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
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
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
- 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
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
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
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
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
