Reports uninitialized-memory reads that seem to originate from when the
frame is copied. The test passes if we remove CPU optimizations from
libyuv, disabling test until we figure out whether it's an unsupported
instruction in DrMemory, bug in libyuv or bug in the test.
BUG=3754
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19149004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6995 4adac7df-926f-26a2-2b94-8c16560cd09d
This should save several gigabytes of traffic and disk space.
On Linux this is about 2.6 GB:
346M src/chrome/tools/test/reference_build
340M src/native_client
170M src/third_party/ffmpeg
1.5G src/third_party/WebKit
196M src/v8
BUG=2863
TESTED=Removed the directories locally, ran a sync and verified they didn't reappear (or fail because of platform-specific ones).
R=iannucci@chromium.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6984 4adac7df-926f-26a2-2b94-8c16560cd09d
Adressing clear races between the test thread and capturer thread shown
as heap-use-after-free in vpx_codec_destroy in
WebRtcVideoMediaChannelTest.SetSend (way later in the rest run).
When capturing a frame the test copied it to a separate frame that would
then be read by the test without synchronization, if the test didn't
manage to examine the frame in between captures the adapted frame would
be overwritten by the following frame during accesses to it.
The actual races are suppressed by race:webrtc/base/messagequeue.cc and
race:webrtc/base/thread.cc. These fixes reduce the suppression count
locally from around 3000 to 30 for VideoAdapterTest.*.
Also removing tsan suppressions for talk/base as it's been moved to
webrtc/base.
R=tommi@webrtc.org
BUG=3671
Review URL: https://webrtc-codereview.appspot.com/22169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6979 4adac7df-926f-26a2-2b94-8c16560cd09d
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.
The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.
BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
Adds a SCRIPT_VERSION and the target_os_list to the flag file content. The
script version is so that we can arbitrarially make all slaves/devs re-sync (in
case we change the implementation but don't want to roll chromium), and the
target_os_list is so that devs who change the target_os_list in their .gclient
file don't mysteriously fail to get the new deps.
R=kjellander@webrtc.org, agable@chromium.org, szager@chromium.org
BUG=2863, chromium:339647
Review URL: https://webrtc-codereview.appspot.com/17189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6952 4adac7df-926f-26a2-2b94-8c16560cd09d
This test verifies bit exactness for the send-side of ACM. The test
setup is a chain of three different test classes:
test::AcmSendTest -> AcmSenderBitExactness -> test::AcmReceiveTest
The receiver side is driving the test by requesting new packets from
AcmSenderBitExactness::NextPacket(). This method, in turn, asks for the
packet from test::AcmSendTest::NextPacket, which inserts audio from the
input file until one packet is produced. (The input file loops
indefinitely.) Before passing the packet to the receiver, the
AcmSenderBitExactness class verifies the packet header and updates a
payload checksum with the new payload. The decoded output from the
receiver is also verified with a (separate) checksum.
The current CL only adds tests for 30 ms and 60 ms iSAC. More codecs
will be added in coming changes.
BUG=3521
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6949 4adac7df-926f-26a2-2b94-8c16560cd09d
This test has been failing every now and then. This is likely due to the
random input that was used. With this change, the input is now read from
an audio file, making it identical on each run.
The encoding is moved to inside the main test loop, so that new data is
added with each packet. (Before this change, the same payload was added
over and over again; only the RTP header was updated.)
BUG=3715
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19079004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6948 4adac7df-926f-26a2-2b94-8c16560cd09d
An unsigned int was passed through %lu instead of %u (harmless on 32bit).
More seriously, a wide string was passed through %s, which means only the
first byte in the string got printed (since the 2nd byte is likely 0 in
UCS-2). Use %ls to include the whole string, even though it might not be
renderable in the target 8bit buffer.
BUG=chromium:82385
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22409004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6946 4adac7df-926f-26a2-2b94-8c16560cd09d
