| 12e21a0d6ce70b86ffafec10a5004ef2b1826dba |
|
19-Nov-2015 |
kwiberg <kwiberg@webrtc.org> |
Remove dead code (we no longer support SILK)
Review URL: https://codereview.webrtc.org/1461043002
Cr-Commit-Position: refs/heads/master@{#10715}
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
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| b04965ccf83c2bc6e2758abab9bea0c18551a54c |
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09-Sep-2015 |
ivoc <ivoc@webrtc.org> |
Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
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| 0d266054acece70259fc1e85026194154f41e5a0 |
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04-May-2015 |
Jelena Marusic <jmarusic@webrtc.org> |
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}
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
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| adf89b7e33cc54dab9365dddead687a46a074cf0 |
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29-Apr-2015 |
Ivo Creusen <ivoc@webrtc.org> |
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}
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
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| 9b2e1144df6e3622354caca00baf4a7462a0809c |
|
13-Mar-2015 |
minyue@webrtc.org <minyue@webrtc.org> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
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| 8315d7de8551963c53162e320835c158088fcdd6 |
|
14-Jan-2015 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
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| adee8f924224e116f041564ddde83c979880e35f |
|
03-Sep-2014 |
minyue@webrtc.org <minyue@webrtc.org> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 6aac93bd9c3da92e92b016d83c8f84c65aae65b6 |
|
12-Aug-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| c1a40a7b68a8d253b0ba32b89f3126931eeaeab3 |
|
28-May-2014 |
minyue@webrtc.org <minyue@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 0f7375504a98e43101f682143ae8f3866aec3ed3 |
|
17-Apr-2014 |
henrika@webrtc.org <henrika@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| d900e8bea84c474696bf0219aed1353ce65ffd8e |
|
03-Jul-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 956aa7e0874f2e08c335a82a2c32f400fac8b031 |
|
21-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 42259e7ebc7126f5a7036940fcab65b3f8d2af38 |
|
11-Dec-2012 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 2cf22a6abce2d38e673505a4cfd5624a3710b5cd |
|
04-Dec-2012 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 767d87cf24be9a3239e0bc26ad9f3e99604615f8 |
|
03-Dec-2012 |
turaj@webrtc.org <turaj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|
| 14b43beb7ce4440b30dcea31196de5b4a529cb6b |
|
22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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
/external/webrtc/webrtc/voice_engine/voe_codec_impl.h
|