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Author SHA1 Message Date
Jelena Marusic
f09e09c7ee VoE: Remove unused interfaces 
BUG=4690

I have removed methods in VoE interfaces that were marked to be removed. I have removed them also in fake and mock implementations. I have also updated the callers in various ways:
1. Project win_test had some calls to the removed methods, but it turned out that the project is not used anymore, so I removed it entirely.
2. There were some calls to removed methods in jni methods. I have removed couple of jni methods as now they seem to do nothing.
3. With the remaining callers I just removed the calls to removed methods.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53519004

Cr-Commit-Position: refs/heads/master@{#9281}
 2015-05-26 08:25:00 +00:00
Jelena Marusic
0d266054ac VoE: apply new style guide on VoE interfaces and their implementations 
Changes:
1. Ran clang-format on VoE interfaces and their implementations.
2. Replaced virtual with override in derived classes.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49239004

Cr-Commit-Position: refs/heads/master@{#9130}
 2015-05-04 12:15:41 +00:00
Ivo Creusen
adf89b7e33 Added SetBitRate function to VoE API to allow changing the audio bitrate. 
If the requested bitrate is not valid for the codec, the codec will decide on
an appropriate value.
Updated VoE command line tool to use new SetBitRate function.
Includes unittests for SetBitRate function.

BUG=
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50789004

Cr-Commit-Position: refs/heads/master@{#9115}
 2015-04-29 14:03:45 +00:00
minyue@webrtc.org
9b2e1144df Supporting Opus DTX in Voice Engine. 
Opus DTX is an Opus specific feature. It does not require WebRTC VAD/DTX, therefore is not set by VoECodec::SetVADStatus(), but rather a dedicated API.

BUG=1014
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43709004

Cr-Commit-Position: refs/heads/master@{#8716}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8716 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-03-13 09:38:55 +00:00
henrik.lundin@webrtc.org
8315d7de85 Remove dual stream functionality in VoiceEngine 
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:07:26 +00:00
minyue@webrtc.org
adee8f9242 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate 
This is to maintain the consistency with the Opus codec option "maxplaybackrate" defined in http://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

BUG=
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 12:28:06 +00:00
minyue@webrtc.org
6aac93bd9c Adding SetOpusMaxBandwidth in VoE and ACM 
This is a step to solve
https://code.google.com/p/webrtc/issues/detail?id=1906

In particular, we add an API in VoE and ACM to call Opus's API of setting maximum bandwidth.

TEST = added a test in voe_cmd_test and listened to the result

BUG=
R=henrika@google.com, henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6869 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-12 08:13:33 +00:00
minyue@webrtc.org
c1a40a7b68 This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. 
This CL is going to be combined with another CL in ACM, which is to be landed.

TEST=passed_try_bots
BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 09:52:06 +00:00
henrika@webrtc.org
0f7375504a Removes VoECodec sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=juberti@webrtc.org, niklas.enbom@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5927 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-17 10:38:08 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +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
turaj@webrtc.org
42259e7ebc VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Committed: https://code.google.com/p/webrtc/source/detail?r=3231
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3257 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 02:15:12 +00:00
perkj@webrtc.org
2cf22a6abc Revert 3231 - VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Review URL: https://webrtc-codereview.appspot.com/970005

TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/929040

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3236 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-04 10:02:02 +00:00
turaj@webrtc.org
767d87cf24 VoE Changes to enable dual_streaming. 
TEST=added new unit-test

This CL depends on issue 933015 http://webrtc-codereview.appspot.com/933015/
which is under review. Should be committed after issue 933015 is committed.
Review URL: https://webrtc-codereview.appspot.com/970005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 22:51:37 +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