Reason for revert:
Suspect of breaking Chrome FYI bots.
See
https://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac%20Builder/builds/23065https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Builder
Example logs:
../../content/renderer/media/gpu/rtc_video_encoder_unittest.cc:18:46: fatal error: third_party/webrtc/video_encoder.h: No such file or directory
#include "third_party/webrtc/video_encoder.h"
^
Original issue's description:
> Move video_encoder.h and video_decoder.h to /api and create GN targets for them
>
> BUG=webrtc:5881
> # Because PRESUBMIT ignores LINT blacklist for moved files and these
> # headers have some not easy to resolve issues.
> NOPRESUBMIT=True
>
> Review-Url: https://codereview.webrtc.org/2780943003
> Cr-Commit-Position: refs/heads/master@{#17511}
> Committed: c42f540570TBR=solenberg@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5881
Review-Url: https://codereview.webrtc.org/2794033002
Cr-Commit-Position: refs/heads/master@{#17514}
BUG=webrtc:5881
# Because PRESUBMIT ignores LINT blacklist for moved files and these
# headers have some not easy to resolve issues.
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2780943003
Cr-Commit-Position: refs/heads/master@{#17511}
We currently leak one local reference to MediaCodecVideoDecoder in
every call to MediaCodecVideoDecoderFactory::CreateVideoDecoder. After
the decoder has been re-initialized 512 times, JNI will crash due to
local reference table overflow (max=512).
The actual leak is in the member initializer list of
MediaCodecVideoDecoder. This CL fixes the leak by adding a
ScopedLocalRefFrame outside of the ctor. All JNI code that originate
from a C++ thread (i.e. the entry point is not a Java thread) must use
a ScopedLocalRefFrame in order to avoid leaking local references.
BUG=webrtc:6969,b/36713034
Review-Url: https://codereview.webrtc.org/2780273002
Cr-Commit-Position: refs/heads/master@{#17464}
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}
The check: 'RTC_CHECK_GE(slice_height, height);' has been observed to
fail after a reconfig. It looks like |slice_height| is still using the
previous resolution. |slice_height| isn't used for texture output and
hopefully this issue is texture specific. This CL only extracts and
checks |slice_height| when it's actually used.
BUG=b/35932686
Review-Url: https://codereview.webrtc.org/2736603003
Cr-Commit-Position: refs/heads/master@{#17065}
When textures are not enabled and we are using byte buffer outputs, the
decoder is currently crashing for odd heights because of an RTC_CHECK.
This CL removes the check and handles the pointer offset to the chroma
planes for the odd height case instead.
This has been verified to work correctly on a Pixel device.
BUG=webrtc:6651
Review-Url: https://codereview.webrtc.org/2709923005
Cr-Commit-Position: refs/heads/master@{#16805}
After this change, all calls to MediaCodecVideoEncoder must be made on
the same task queue. Removes OnCodecThread suffix from methods since it
is no longer meaningful.
BUG=webrtc:6290
Review-Url: https://codereview.webrtc.org/2669093004
Cr-Commit-Position: refs/heads/master@{#16792}
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}
If android_setsocknetwork() is available, and it fails, then bind()
should *not* be called, and an error should be returned.
If it succeeds, then bind should be called, but with an "any" address.
This is to prevent cases where sockets are sent with a source address
that doesn't match the network interface they're sent on. See bug below.
This CL also changes "NetworkBinderResults" to an enum class, and
renames it to "NetworkBinderResult".
BUG=webrtc:7026
Review-Url: https://codereview.webrtc.org/2646863005
Cr-Commit-Position: refs/heads/master@{#16597}
To ensure compliance with older version high profile should appear in local SDP
before baseline profile.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2696733002
Cr-Commit-Position: refs/heads/master@{#16596}
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}
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}
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}
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}
