Replaces VideoCodec in VideoSendStream::Config with an EncoderSettings
struct. The EncoderSettings struct uses an external encoder for all
codecs. This means that external users, such as libjingle, will provide
the encoders themselves, removing the previous distinction of internal
and external codecs.
For now VideoSendStream translates to VideoCodec internally. In the
interrim (before the corresponding change is implemented in
VideoReceiveStream) tests convert EncoderSettings to VideoCodecs.
Removes Call::GetVideoCodecs().
Disables RampUpTest.WithPacingAndRtx as its further exposed with changes
to bitrates used in tests.
BUG=2854,2992
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5722 4adac7df-926f-26a2-2b94-8c16560cd09d
We've had problems that seem to manifest in run_tests.mm getting stuck
on exit. For our automated test targets only full_stack.cc was making
use of the platform-specific renderers provided by webrtc_test_common
and since no one currently monitors these the use case is hypothetical.
Readding platform-specific renderers to video_loopback is tracked with
issue 3039, though as far as I'm aware no one's currently using the
video_loopback target.
BUG=2987
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5686 4adac7df-926f-26a2-2b94-8c16560cd09d
Optionally prevents doing a frame copy when putting frames into a
VideoSendStream. PutFrame() is still there, which copies the frame.
Also removes time_since_capture_ms as a parameter, since
I420VideoFrame::render_time_ms() denotes when the frame was captured.
BUG=2657
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5119004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5265 4adac7df-926f-26a2-2b94-8c16560cd09d
Before SSRCs were not set up correctly, as the old VideoEngine API
doesn't support setting additional SSRCs before a codec with as many
streams are set.
No test was in place to catch this, so two tests are added to make sure
that we send the SSRCs that are set, and also that we can switch from
using one to using all SSRCs, even though initially not all of them are
set up.
BUG=
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5188 4adac7df-926f-26a2-2b94-8c16560cd09d
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.
- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.
TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2449005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
When WebRTC is built as a part of Chromium, some of
the stuff in webrtc.gyp will not be found. This CL
fixes this.
TEST=trybots passing. I also did some manual builds for Android with the android_builder_webrtc target in build/all_android.gyp of a Chromium checkout.
BUG=chromium:304143
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2353004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4949 4adac7df-926f-26a2-2b94-8c16560cd09d
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.
The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.
TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).
I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).
I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.
Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc
BUG=1916
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
To run this test, the executable in
libjingle_peerconnection_objc_test.app/Contents/MacOS/libjingle_peerconnection_objc_test
must be run. Running executables like this is not supported by
the buildbot scripts, which makes the paths to the stdio invalid.
Adding it to our command-line wrapping script and executing it as
'libjingle_peerconnection_objc_test' on the bots should work around
this limitation.
BUG=2120
TEST=locally building and executing the test on Mac using:
GYP_DEFINES="host_arch=x64 target_arch=x64" gclient runhooks
ninja -C out/Release
out/Release/buildbot_tests.py -t libjingle_peerconnection_objc_test
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2275007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4826 4adac7df-926f-26a2-2b94-8c16560cd09d
Allows us to ignore flags passed on to us by Chromium build bots
without having to explicitly disable them. (Thanks pbos!)
TESTED=Running modules_unittests with a bogus flag did not result in an
error.
TBR=kjellander
Review URL: https://webrtc-codereview.appspot.com/2222005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4757 4adac7df-926f-26a2-2b94-8c16560cd09d
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.
Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort
Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled: 666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled: 673 ms (1.01x)
BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2160005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
On Windows, when tests are run under memory tools like
DrMemory and TSan, slashes occur in the path as directory
separators. This makes the fileutils.cc fail to figure out
the project root, making paths to resource files invalid.
By replacing slashes in the path with backslashes for Windows
this can be fixed.
BUG=2318
TEST=running the same command as the bot locally, succeeding after this patch is applied.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2137004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4640 4adac7df-926f-26a2-2b94-8c16560cd09d
The metrics_unittests tried to write an empty file to the
current working directory, which caused a crash on Android. By
using the output folder instead, this is avoided.
This makes it possible to re-enable the tests disabled in r4474
and also remove the empty dummy test added in r4627.
BUG=1882,
TEST=Setup a Chromium+WebRTC checkout as described in issue 1882
. build/android/envsetup.sh
export GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1"
gclient runhooks
ninja -C out/Release metrics_unittests_apk
build/android/test_runner.py gtest -s metrics_unittests --verbose --release
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2127006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4634 4adac7df-926f-26a2-2b94-8c16560cd09d
