Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
index 339e4f0d5654333f6225da5d240b59c211bab225..ca7ba3502983032d6448ae527dad94c133899683 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc |
@@ -777,11 +777,9 @@ int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband( |
return rtp_sender_.SendTelephoneEvent(key, time_ms, level); |
} |
-// Set audio packet size, used to determine when it's time to send a DTMF |
-// packet in silence (CNG). |
int32_t ModuleRtpRtcpImpl::SetAudioPacketSize( |
const uint16_t packet_size_samples) { |
- return rtp_sender_.SetAudioPacketSize(packet_size_samples); |
+ return audio_ ? 0 : -1; |
} |
int32_t ModuleRtpRtcpImpl::SetAudioLevel( |