henrik.lundin@webrtc.org 
							
						 
					 
					
						
						
							
						
						35dcc23110 
					 
					
						
						
							
							Adding regression test to NetEQ  
						
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						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.  
						
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						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).  
						
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						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.  
						
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						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.  
						
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						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.  
						
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						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.  
						
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						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.  
						
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						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  
						
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						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.  
						
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						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.  
						
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						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.  
						
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						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.  
						
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						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.  
						
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						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