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2017-02-23Leave only an empty top level OWNERS file.Chih-Hung Hsieh
We should not copy OWNERS files from upstream, or the owners should be registered in Gerrit Code Review. Bug: 33166666 Test: default build targets Change-Id: Ibfd47e643f03678bb65880653383adb84809169d
2016-01-08Cleaning neteq_unittest resource files.minyue
BUG=webrtc:2692 Review URL: https://codereview.webrtc.org/1563983003 Cr-Commit-Position: refs/heads/master@{#11189}
2015-12-22Adding bit exactness test for Opus decoding in NetEq.minyue
Opus has become the mostly used codec in WebRTC. There, however, is no bit exactness test for Opus decoding in NetEq. The new RTP file is generated by the following steps: 1. Encode a clean RTP file with Opus RTPencode resources/audio_coding/speech_mono_32_48kHz.pcm neteq_opus_raw.rtp 960 opus 1 2. Adding jitter to the clean RTP file RTPjitter neteq_opus_raw.rtp jitter.dat neteq_opus.rtp (Note: jitter.dat does not exist in WebRTC resources folder. Check the source code for RTPjitter to know how to define such a file.) BUG=webrtc:3987 TEST=observed Opus normal decoding and FEC decoding were used, listened to the reference output. Review URL: https://codereview.webrtc.org/1515113002 Cr-Commit-Position: refs/heads/master@{#11113}
2015-12-16Update NetEq network statistics in neteq_unittest.minyue
NetEqNetworkStatistics has been updated some time ago. A bit exactness test in neteq unittests is still using the old NetEqNetworkStatistics. New neteq4_network_stats.dat generated by running TestBitExactness with flag "genref" BUG= Review URL: https://codereview.webrtc.org/1522103002 Cr-Commit-Position: refs/heads/master@{#11052}
2015-05-04Enable GoogleWifiTrace3Mbps simulations.Stefan Holmer
BUG=3277 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50829004 Cr-Commit-Position: refs/heads/master@{#9131}
2015-02-18Offline screenshare quality test, plus loopback.sprang@webrtc.org
BUG=4171 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34109004 Cr-Commit-Position: refs/heads/master@{#8408} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8408 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15resources/audio_processing: Removed unused test filesbjornv@webrtc.org
Two files not used by any tests are removed. BUG= R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7900 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15Merge audio_processing changes.pbos@webrtc.org
R=aluebs@webrtc.org, bjornv@webrtc.org BUG= Review URL: https://webrtc-codereview.appspot.com/32769004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7893 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30Minor modifications to test::RtpFileReaderhenrik.lundin@webrtc.org
Adding original_length to the Packet struct. This is populated with the plen value from the RTP dump file. In the case of reading a pcap file, original_length will be equal to length. Also increasing the maximum packet size to 3500 bytes. This is to accomodate some test files that contain PCM16b audio encoding. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28609004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7333 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-13NetEq background noise generation off by defaulthenrik.lundin@webrtc.org
This CL turns the background noise generation in NetEq off by default. The noise generation used to kick in during long-duration packet losses, when there was no point in extrapolating the latest audio any longer. However, this sometimes produces annoying noise in situations where silence would have been preferable. With this change, a long packet-loss concealment will be faded out to zeros instead of a low noise. Reference files are updated where needed. BUG=3519 R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20109004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6882 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-11Remove unused resourcehenrik.lundin@webrtc.org
The file resources/audio_coding/neteq_universal.rtp is no longer used in any test. Removing the hash file neteq_universal.rtp.sha1. BUG=2996 R=kjellander@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17679004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-26Fix a bug preventing FilePlayer from playing encoded wav fileshenrik.lundin@webrtc.org
A bug in ACM2 prevented decoding and playout of wav files where the audio data was encoded (i.e., not just linear PCM 16 bit data). This CL fixes the issue, and adds a unit test for the FilePlayer. BUG=3386 R=henrike@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21499005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6248 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20Enabling NetEq bit-exactness test for Win x64henrik.lundin@webrtc.org
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and uploaded. Also removing the old hack to have different reference files for different version of Visual Studio. The test is now only supporting VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit output identical to the generic reference file (neteq4_universal_ref.pcm), so the specialized one (neteq4_universal_ref_win_32.pcm) could have been removed. However, since the resources sync mechanism does not include removing of old files, a client could pick up the old reference and fail. Therefore, this cl also updates neteq4_universal_ref_win_32.pcm to be identical to neteq4_universal_ref.pcm. BUG=1458 R=kjellander@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14569005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-20Add resource audio for audio processing tests.andrew@webrtc.org
This is a prerequisite of: http://review.webrtc.org/9919004/ TBR=bjornv BUG=2894 Review URL: https://webrtc-codereview.appspot.com/12219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5945 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10Add tests and modify tools for new float deinterleaved interface.andrew@webrtc.org
