* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
- sqcif7 at 30 kbps: MediaCodec and libvpx.
- 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
BUG=webrtc:8219
Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.
The cpplint complaint is:
Include the directory when naming .h files [build/include] [4]
This CL disables the error but we have to remove these two headers
from the root directory.
NOPRESUBMIT=true
Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
This CL does not attempt to fix the style issues in the moved tb_ files, at least not yet. In general I've tried to avoid dependencies between the library and vie_auto_test: vie_auto_test depends on the library but not the other way around. I had to make some slight changes to achieve this. I had to remove some ViETest::Log statements in tb_interfaces.cc and I had to move knowledge of where to put output files to the library. I think it ended up being pretty clean in the end but let me know if I missed something. I tried to convert all paths I touched to src-relative paths, so look out if I missed something there.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/450004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1923 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose for the xvfb mode is to be able to run tests on the windowless environment on the Chromebot. Given the right input video, we can then write a relatively simple algorithm to analyze the screenshots and thereby conclude that video is playing.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/447004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1890 4adac7df-926f-26a2-2b94-8c16560cd09d
Restructured paths to new directory layout.
Stefan: common_video/*
Magnus: video_engine/*
Niklas: Android.mk
BUG=
TEST=jpeg_unittests on Debug+Release on Linux, Mac, Windows. Valgrind on Linux passes without warnings.
Review URL: https://webrtc-codereview.appspot.com/388007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1691 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed reproducable random behavior in packet_manipulator.h.
Test is now fully reproducable (runs on only one core) so much tighter limits are now set for the SSIM/PSNR values for the encoding/decoding (verified on all platforms)
BUG=
TEST=out/Debug/video_codecs_test_framework_integrationtests in Debug+Release on Linux, Mac, Windows and in Linux Valgrind.
Review URL: https://webrtc-codereview.appspot.com/381005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1649 4adac7df-926f-26a2-2b94-8c16560cd09d
The files are shorter (7 s) with one set provided for each sample rate.
Will be accompanied by the following set of files in the resource bundle:
far8_stereo.pcm
far16_stereo.pcm
far32_stereo.pcm
near8_stereo.pcm
near16_stereo.pcm
near32_stereo.pcm
BUG=114
TEST=audioproc_unittest
Review URL: https://webrtc-codereview.appspot.com/380003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1617 4adac7df-926f-26a2-2b94-8c16560cd09d
- Only showing text about browser needing WebRTC support if support not detected. Text is now contains more information and link to blog post.
- Removed the debug buttons.
- Clarifications and corrections in the readme file.
Review URL: https://webrtc-codereview.appspot.com/352015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1491 4adac7df-926f-26a2-2b94-8c16560cd09d
Performance verified on a few 32 kHz files.
BUG=
TEST=audioproc, audioproc_unittest
Updated output_data_float.pb
Changes in SWB tests (3, 6, 9 and 12) as
Running test 3 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 1363
Expected: test->max_output_average()
Which is: 1386
Running test 6 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 2070
Expected: test->max_output_average()
Which is: 2109
Running test 9 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 1314
Expected: test->max_output_average()
Which is: 1336
Running test 12 of 12...
src/modules/audio_processing/test/unit_test.cc:1182: Failure
Value of: max_output_average
Actual: 2049
Expected: test->max_output_average()
Which is: 2085
Review URL: https://webrtc-codereview.appspot.com/344013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1465 4adac7df-926f-26a2-2b94-8c16560cd09d
Example: .../webrtc_test.html?server=foo
This simplifies when one has to close and re-open the browser several times or use different servers and names, since it can be stored as bookmarks instead of changing it manually every time.
Review URL: http://webrtc-codereview.appspot.com/339006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1351 4adac7df-926f-26a2-2b94-8c16560cd09d
Removed old PSNR/SSIM implementations in:
* test/testsupport/metrics/video_metrics.cc
* src/modules/video_coding/codecs/test_framework/test.cc
The functions in video_metrics.cc is now using code in libyuv instead. Old code in test.cc is using the same functions.
The code for video_metrics.h had to be moved into a separate GYP file to avoid circular dependency error on Mac (see issue 160 for more details). The reason for this is that libyuv's unittest target depends on test_support_main.
BUG=
TEST=metrics_unittests in Debug+Release on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/333025
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1325 4adac7df-926f-26a2-2b94-8c16560cd09d
Robustness improvements to the delay estimator used in AECM and AEC. In AEC only for logging. Faster convergence.
TEST=audioproc_unittest + offline file tests.
output_data_fixed.pb updated despite unverified changes in r1112.
Review URL: http://webrtc-codereview.appspot.com/337006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1306 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays.
