12e21a0d6ce70b86ffafec10a5004ef2b1826dba |
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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 |
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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 |
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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 |
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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
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6aac93bd9c3da92e92b016d83c8f84c65aae65b6 |
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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
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c1a40a7b68a8d253b0ba32b89f3126931eeaeab3 |
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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
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0f7375504a98e43101f682143ae8f3866aec3ed3 |
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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
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d900e8bea84c474696bf0219aed1353ce65ffd8e |
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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
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956aa7e0874f2e08c335a82a2c32f400fac8b031 |
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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
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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
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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
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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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
|