From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL adds support for reading .y4m files to the infra in
video_quality_analysis.cc, adding new functions
ExtractFrameFromYuvFile() and ExtractFrameFromY4mFile(),
instad of the previous ExtractFrameFromI420(). The decision
as to which one to use is taken from the file extension,
if it is .y4m then is considered a YUV4MPEG file, otherwise
is taken as a raw .yuv file.
It also removes the pseudo duplicated function
GetNextI420Frame(), that is used from psnr_ssim_analyzer.c,
and adds support for y4m files there.
Tested/validated via local compile-run.
YUV4MPEG is a trivial container with a file header
and a per-frame header, see [1]
[1]
http://wiki.multimedia.cx/index.php?title=YUV4MPEG2
BUG=https://code.google.com/p/chromium/issues/detail?id=343504
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5702 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
On Windows, sometimes the compare_videos.py script fails because
the inherited stdin handle from the parent process fails.
It seems we can work around this by passing null to stdin to
the subprocesses, since we're not using it anyway.
The errors looked like this:
Traceback (most recent call last):
File "C:\b\build\slave\Win7_Tester\build\src\third_party/webrtc/tools/compare_videos.py", line 116, in <module>
sys.exit(main())
File "C:\b\build\slave\Win7_Tester\build\src\third_party/webrtc/tools/compare_videos.py", line 91, in main
barcode_decoder = subprocess.Popen(cmd, stdout=sys.stdout, stderr=sys.stderr)
File "C:\b\depot_tools\python_bin\lib\subprocess.py", line 588, in __init__
errread, errwrite) = self._get_handles(stdin, stdout, stderr)
File "C:\b\depot_tools\python_bin\lib\subprocess.py", line 686, in _get_handles
p2cread = GetStdHandle(STD_INPUT_HANDLE)
WindowsError: [Error 6] The handle is invalid
Example from http://build.chromium.org/p/chromium.webrtc/builders/Win7%20Tester/builds/4498/steps/webrtc_manual_browser_tests_test/logs/stdio
BUG=302915
TEST=successful runs on Windows and Linux.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2334005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4902 4adac7df-926f-26a2-2b94-8c16560cd09d
We encounter a sample-underrun if NetEq is initialized with a sampling rate fs =16000 and receive Opus packets with frame-size less than 5 ms. The reason is as follows.
Let say NetEq buffer has 4 packets of Opus each of size 2.5ms this means that internally timestamp of packets incremented by 80 (internally Opus treated as 32 kHz codec). Given the initial sampling rate of NetEq, at the first time that it wants to fetch packets, it targets to fetch 160 samples. Therefore, it will only extracts 2 packets. Decoding these packets give us exactly 160 samples (5 ms at 32 kHz), however, upon decoding the first packet the internal sampling rate will be updated to 32 kHz. So it is expected that sync buffer to deliver 320 samples while it does only have 160 samples (or maybe few more as it starts with some zeros). And we encounter and under-run.
Even if we ignore the under-run "assert(sync_buffer_->FutureLength() >= expand_->overlap_length())" (neteq_impl.cc::811) is trigered. I'm not sure what happens if we remove this assert perhaps NetEq will work fine in subsequent calls. However the first under-run is blocking ACM2 test to pass.
Here I have a solution to update sample rate as soon as a packet is inserted, if required. It not a very efficient approach as we do the same reset in NetEqImpl::Decode().
It is a bit tricky to reproduce this because the TOT ACM tests do not run ACM2. In https://webrtc-codereview.appspot.com/2192005/ I have a patch to run both ACMs. To reproduce the problem, one can patch that CL and run
$ out/Debug/modules_tests --gtest_filter=AudioCodingModuleTest.TestOpus
Note that we would not encounter any problem if NetEq4 is initiated with 32000 Hz sampling rate. You can test this by setting |kNeteqInitSampleRateHz| to 32000 in webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
BUG=
R=andrew@webrtc.org, henrik.lundin@webrtc.org, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2306004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4896 4adac7df-926f-26a2-2b94-8c16560cd09d
Add support for --label flag to the frame_analyzer, that
decides what label shall be used for the perf output.
