| Index: webrtc/voice_engine/channel.cc
|
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
|
| index 8f1d0c18bda724f40c6fd6a82ccc7b6c07347c41..1c0cb340fde76a51d14f900a2822e4b265174d2b 100644
|
| --- a/webrtc/voice_engine/channel.cc
|
| +++ b/webrtc/voice_engine/channel.cc
|
| @@ -360,18 +360,6 @@ void Channel::OnPlayTelephoneEvent(uint8_t event,
|
| "Channel::OnPlayTelephoneEvent(event=%u, lengthMs=%u,"
|
| " volume=%u)",
|
| event, lengthMs, volume);
|
| -
|
| - if (!_playOutbandDtmfEvent || event > 15) {
|
| - // Ignore callback since feedback is disabled or event is not a
|
| - // Dtmf tone event.
|
| - return;
|
| - }
|
| -
|
| - assert(_outputMixerPtr != NULL);
|
| -
|
| - // Start playing out the Dtmf tone (if playout is enabled).
|
| - // Reduce length of tone with 80ms to the reduce risk of echo.
|
| - _outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume);
|
| }
|
|
|
| void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
|
| @@ -791,7 +779,6 @@ Channel::Channel(int32_t channelId,
|
| _panLeft(1.0f),
|
| _panRight(1.0f),
|
| _outputGain(1.0f),
|
| - _playOutbandDtmfEvent(false),
|
| _lastLocalTimeStamp(0),
|
| _lastPayloadType(0),
|
| _includeAudioLevelIndication(false),
|
| @@ -2215,8 +2202,6 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
|
| return -1;
|
| }
|
|
|
| - _playOutbandDtmfEvent = false;
|
| -
|
| if (_rtpRtcpModule->SendTelephoneEventOutband(
|
| event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
|
| _engineStatisticsPtr->SetLastError(
|
|
|