Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
index 66db7502e98c8e58f237a880cc0c26eea5687f44..080165a0bdf350c87c1bf1179a81c55117a03d07 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -179,9 +179,9 @@ class RTPSender { |
// Send a DTMF tone using RFC 2833 (4733). |
int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level); |
- // Set audio packet size, used to determine when it's time to send a DTMF |
- // packet in silence (CNG). |
- int32_t SetAudioPacketSize(uint16_t packet_size_samples); |
+ // This function is deprecated. It was previously used to determine when it |
+ // was time to send a DTMF packet in silence (CNG). |
+ RTC_DEPRECATED int32_t SetAudioPacketSize(uint16_t packet_size_samples); |
// Store the audio level in d_bov for |
// header-extension-for-audio-level-indication. |