The purpose of this CL is to use jlongFromPointer() for converting frame pointers to jlong instead of implicit casts which is not safe.
In order to respect constness, I had to make a small helper function for this.
BUG=webrtc:4993
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1373233002 .
Cr-Commit-Position: refs/heads/master@{#10108}
It turns out that if a bot is interrupted during test execution,
the existence of this directory causes the next build to fail.
This usually happens when the Buildbot master is restarted.
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1376723002 .
Cr-Commit-Position: refs/heads/master@{#10105}
Reason for revert:
The top row in the video stream from the camera is messed up. The byte[] pointer is not the same as GetDirectBufferAddress() apparently.
Original issue's description:
> Android VideoCapturer: Send ByteBuffer instead of byte[]
>
> The purpose with this CL is to replace GetByteArrayElements() and ReleaseByteArrayElements() with GetDirectBufferAddress().
>
> R=hbos@webrtc.org
>
> Committed: https://crrev.com/cb3649b40b3fd6d5bbb0a92003b717e46ce90924
> Cr-Commit-Position: refs/heads/master@{#10091}
TBR=hbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1377783002
Cr-Commit-Position: refs/heads/master@{#10103}
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).
Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).
Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/
Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).
Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS
Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh
BUG=481034, 535973
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1355083002
Cr-Commit-Position: refs/heads/master@{#10101}
This CL essentially converts [Local|Remote]TrackHandler to
Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender.
It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer,
since these classes weren't really anything more than containers.
PeerConnection now manages the RtpSenders and RtpReceivers directly.
Review URL: https://codereview.webrtc.org/1351803002
Cr-Commit-Position: refs/heads/master@{#10100}
This is the same sort of thing we do in C++ end-to-end PeerConnection
tests.
Review URL: https://codereview.webrtc.org/1361213002
Cr-Commit-Position: refs/heads/master@{#10098}
Fixes code formatting and uses size_t properly. Also makes use of
IsNewerTimestamp instead of a simple > check, which should fix an
edge-case bug.
BUG=
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1358863002
Cr-Commit-Position: refs/heads/master@{#10094}
whose network has ever been removed. It is unlikely the sockets/ports/candidates created from
those AllocationSequences will still be valid.
BUG=
Review URL: https://codereview.webrtc.org/1361183004
Cr-Commit-Position: refs/heads/master@{#10093}
p2ptransportchannel. This CL does not use the new policy yet.
BUG=
Review URL: https://codereview.webrtc.org/1369773003
Cr-Commit-Position: refs/heads/master@{#10092}
The purpose with this CL is to replace GetByteArrayElements() and ReleaseByteArrayElements() with GetDirectBufferAddress().
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1372813002 .
Cr-Commit-Position: refs/heads/master@{#10091}
The playout mode in NetEq can still be set through the constructor
configuration.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1362943004
Cr-Commit-Position: refs/heads/master@{#10089}
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
This cl moves back loading java SurfaceTextureHolder to the ClassReferenceHolder and use FindClass through ClassReferenceHolder. Without this, jni->FindClass returns nullptr in surfacetexturehelper_jni.cc.
BUG=webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1370013002 .
Cr-Commit-Position: refs/heads/master@{#10086}
This CL also makes some small non-functional changes in ThreadUtils and EglBase to support SurfaceTextures and SurfaceTextureHelper.
BUG=webrtc:4993
R=hbos@webrtc.org
Review URL: https://codereview.webrtc.org/1368093003 .
Cr-Commit-Position: refs/heads/master@{#10085}
This cl adds a C++ counterpart of the Java SurfaceTextureHandler. It can be used for creating a webrtc::VideoFrames from a native handle and also guarantee that the Java SurfaceTexture is notified when the VideoFrame is no longer in use.
BUG=webrtc:4993
R=magjed@webrtc.org
Review URL: https://codereview.webrtc.org/1366413003 .
Cr-Commit-Position: refs/heads/master@{#10084}
This updates the isolate.gypi copies we have to maintain in our
code repo to Chromium's revision 310ea93.
