11 Commits

Author SHA1 Message Date
minyue@webrtc.org
cc92e000f3 1. adding request of ACM version in the manual mode of voe_auto_test
2. adding command line flag for automated mode of voe_auto_test to choose between ACMs

3. adding request of ACM version in voe_cmd_test

R=phoglund@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2281004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4877 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-30 08:43:50 +00:00
pbos@webrtc.org
956aa7e087 Include files from webrtc/.. paths in voice_engine/
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-21 13:52:32 +00:00
pbos@webrtc.org
9213521ea9 Remove const for plain data types in voice_engine/
BUG=1644
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1463004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-14 08:31:39 +00:00
andrew@webrtc.org
ea83c6ac9d Allow voe_cmd_test to select Opus mono (now the default).
* Opus handles stereo and mono on the same payload type, so we need a different mechanism to choose between them.
* Assorted cleanups.

BUG=webrtc:1710
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3937 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-02 15:57:36 +00:00
pwestin@webrtc.org
835dbf4516 Fix no received audio in tests.
BUG=1582, 1581
Review URL: https://webrtc-codereview.appspot.com/1281005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3763 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-04 17:24:15 +00:00
pwestin@webrtc.org
e30823911c Move the VoE tests to use external transport instead of the built in udp transport
Review URL: https://webrtc-codereview.appspot.com/1223006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22 16:12:57 +00:00
pwestin@webrtc.org
684f0577fb Revert r3667 and r3665
Review URL: https://webrtc-codereview.appspot.com/1199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
Review URL: https://webrtc-codereview.appspot.com/1029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 17:52:42 +00:00
turaj@webrtc.org
b0dff12d2b 48 kHz extension to iSAC.
Test:
-manual test with voe_cmd_test.
-manual test with RTPEncode & NetEqRTPPlay.
-manual test with simpleKenny.
-Bit-exact test of iSAC-swb and iSAC-wb with head revision of trunk. The bit-exactness is confirmed on all files generated by running webrtc/modules/audio_coding/codecs/isac/main/test/QA/runiSACLongtest.txt
Review URL: https://webrtc-codereview.appspot.com/937025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3226 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-03 17:43:52 +00:00
andrew@webrtc.org
ddcc9429e7 Check the channels in receive-side processing frames.
The number of channels must be set correctly before calling ProcessStream. This
was preventing stereo frames from being processed.

Also fix voe_cmd_test, which wasn't enabling rx NS properly.

BUG=issue713, 7375579

Review URL: https://webrtc-codereview.appspot.com/929013

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3047 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-06 18:39:40 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00