BUG=none
TEST=
Make sure to have zxing and ffmpeg in the PATH.
Create a captured video (from running vie_auto_test custom call)
webrtc/tools/compare_videos.py --ref_video=reference_video.yuv --test_video=captured_output.yuv --frame_analyzer=out/Release/frame_analyzer --label=TEST_VGA
And then inspecting the output that is prefixed with RESULT.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4714 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a re-land attempt of http://review.webrtc.org/1673004/
It now includes a build/isolate.gypi in WebRTC that includes the same
file as the one that would be included when WebRTC is used in a Chromium
checkout. It is needed since it is not possible to use variables in GYP's
includes sections.
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing
Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
actually executing it:
tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
tools/swarm_client/googletest/fix_test_cases.py --isolated out/Release/testname.isolated
All tests that run on the bots for WebRTC has got _run target
and .isolate file created.
"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_tests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests
Tests that requires bare-metal and audio/video devices:
* audio_device_tests
* video_capture_tests
I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_tests
* vie_auto_test
* voe_auto_test
TEST=running isolate.py as described above. WebRTC trybots passing. Created a Chromium checkout with third_party/webrtc ToT and this patch applied, passing the runhooks step.
BUG=1916
R=henrike@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2056004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4590 4adac7df-926f-26a2-2b94-8c16560cd09d
As this breaks the FYI bots in
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)
> Isolate GYP target and .isolate files for tests
>
> Implemented according to the instructions at
> http://www.chromium.org/developers/testing/isolated-testing
>
> Workflow has been like this:
> 1. create _run GYP target
> 2. create a stripped down .isolate file
> 3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
> actually executing it:
> tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
> tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
>
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
>
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
>
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
>
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
>
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/1673004TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2040004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing
Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
actually executing it:
tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
All tests that run on the bots for WebRTC has got _run target
and .isolate file created.
"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_integrationtests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests
Tests that requires bare-metal and audio/video devices:
* audio_device_integrationtests
* video_capture_integrationtests
I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_integrationtests
* vie_auto_test
* voe_auto_test
TEST=running isolate.py as described above.
BUG=1916
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1673004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
Now that these scripts are called from browser tests, we need to print everything on stdout since the tests will throw away stderr when invoking programs. I chose to assign sys.stderr to sys.stdout. Otherwise I would have missed stuff like parser.error, which print to stderr.
The error message will get improved because the old code did not catch the case when the binary was missing, which lead to a very confusing error when that was the case. This gets fixed now.
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1886004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4416 4adac7df-926f-26a2-2b94-8c16560cd09d
This test was initializing strings for reference video files in a static
context, which makes is against the style guide and also makes the paths
become invalid when the test is launched from a working directory
outside the checkout.
Moving the initialization into the test fixture solves this.
BUG=none
TEST=Local execution launched from a directory outside the checkout tree on Win, Mac and Linux + trybots (for
compilation as they don't yet run the tools_unittests).
Review URL: https://webrtc-codereview.appspot.com/1178004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3647 4adac7df-926f-26a2-2b94-8c16560cd09d
By using the C++ version of Zxing, we can avoid having Java and Ant
as a dependency when running Video quality analysis on the bots.
This makes it far more easy to setup automation on new machines.
I also moved the scripts into the webrtc/ folder so it will be synced by default when building in Chrome (eliminating the need of a separate solution).
This CL also removes the need of the FFMPEG_HOME variable and replaces
its use with a command line flag to make the tool run smoothly on
Windows.
BUG=none
TEST=locally running the script on Windows, Mac and Linux.
Review URL: https://webrtc-codereview.appspot.com/1099007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3640 4adac7df-926f-26a2-2b94-8c16560cd09d
