Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
index 4808a4e7eadc984ac1a6591df1de5fab11ab9acb..7f34957da56824c8a63f3f235a24c3e451237d69 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
@@ -260,8 +260,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp { |
// Audio part. |
- // Set audio packet size, used to determine when it's time to send a DTMF |
- // packet in silence (CNG). |
+ // This function is deprecated. It was previously used to determine when it |
+ // was time to send a DTMF packet in silence (CNG). |
int32_t SetAudioPacketSize(uint16_t packet_size_samples) override; |
// Send a TelephoneEvent tone using RFC 2833 (4733). |