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author | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-10-01 08:23:21 +0000 |
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committer | henrik.lundin@webrtc.org <henrik.lundin@webrtc.org> | 2014-10-01 08:23:21 +0000 |
commit | d29f51c6ad1df3b56739b3dd9a8e520f5b7876dd (patch) | |
tree | 870e44284c7b61ced7296cf1bb93a3637a0ead1c /voice_engine/channel.cc | |
parent | 7d0ed969e493a8aedfb8eed503248e8af4623ac2 (diff) | |
download | webrtc-d29f51c6ad1df3b56739b3dd9a8e520f5b7876dd.tar.gz |
Reland "Remove DTMF status methods from Voice Engine" r7276
This reverts r7277.
TBR=henrika@webrtc.org,pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/29599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
Diffstat (limited to 'voice_engine/channel.cc')
-rw-r--r-- | voice_engine/channel.cc | 21 |
1 files changed, 0 insertions, 21 deletions
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc index 93a241be..1d23afd4 100644 --- a/voice_engine/channel.cc +++ b/voice_engine/channel.cc @@ -2627,27 +2627,6 @@ int Channel::SendTelephoneEventInband(unsigned char eventCode, } int -Channel::SetDtmfPlayoutStatus(bool enable) -{ - WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), - "Channel::SetDtmfPlayoutStatus()"); - if (audio_coding_->SetDtmfPlayoutStatus(enable) != 0) - { - _engineStatisticsPtr->SetLastError( - VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning, - "SetDtmfPlayoutStatus() failed to set Dtmf playout"); - return -1; - } - return 0; -} - -bool -Channel::DtmfPlayoutStatus() const -{ - return audio_coding_->DtmfPlayoutStatus(); -} - -int Channel::SetSendTelephoneEventPayloadType(unsigned char type) { WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), |