generic-library/webrtc
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity
1,189 Commits 1 Branch 0 Tags 108 MiB
7752d11056
Commit Graph

13 Commits

Author SHA1 Message Date
andrew@webrtc.org
a919d3a643 Don't return a zero delay with insufficient data. 
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
 2011-11-27 23:40:58 +00:00
andrew@webrtc.org
755b04a06e Add RMS computation for the RTP level indicator. 
- 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
 2011-11-15 16:57:56 +00:00
andrew@webrtc.org
5b5c31d8dd Update fixed point audio processing output. 
Review URL: http://webrtc-codereview.appspot.com/247008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@810 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-25 03:29:08 +00:00
bjornv@webrtc.org
4c636764b7 Updated the AEC delay logging to output values in ms. PB output updated. 
Review URL: http://webrtc-codereview.appspot.com/223003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@770 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-19 08:47:40 +00:00
bjornv@webrtc.org
a59d80db45 Updated fixed point output data file after changes in nsx. Verified bitexactness before that CL and the CLs afterwards towards the new file. 
Review URL: http://webrtc-codereview.appspot.com/213003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@745 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-14 12:16:43 +00:00
bjornv@google.com
1ba3dbecbb Adds possibility to log delay estimates in AEC. 
Review URL: http://webrtc-codereview.appspot.com/178001

git-svn-id: http://webrtc.googlecode.com/svn/trunk@674 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-10-03 08:18:10 +00:00
andrew@webrtc.org
5daeae2e5f Update fixed profile data due to AECM sqrt change (no presubmit). 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@382 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-16 17:19:02 +00:00
leozwang@google.com
325bca7ccf Add unit test output. webrtc r319, ran on Xoom, synced source code on 8/8. 
Review URL: http://webrtc-codereview.appspot.com/100005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 18:13:15 +00:00
andrew@webrtc.org
14acdbc14d Update fixed-point profile output due to r313. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@333 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-09 01:54:03 +00:00
ajm@google.com
59e41405d1 Add a fixed-point profile to the APM unit test. 
It uses fixed-point NS, AECM and adaptive digital AGC. It's selected by enabling "prefer_fixed_point" in common.gypi.
Review URL: http://webrtc-codereview.appspot.com/88009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@266 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-28 17:34:04 +00:00
ajm@google.com
a769fa51c0 Adding more output data checks to APM unittest. Blowing out the protobuf definition (changing the tags) since we're still in the formative stages. Later, this would be very bad. Leaving a Frame message in case we want frame-by-frame data, but we prefer to keep the output storage small in general so avoiding it thus far. 
Review URL: http://webrtc-codereview.appspot.com/68004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-13 21:57:58 +00:00
ajm@google.com
7c4469bf61 Revamp of audio_processing unit test to use protocol buffers. Chromium's protobuf version is synced to third_party. This isn't really needed for the unit test, but I'd like to use it soon for echo recordings, so I used this as a warm up. 
Review URL: http://webrtc-codereview.appspot.com/56002

git-svn-id: http://webrtc.googlecode.com/svn/trunk@151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-06 17:45:37 +00:00
ajm@google.com
95fa29ec96 Creating a new directory for test data files, and moving audio_processing files there. 
Review URL: http://webrtc-codereview.appspot.com/48004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-06-23 11:45:12 +00:00