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) { |