| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
|
| index 5dc82fc8fce2dfb6aeb504d537ada76088d4f8fc..63d27920380047d932141dfc1ba30c09a07e631b 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
|
| @@ -50,10 +50,6 @@ RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
|
|
|
| RTPSenderAudio::~RTPSenderAudio() {}
|
|
|
| -int RTPSenderAudio::AudioFrequency() const {
|
| - return kDtmfFrequencyHz;
|
| -}
|
| -
|
| // set audio packet size, used to determine when it's time to send a DTMF packet
|
| // in silence (CNG)
|
| int32_t RTPSenderAudio::SetAudioPacketSize(uint16_t packet_size_samples) {
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|
|