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}
This helps show where ownership is transfered between objects.
Specifically, this CL wraps cricket::VideoCapturer, MediaEngineInterface
and DataEngineInterface in unique_ptr.
BUG=None
TBR=magjed@webrtc.org
Review-Url: https://codereview.webrtc.org/2685093002
Cr-Commit-Position: refs/heads/master@{#16548}
Currently filed trial value which controls H.264 high profile support is
read once only when factory is created. If field trial value is changed for
the next WebRTC call supported codec list need to be updated as well.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2685183004
Cr-Commit-Position: refs/heads/master@{#16543}
Create the RTCRtpReceiverDelegate which is a wrapper over
webrtc::RtpReceiverObserverInterface.
The callback will be called whenever the first rtp packet is received.
Related CL: https://codereview.webrtc.org/2531333003/
BUG=webrtc:6742
Review-Url: https://codereview.webrtc.org/2641923003
Cr-Commit-Position: refs/heads/master@{#16501}
This tag is supposed to be temporary and removed when all Chromium tests
have been migrated to JUnit4.
BUG=webrtc:7123,chromium:640116
NOTRY=True
Review-Url: https://codereview.webrtc.org/2683583002
Cr-Commit-Position: refs/heads/master@{#16465}
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}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
Reason for revert:
It seems that we cannot skip the generation of "//webrtc/base/base_java" in chromium without some refactoring because it is included as a dependency in some places.
Original issue's description:
> Revert of Creating libwebrtc bundle jar (patchset #4 id:60001 of https://codereview.webrtc.org/2646443002/ )
>
> Reason for revert:
> This breaks some chromium.webrtc.fyi buildbots with the following error:
>
> ERROR Unresolved dependencies.
> //third_party/webrtc/base:base(//build/toolchain/android:android_arm)
> needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
>
>
> Original issue's description:
> > Creating libwebrtc bundle jar
> >
> > Creates a JAR which includes:
> > - //webrtc/base:base_java
> > - //webrtc/modules/audio_device:audio_device_java
> > - //webrtc/sdk/android:libjingle_peerconnection_java
> > - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
> >
> > The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
> >
> > BUG=webrtc:6356
> >
> > Review-Url: https://codereview.webrtc.org/2646443002
> > Cr-Commit-Position: refs/heads/master@{#16189}
> > Committed: a62a82b7e7
>
> TBR=kjellander@webrtc.org,sakal@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2640023010
> Cr-Commit-Position: refs/heads/master@{#16190}
> Committed: 3c9151b953TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646093004
Cr-Commit-Position: refs/heads/master@{#16203}
The folder webrtc/sdk/ now contains android as well, so we should move the objc DEPS file to the objc folder.
TBR=tommi
BUG=None
Review-Url: https://codereview.webrtc.org/2644733008
Cr-Commit-Position: refs/heads/master@{#16191}
Reason for revert:
This breaks some chromium.webrtc.fyi buildbots with the following error:
ERROR Unresolved dependencies.
//third_party/webrtc/base:base(//build/toolchain/android:android_arm)
needs //third_party/webrtc/base:base_java(//build/toolchain/android:android_arm)
Original issue's description:
> Creating libwebrtc bundle jar
>
> Creates a JAR which includes:
> - //webrtc/base:base_java
> - //webrtc/modules/audio_device:audio_device_java
> - //webrtc/sdk/android:libjingle_peerconnection_java
> - //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
>
> The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
>
> BUG=webrtc:6356
>
> Review-Url: https://codereview.webrtc.org/2646443002
> Cr-Commit-Position: refs/heads/master@{#16189}
> Committed: a62a82b7e7TBR=kjellander@webrtc.org,sakal@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2640023010
Cr-Commit-Position: refs/heads/master@{#16190}
Creates a JAR which includes:
- //webrtc/base:base_java
- //webrtc/modules/audio_device:audio_device_java
- //webrtc/sdk/android:libjingle_peerconnection_java
- //webrtc/sdk/android:libjingle_peerconnection_metrics_default_java
The libwebrtc.jar file will be generated at '<output_dir>/lib.java/webrtc/sdk/android/libwebrtc.jar'.
BUG=webrtc:6356
Review-Url: https://codereview.webrtc.org/2646443002
Cr-Commit-Position: refs/heads/master@{#16189}
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}
Reason for revert:
The new method doesn't work as intended.
It can't pass ownership, because the StatsReports is a vector of raw pointers to StatReport objects owned by the StatsCollector.
Original issue's description:
> New method StatsObserver::OnCompleteReports, passing ownership.
>
> The new name, OnCompleteReports rather than OnComplete, is needed
> because in C++ method lookup, overriding a method hides all otherwise
> inherited methods with the same name, even if they have a different
> signature. And here, the intention is that each subclass should
> override one or the other of the two methods, and inherit the method it
> doesn't override.
>
> This cl is a prerequisite for
> https://codereview.webrtc.org/2567143003/, because the Chrome glue
> code needs to retain the stats report after the OnComplete method has
> returned.
>
> Currently, Chrome makes a copy of the stats mapping (which breaks when
> changing ValuePtr from an rtc::linked_ptr to an std::unique_ptr). After
> this cl, Chrome can be fixed to take ownership and no longer needs to
> copy anything, unblocking cl 2567143003.
>
> BUG=webrtc:6424
>
> Review-Url: https://codereview.webrtc.org/2584553002
> Cr-Commit-Position: refs/heads/master@{#15708}
> Committed: b36ee8d498TBR=solenberg@webrtc.org,magjed@webrtc.org,tkchin@webrtc.org,hbos@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2641783002
Cr-Commit-Position: refs/heads/master@{#16144}
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}
Move file capturer/renderer tests from the AppRTCMobile tests directory
to the WebRTC tests directory. These tests do not test AppRTCMobile but
rather WebRTC functionality. Therefore, they belong in WebRTC tests
directory.
BUG=webrtc:6545
Review-Url: https://codereview.webrtc.org/2632233002
Cr-Commit-Position: refs/heads/master@{#16115}
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}
Also did some slight refactoring of the code that turns constraints
into rtc::Optionals. Used a template method to avoid code duplication,
and used the same pattern for "CopyConstraintsIntoAudioOptions" as was
being used for "CopyConstraintsIntoRtcConfiguration".
BUG=webrtc:6752
Review-Url: https://codereview.webrtc.org/2628523003
Cr-Commit-Position: refs/heads/master@{#16063}
A decision has been made to not use prefix header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2590823002
Cr-Commit-Position: refs/heads/master@{#16056}