- Add an Initialize() overload to allow specification of format parameters. This is mainly useful for testing, but could be used in the cases where a consumer knows the format before the streams arrive. - Add a reverse_sample_rate_hz_ parameter to prepare for mismatched capture and render rates. There is no functional change as it is currently constrained to match the capture rate. - Fix a bug in the float dump: we need to use add_ rather than set_. - Add a debug dump test for both int and float interfaces. - Enable unpacking of float dumps. - Enable audioproc to read float dumps. - Move more shared functionality to test_utils.h, and generally tidy up a bit by consolidating repeated code. BUG=2894 TESTED=Verified that the output produced by the float debug dump test is correct. Processed the resulting debug dump file with audioproc and ensured that we get identical output. (This is crucial, as we need to be able to exactly reproduce online results offline.) R=aluebs@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9489004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5676 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-10Adding new data files for audio classifier unit testing on Android try botsjan.skoglund@webrtc.org
BUG= R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5675 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07This CL is to add Opus complexity knob and to test it.minyue@webrtc.org
As a by-product, a generic tool for testing and comparing the complexity of codecs is added, and new audio files are added to the resources. Three complexity tests are included 1. Default Opus complexity 2. Opus complexity knob 3. Default iSAC complexity (to compare with Opus) The complexity tests are only meant for development reasons and not to be run at bots. The .isolate file is only needed for the APK packaging and test execution on Android. TEST=passes all trybots BUG= R=kjellander@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5655 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-05adding sha1 files for audio classifier testjan.skoglund@webrtc.org
This needs to done in a separate CL since the Android APK trybots cannot handle patches into the resources directory due to the fact that they work from a Chromium checkout and applies the patch into src/third_party/webrtc. BUG= R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9389004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5643 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-24Add trace-based delivery filter to BWE test framework.stefan@webrtc.org
R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5889005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5423 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03Add baseline generation/verification to BWE test framework.solenberg@webrtc.org
Updating resource file separately, once LGTM. Generates ~628k of files for current tests, highly compressable, once/if we need that. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4639004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5204 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05Fix bad Google Storage uploads of resource files.kjellander@webrtc.org
The files in this CL seem to have hit some kind of bug during upload, causing the downloaded files to get another SHA-1 hash than the .sha1 file. This makes them become redownloaded every time runhooks execute. Re-uploading them one by one seems to have resolved this. TEST=trybots passing BUG=2294 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5086 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05Add OWNERS for resources/kjellander@webrtc.org
Make it possible for all our committers to upload resource .sha1 files in here. TEST=none BUG=2294 R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5085 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-05Add svn:ignore to avoid re-download of resourceskjellander@webrtc.org
Without this, the bots will download all resources for every build. This consumes a lot of unnecessary traffic. I tried experimenting with patterns ignoring everything except the .sha1 files but wasn't able to get it working, so this will have to do for now. git-svn-id: http://webrtc.googlecode.com/svn/trunk@5082 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04Fix invalid .sha1 files for audio_codingkjellander@webrtc.org
It seems like multiple runs of the upload_to_google_storage.py script created .sha1.sha1 files that sneaked in with https://code.google.com/p/webrtc/source/detail?r=5076 This caused the wrong files getting downloaded during sync. This affected the modules_unittests and the neteq_unittests which started failing (due to wrong version of the resource files). TEST=trybots passing BUG=2294 TBR=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3329004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5077 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04Replace old resources download script with depot_toolskjellander@webrtc.org
With help from hinoka@, we're now using a more efficient approach to download only the files that have changed from Google Storge. When uploading new resource files, use upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename which of course requires gsutil authentication setup. NOTICE: Before deploying this, svn:ignore should be removed for the resources folder, or the bots will run into problems with a non-versioned file being found in the checkout during sync (as this CL adds resources to version control). All developers will also need to be informed to wipe their local resources dir to avoid getting an error during checkout due to the already existing non-versioned resources directory. BUG=2294 TEST=locally running gclient runhooks R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2095004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d
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