TEST=audioproc_unittest(float and fix), voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/335012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
The recorded frame timestamps are used to modify the output video on a frame-per-frame so it can be compared with the reference video using PSNR. This code will make it possible to use vie_auto_test for full stack comparisons with network interference and similar interesting simulations.
There's some refactoring done in vie_comparison_test.cc to make it fit to the new test.
Compiled and executed in Debug+Release on Linux, Mac and Windows.
BUG=
TEST=vie_auto_test --automated --gtest_filter=ViEVideoVerificationTest.*
Review URL: http://webrtc-codereview.appspot.com/320002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1269 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixed unit tests so they don't use ASSERT_DEATH since that doesn't work with Valgrind.
Fixed all Valgrind warnings except the one caused by CriticalSectionWrapper in system_wrappers.
Reworked all includes and GYP include paths to use full directory paths.
Removed util.h for logging, since it rendered warnings in Valgrind because of gflags. Replaced it with a verbose flag and a new function in video_quality_measurement.cc
BUG=
TEST=Passed test_support_unittests and video_codecs_test_framework_unittests on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/311001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1126 4adac7df-926f-26a2-2b94-8c16560cd09d
For estimating a delay over a long segment (e.g. a file) this can
throw off the estimate. Better not to add values to the AEC's histogram
until they're reliable.
BUG=
TEST=audiproc, audioproc_unittest
Review URL: http://webrtc-codereview.appspot.com/292004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1035 4adac7df-926f-26a2-2b94-8c16560cd09d
- Compute RMS over a packet's worth of audio to be sent in Channel, rather than the captured audio in TransmitMixer.
- We now use the entire packet rather than the last 10 ms frame.
- Restore functionality to LevelEstimator.
- Fix a bug in the splitting filter.
- Fix a number of bugs in process_test related to a poorly named
AudioFrame member.
- Update the unittest protobuf and float reference output.
- Add audioproc unittests.
- Reenable voe_extended_tests, and add a real function test.
- Use correct minimum level of 127.
TEST=audioproc_unittest, audioproc, voe_extended_test, voe_auto_test
Review URL: http://webrtc-codereview.appspot.com/279003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@950 4adac7df-926f-26a2-2b94-8c16560cd09d
Rewrote the codec test to render to file and do video comparisons.
Refactored the coded tests somewhat. I still need to figure out how to do comparison in the automated case.
Added video analysis to the test. This will make sure that the system output roughly the right thing.
Moved the video metrics library into the test_support library. Made the metrics library available in the automated tests.
Made sure no one passes in too large YUV videos into the autotest.
The standard test's output now gets captured for both the left and right windows.
Wrote a rendering device which just writes the raw frames to file, for analysis. Updated the base standard test to dump its left window output to file. We don't do anything with it yet though.
BUG=
TEST=
Review URL: http://webrtc-codereview.appspot.com/249001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@931 4adac7df-926f-26a2-2b94-8c16560cd09d
The unittest is not ideal for this, but I would have to use similar code as the implementation of the GetOutputDir in order to verify that it actually runs, so it wouldn't make much sense with a test like that.
It compiles and runs on Linux, Win and Mac. The folder gets created and is writeable from other tests.
I have tried using the GetOutputDir from another project that writes output files and it works as intended on all platforms.
Review URL: http://webrtc-codereview.appspot.com/270001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@906 4adac7df-926f-26a2-2b94-8c16560cd09d
The code has no intent to be superportable in all possible scenarios, since it will only be used by our own test code.
I reviewed more sophisticated libraries for doing similar things but came to the conclusion that they introduced more dependencies than motivated for this single purpose.
The unit test has been tested successfully executed on Linux (cmd line and Eclipse), Mac (XCode) and Windows (VS2008).
Review URL: http://webrtc-codereview.appspot.com/223002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@734 4adac7df-926f-26a2-2b94-8c16560cd09d
Switched to WebRtcSpl_SqrtFloor instead of WebRtcSpl_Sqrt in
NetEQ. The output is not bit-exact, but subjective listening
tests show no audible difference. Analysis shows that almost
all of the difference is in changed delay.
The reference file for NetEQ's unit test was updated.
Review URL: http://webrtc-codereview.appspot.com/139019
git-svn-id: http://webrtc.googlecode.com/svn/trunk@583 4adac7df-926f-26a2-2b94-8c16560cd09d
The test inputs RTP packets from an RTPdump file into NetEQ
and compares the output to the corresponding reference file.
Test files are included.
The change also includes a new method in NETEQTEST_RTPpacket
class, which reads past the initial file header in an RTPdump
file.
Finally, a few warnings are removed.
Review URL: http://webrtc-codereview.appspot.com/138012
git-svn-id: http://webrtc.googlecode.com/svn/trunk@568 4adac7df-926f-26a2-2b94-8c16560cd09d
