./content/renderer/media/webrtc_audio_device_impl.h:19:81: error: third_party/webrtc/modules/audio_device/main/interface/audio_device.h: No such file or directory
We have to create a temporary files until the new webrtc is rolled the chrome.
Review URL: https://webrtc-codereview.appspot.com/867008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2905 4adac7df-926f-26a2-2b94-8c16560cd09d
These changes make the execution abort earlier on an error (like a tool is not found) and makes it easier to figure out what's wrong.
Made build_zxing.py executable.
BUG=None
TEST=Local runs of the PyAuto test src/chrome/test/functional/webrtc_video_quality.py in Chromium.
Review URL: https://webrtc-codereview.appspot.com/840005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2899 4adac7df-926f-26a2-2b94-8c16560cd09d
Also Suppressed FakeMediaProcess errors (bug 898) and took out a test and suppressed general errors (bug 332). Lastly, fixed memory leak in misc test.
BUG=898, 332
TEST=Ran voe_auto_test with repeat=10 through the whole standard suite, under valgrind. Ran without valgrind. Tested that the extended and standard tests still start and are reachable from the menu.
Review URL: https://webrtc-codereview.appspot.com/855009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2898 4adac7df-926f-26a2-2b94-8c16560cd09d
Added new helper for getting regular input. Rewrote remaining input handling for custom calls. Added a --choose_defaults flag which makes it possible to default on everything (e.g. with the flag, choosing custom call will accept all defaults and go directly to the call).
The next patch will add support for overriding arbitrary choices using flags. That is the point I want to arrive at and this patch paves the way for that. Fortunately it gets rid of some repetitive and bug-prone code on the way.
BUG=
Review URL: https://webrtc-codereview.appspot.com/858005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2878 4adac7df-926f-26a2-2b94-8c16560cd09d
Make sure that playing dummy RTP files works, i.e., RTP dumps with only
headers, no actual payloads. It turs out that some RTP dumps did not
state how long the actual packet was (including the discarded payload),
which resulted in packets with zero length payload being inserted into
NetEQ. Now, the dummy packet will contain at least one byte of payload
(set to zero).
Also added capability to filter out a specific SSRC from the RTP
dump file.
Finally, added SSRC printout from RTPanalyze, and corresponding
change in RTPchange.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/857007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2876 4adac7df-926f-26a2-2b94-8c16560cd09d
Ensured there are bugs for all open valgrind issues in the standard tests and suppressed the known issues. This way, we can get it running in continuous integration and keep new issues from entering.
Removed bad check in codec test, added suppressions.
Fixed simple memory leaks in tests.
BUG=Related to bug 329
TEST=Ran the vie_auto_test standard suite many times under valgrind to root out flakiness. Ran the standard suite without valgrind to ensure I didn't break anything.
Review URL: https://webrtc-codereview.appspot.com/843005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2854 4adac7df-926f-26a2-2b94-8c16560cd09d
Added new wrapper script webrtc_tests.bat for executing memory/threading tests on Windows.
Updated webrtc_tests.sh to include modifications in chrome_tests.sh that has happened since we copied it.
To setup TSAN for Windows, see http://www.chromium.org/developers/how-tos/using-valgrind/threadsanitizer
I did like this:
1. Added "third_party/tsan": "http://src.chromium.org/chrome/trunk/deps/third_party/tsan"
to custom_deps in my .gclient file
2. gclient sync
3. SET GYP_DEFINES=build_with_tool=tsan && gclient runhooks
4. Compiled.
5. Ran the test using the wrapper script (see below).
To setup Dr Memory for Windows, see http://www.chromium.org/developers/how-tos/using-drmemory
I did like this:
1. Added "third_party/drmemory": "http://src.chromium.org/svn/trunk/deps/third_party/drmemory",
to custom_deps in my .gclient file (using the drmemory.DEPS as described on Chromium's wiki ends up in the wrong location)
2. gclient sync
3. SET GYP_DEFINES=build_with_tool=drmemory && gclient runhooks
4. Compiled.
5. Ran the test using the wrapper script (see below).
TEST=
On Windows:
tools\valgrind-webrtc\webrtc_tests.bat --tool=tsan --test build\Debug\voice_engine_unittests.exe
tools\valgrind-webrtc\webrtc_tests.bat --tool=drmemory_light --test build\Debug\voice_engine_unittests.exe
tools\valgrind-webrtc\webrtc_tests.bat --tool=drmemory_full --test build\Debug\voice_engine_unittests.exe
On Linux:
tools/valgrind-webrtc/webrtc_tests.sh --tool=memcheck --test out/Release/test_support_unittests
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan --test out/Release/test_support_unittests
tools/valgrind-webrtc/webrtc_tests.sh --tool=asan --test out/Release/test_support_unittests
Review URL: https://webrtc-codereview.appspot.com/845004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2846 4adac7df-926f-26a2-2b94-8c16560cd09d