The changes about generating .isolated.gen.json files are needed
to support running with Swarming (https://www.chromium.org/developers/testing/isolated-testing)
Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that's added to our links
script.
In order to use isolate_driver.py, the .isolate files must be in the
same directory as the test_name_run target is defined, which meant
I had to move around some of the isolate files and targets below
webrtc/modules.
BUG=497757
R=maruel@chromium.orgTBR=henrik.lundin@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org
TESTED=Clobbered trybots:
git cl try -c --bot=linux_compile_rel --bot=mac_compile_rel --bot=win_compile_rel --bot=android_compile_rel --bot=ios_rel -m tryserver.webrtc
Review URL: https://codereview.webrtc.org/1373513002 .
Cr-Commit-Position: refs/heads/master@{#10081}
It has become extra flaky lately, and is preventing people from
using the CQ.
BUG=webrtc:4958
Review URL: https://codereview.webrtc.org/1368763002
Cr-Commit-Position: refs/heads/master@{#10080}
Android hardware H.264 seems to keep a steady high-QP flow instead of
dropping frames, so framedrops aren't sufficient to detect a bad state
where downscaling would be beneficial.
BUG=webrtc:4968
R=magjed@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1364253002 .
Cr-Commit-Position: refs/heads/master@{#10078}
Fix an issue where using setNeedsDisplay on a GLKView which has a frame
with size zero will make GLKView/iOS output the following error:
Failed to bind EAGLDrawable: <CAEAGLLayer: 0x1742282e0> to
GL_RENDERBUFFER 1 Failed to make complete framebuffer object 8cd6
(The error code 8cd6 corresponds to
GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT.)
GLKView will internally setup it's render buffer when the delegate is
about to draw into it. Previously when enableSetNeedsDisplay was set to
YES (default), then GLKView would still attempt to setup it's internal
buffer even if it's frame size is zero and that would cause
GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT.
By using enableSetNeedsDisplay = NO, RTCEAGLVideoView can guard against
calling -[GLKView display] if it's current frame size is empty.
Review URL: https://codereview.webrtc.org/1347013002
Cr-Commit-Position: refs/heads/master@{#10076}
Ensures that we can restart audio recording on Android without hitting
a DCHECK. Also adds a symmetric design for the playout side.
BUG=webrtc:5000
TEST=modules_unittests --gtest_filter=AudioDevice*
Review URL: https://codereview.webrtc.org/1373443003
Cr-Commit-Position: refs/heads/master@{#10072}
Ensure the checkout is updated _before_ reading the current
revision from the DEPS file. This will avoid errors when a checkout
is not up to date, as updating happened after reading the revision
before this change.
BUG=webrtc:4688
R=phoglund@webrtc.org
Review URL: https://codereview.webrtc.org/1361923004 .
Cr-Commit-Position: refs/heads/master@{#10071}
Reason for revert:
Breaking Chromium FYI bots.
Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}
TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368933002
Cr-Commit-Position: refs/heads/master@{#10069}
This CL adds a slider that can change capture resolution and fps during a call. The camera will no be reconfigured, but the frames will be downscaled/dropped in software by cricket::VideoAdapter in the cricket::VideoCapturer. This is controlled with VideoCapturerAndroid.onOutputFormatRequest(). The slider is turned off by default and can be enabled with a checkbox under 'WebRTC Video Settings'.
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1361083002 .
Cr-Commit-Position: refs/heads/master@{#10067}
Instead of FATAL on a bad codec specification, log and return an error
code. This is a band-aid until callers are taught to only give it good
specifications.
BUG=webrtc:5033, chromium:526478
Review URL: https://codereview.webrtc.org/1364193002
Cr-Commit-Position: refs/heads/master@{#10066}
This will allow Chromium's build files to be updated, so that when the
real RtpSender CL is submitted, it doesn't break the FYI bots.
Review URL: https://codereview.webrtc.org/1364813004
Cr-Commit-Position: refs/heads/master@{#10065}
Otherwise, we may delete a useful connection because the current best connection may be failing.
BUG=
Review URL: https://codereview.webrtc.org/1364683002
Cr-Commit-Position: refs/heads/master@{#10063}
