CL is dependent on the CL https://codereview.webrtc.org/2090583002/ which contains
the level control functionality.
NOTRY=true
BUG=webrtc:5920
Review-Url: https://codereview.webrtc.org/2095563002
Cr-Commit-Position: refs/heads/master@{#13336}
When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
This effectively reduces the number of TURN candidates and connections created by TURN ports.
BUG=
R=deadbeef@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2093623004 .
Cr-Commit-Position: refs/heads/master@{#13335}
The parameters for Logging.enableTracing() were creating the impression
that they control level and severity of one tracing system and they are
meant to be used together. In fact the "levels" parameter controlled one
tracing system (WEBRTC_TRACE), and the "severity" parameter was
responsible for a completely different one: setting the severity level
above which log messages from LOG() will be directed to the
platform-specific debug output (logcat on Android).
The method signature suggested that the "path" parameter applied to both
systems - while it was only meaningful for the WEBRTC_TRACE; LOG
messages were directed to ADB logcat no matter what the Path value was.
It is possible to redirect LOG messages to a file, but that is done
using a completely different set of APIs
- PeerConnectionFactory.startInternalTracingCapture().
I've separated these two methods to make it more clear which of the
parameters controls which system.
NOTRY=true
Review-Url: https://codereview.webrtc.org/2110853003
Cr-Commit-Position: refs/heads/master@{#13334}
Test worked by creating a dummy data channel just to trigger the
SDP generation, then creating two data channels after negotiation.
However the dummy data channel is then racing with the "real" data
channel to get negotiated, so they could be signaled in the reverse
of the expected order.
Fixed this by simply waiting for the dummy data channel to be
signaled before creating the other data channels.
BUG=webrtc:3980
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2112593002 .
Cr-Commit-Position: refs/heads/master@{#13329}
In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
BUG=webrtc:4741,webrtc:5603,chromium:609749
R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1748403002 .
Cr-Commit-Position: refs/heads/master@{#13321}
Also move getDeviceNames to a more appropriate location in the file.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2105813002
Cr-Commit-Position: refs/heads/master@{#13312}
Relanding again after fixing issue with RTC_DCHECKs.
This CL eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13305}
The plan is that the CameraEnumerationAndroid will in the future have
method called getEnumerator that will return an enumerator that can be
used to create CameraVideoCapturer objects. It will return
Camera2Enumerator if it is supported or else Camera1Enumerator. Some
apps want to capture to byte buffers which is no longer supported in the
camera2 version of CameraVideoCapturer. Camera1Enumerator constructed
with false parameter as captureToTexture will be returned to these apps.
BUG=webrtc:5519
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/2071213003 .
Cr-Commit-Position: refs/heads/master@{#13294}
Reason for revert:
Broke video sending for iOS AppRTCDemo. To repro, run iOS AppRTCDemo in Release in loopback mode. The revision prior to this change worked.
Original issue's description:
> Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: 2d5491783aTBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2092273003
Cr-Commit-Position: refs/heads/master@{#13289}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13287}
Reason for revert:
Broke peerconnection_unittest somehow, due to introduction of thread check. Will fix and reland.
Original issue's description:
> Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
>
> This eliminates the need for the extra layer of indirection provided by
> mediastreamprovider.h. It will thus make it easier to implement new
> functionality in RtpSender/RtpReceiver.
>
> It also brings us one step closer to the end goal of combining "senders"
> and "send streams". Currently the sender still needs to go through the
> BaseChannel and MediaChannel, using an SSRC as a key.
>
> R=pthatcher@webrtc.org
>
> Committed: https://crrev.com/bc5831999d3354509d75357b659b4bb8e39f8a6c
> Cr-Commit-Position: refs/heads/master@{#13285}
TBR=pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review-Url: https://codereview.webrtc.org/2099843003
Cr-Commit-Position: refs/heads/master@{#13286}
This eliminates the need for the extra layer of indirection provided by
mediastreamprovider.h. It will thus make it easier to implement new
functionality in RtpSender/RtpReceiver.
It also brings us one step closer to the end goal of combining "senders"
and "send streams". Currently the sender still needs to go through the
BaseChannel and MediaChannel, using an SSRC as a key.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2046173002 .
Cr-Commit-Position: refs/heads/master@{#13285}
Reason for revert:
Didn't intend to land yet. Chromium CL still needed.
Original issue's description:
> Add virtual Initialize methods to PortAllocator and NetworkManager.
>
> This will allow PeerConnection to handle hopping to the right thread
> and doing thread-specific initialization for the PortAllocator.
> This eliminates a required thread-hop for whatever is passing the
> PortAllocator into CreatePeerConnection.
>
> BUG=617648
> R=pthatcher@webrtc.org, skvlad@webrtc.org
>
> Committed: https://crrev.com/a6bdb0990a659ff9e7c4374f5033a6bcc4fbfb21
> Cr-Commit-Position: refs/heads/master@{#13283}
TBR=pthatcher@webrtc.org,skvlad@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=617648
Review-Url: https://codereview.webrtc.org/2092023004
Cr-Commit-Position: refs/heads/master@{#13284}
This will allow PeerConnection to handle hopping to the right thread
and doing thread-specific initialization for the PortAllocator.
This eliminates a required thread-hop for whatever is passing the
PortAllocator into CreatePeerConnection.
BUG=617648
R=pthatcher@webrtc.org, skvlad@webrtc.org
Review URL: https://codereview.webrtc.org/2097653002 .
Cr-Commit-Position: refs/heads/master@{#13283}
This will allow media to be sent on these pairs before a binding
response is received, shortening call setup time. However, this is only
possible if the TURN servers don't require CreatePermission when
communicating with each other.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
NOTRY=True
Review-Url: https://codereview.webrtc.org/2063823008
Cr-Commit-Position: refs/heads/master@{#13268}
Reason for revert:
Breaking webrtc builder.
Original issue's description:
> Adding IceConfig option to assume TURN/TURN candidate pairs will work.
>
> This will allow media to be sent on these pairs before a binding
> response is received, shortening call setup time. However, this is only
> possible if the TURN servers don't require CreatePermission when
> communicating with each other.
>
> R=honghaiz@webrtc.org, pthatcher@webrtc.org
>
> Committed: https://crrev.com/8e6134eae4117a239de67c9a9dae8f5e3235d803
> Cr-Commit-Position: refs/heads/master@{#13263}
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
Review-Url: https://codereview.webrtc.org/2090823002
Cr-Commit-Position: refs/heads/master@{#13264}
This will allow media to be sent on these pairs before a binding
response is received, shortening call setup time. However, this is only
possible if the TURN servers don't require CreatePermission when
communicating with each other.
R=honghaiz@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2063823008 .
Cr-Commit-Position: refs/heads/master@{#13263}
VideoDecoderParams contains the id of the receive video
stream. Motivation behind this change is to enable down
stream apps easier map raw non-decoded data to incoming
streams.
BUG=b/28636393
Review-Url: https://codereview.webrtc.org/2052233002
Cr-Commit-Position: refs/heads/master@{#13250}
