Modification affects EglRenderer on Android. Moves frame dropping to the
renderer thread. Frame listeners are triggered even when FPS reduction is
active unless applyFpsReduction is set to true.
BUG=webrtc:7149
Review-Url: https://codereview.webrtc.org/2688843002
Cr-Commit-Position: refs/heads/master@{#17206}
Add an attribute to the RTCConfiguration which can be used by specific
mobile devices so that the IPv6 ICE candidates on WiFi will not be collected.
BUG=b/35725283
Review-Url: https://codereview.webrtc.org/2731813002
Cr-Commit-Position: refs/heads/master@{#17100}
Moves CameraCapturer, CameraSession, Camera1Session and Camera2Session
away from the public API.
BUG=webrtc:7172
Review-Url: https://codereview.webrtc.org/2699713004
Cr-Commit-Position: refs/heads/master@{#16723}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16675}
This is necessary in case the drawer doesn't cover all the pixels.
BUG=None
Review-Url: https://codereview.webrtc.org/2704663002
Cr-Commit-Position: refs/heads/master@{#16671}
Previously, was only checking the Android SDK version. But it also needs
to check for the presence of the connectivity manager service.
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2697943002
Cr-Commit-Position: refs/heads/master@{#16631}
Reason for revert:
Breaks AppRTCMobile interoperability. The ICE candidate URL shouldn't be signaled between endpoints, it's only there for informational purposes.
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Original-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16615}
> Committed: 45efce01c7TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2699533002
Cr-Commit-Position: refs/heads/master@{#16616}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Original-Commit-Position: refs/heads/master@{#16593}
Committed: 8586c8ee88
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16615}
Reason for revert:
Breaks downstream application's build
Original issue's description:
> Add the url attribute to the IceCandidate (Java Wrapper)
>
> The url of the ICE server is added to the IceCandiate class.
> This can be used to tell which server this candidate was gathered from.
>
> BUG=webrtc:7128
>
> Review-Url: https://codereview.webrtc.org/2690593002
> Cr-Commit-Position: refs/heads/master@{#16593}
> Committed: 8586c8ee88TBR=magjed@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2692993002
Cr-Commit-Position: refs/heads/master@{#16595}
The url of the ICE server is added to the IceCandiate class.
This can be used to tell which server this candidate was gathered from.
BUG=webrtc:7128
Review-Url: https://codereview.webrtc.org/2690593002
Cr-Commit-Position: refs/heads/master@{#16593}
These structs will be used for ORTC objects (and their WebRTC
equivalents).
This CL also introduces some minor changes to the existing implemented
structs:
- max_bitrate_bps uses rtc::Optional instead of "-1 means unset"
- "mime_type" turned into "name"/"kind" (which can be used to form the
MIME type string, if needed).
- clock_rate and channels changed to rtc::Optional, since they will
need to be for RtpSender.send().
- Renamed "channels" to "num_channels" (the ORTC name, which I prefer).
BUG=webrtc:7013, webrtc:7112
Review-Url: https://codereview.webrtc.org/2651883010
Cr-Commit-Position: refs/heads/master@{#16437}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Adds ignore for all lint errors in Chromium code. Changes minimum SDK for
instrumentation tests to 16 from 14. Adds TargetApi annotations.
BUG=webrtc:6597
Review-Url: https://codereview.webrtc.org/2670473004
Cr-Commit-Position: refs/heads/master@{#16412}
Moves setting state as stopped to stopInternal. Checks that state is not
stopped in stopInternal.
BUG=webrtc:7015
Review-Url: https://codereview.webrtc.org/2640093003
Cr-Commit-Position: refs/heads/master@{#16182}
This allows Camera2Session to correctly signal camera disconnect
when starting the camera.
BUG=webrtc:7008
Review-Url: https://codereview.webrtc.org/2642703002
Cr-Commit-Position: refs/heads/master@{#16142}
Old implementation also had a bug where it had the framebuffer bound
while changing the texture size. This causes problems on some phones.
BUG=webrtc:6751
Review-Url: https://codereview.webrtc.org/2635133003
Cr-Commit-Position: refs/heads/master@{#16141}
For low bitrates actual encoder output bitrate may differ from target
by 3-4 times.
Also adjust adaptation speed based on bitrate variation from the target.
BUG=b/34233384
R=wzh@webrtc.org
Review-Url: https://codereview.webrtc.org/2640543003 .
Cr-Commit-Position: refs/heads/master@{#16128}
Calling event handlers when the camera is closing is safe because
CameraCapturer checks if the errors are coming from the current session.
Calling onFailure after camera has already started might lead to strange
behavior.
BUG=b/34112992
Review-Url: https://codereview.webrtc.org/2634973002
Cr-Commit-Position: refs/heads/master@{#16101}
Ignore CAMERA_ERROR_EVICTED error if camera is about to be closed.
This is valid use case when other app is opening camera while
WebRTC Android app is trying to close it.
BUG=b/34112992
Review-Url: https://codereview.webrtc.org/2627153002
Cr-Commit-Position: refs/heads/master@{#16039}
Also enable Intel HW Vp8 encoder by default in AppRTCMobile.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2614373004
Cr-Commit-Position: refs/heads/master@{#16002}
The purpose is to be able to add field trials in Java code.
BUG=webrtc:6683
Review-Url: https://codereview.webrtc.org/2621003002
Cr-Commit-Position: refs/heads/master@{#15994}
The intention of SetConfiguration is that it modifies the configuration,
while keeping the constraints passed into CreatePeerConnection. Right
now that's now happening. See bug for more explanation.
BUG=webrtc:6942
Review-Url: https://codereview.webrtc.org/2603653002
Cr-Commit-Position: refs/heads/master@{#15974}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
Created a java wrapper for the callback OnAddTrack in this CL since it has been added to native C++ API
The callback function is called when a track is signaled by remote side and a new RtpReceiver is created.
The application can tell when tracks are added to the streams by listening to this callback.
BUG=webrtc:6112
Review-Url: https://codereview.webrtc.org/2513723002
Cr-Commit-Position: refs/heads/master@{#15745}
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.
Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 1) The policy is always explicitly initialized to secure.
> 2) API users have to explicitly integrate with the feature to
> use it, and will otherwise get no change in behavior.
> 3) The feature is not immediately exposed in non-native
> contexts. For example, disabling of certificate validation
> is not implemented via URI parsing since this would
> immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9eTBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS.
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
Create the RtpReceiver.Observer which is a Java wrapper over the webrtc::RtpReceiverObserverInterface.
The callback function onFirstPacketReceived will be called whenever the first audio or video packet it received.
BUG=webrtc:6742
Review-Url: https://codereview.webrtc.org/2531333003
Cr-Commit-Position: refs/heads/master@{#15464}
I decided to make one webrtc/sdk/android/BUILD.gn for both tests and Java/jni src.
External dependencies needs to be updated after this CL.
Future work is required to clean up the Android api and move
implementation details to /webrtc/sdk/android/src.
BUG=webrtc:5882,webrtc:6804
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2547483003
Cr-Commit-Position: refs/heads/master@{#15443}
