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20 Commits

Author SHA1 Message Date
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
henrik.lundin@webrtc.org
9f710d08e1 Switch to new sqrt in NetEQ 
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
 2011-09-12 16:44:37 +00:00
henrik.lundin@webrtc.org
35dcc23110 Adding regression test to NetEQ 
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
 2011-09-09 08:01:16 +00:00
tina.legrand@webrtc.org
af931bdb39 Update of iLBC reference files for version 1.1.1, new SQRT. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@479 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-08-29 08:27:48 +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
hellner@google.com
1b627c72b5 Tests using the rtp_rtcp test data should now be run from inside trunk/test/data/rtp_rtcp. I.e. all test files were moved to the test folder. 
Review URL: http://webrtc-codereview.appspot.com/60006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-08 17:16:47 +00:00
tlegrand@google.com
3675f9b121 Review URL: http://webrtc-codereview.appspot.com/56003 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-08 06:43:34 +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
henrika@google.com
c5758f8c51 Uploaded test files for ADM functional tests. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-06 08:34:04 +00:00
tlegrand@google.com
0adca82c35 Move iLBC test and reference files to new location. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-05 09:10:23 +00:00
henrika@google.com
2e8a1a2092 Creates new test folder for VoiceEngine test files and adds the required files. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2011-07-04 15:39:40 +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