Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
index 381bc13f97b003fdfe81cf7335d18524c379a237..a3cee5e70720ba62e851858e2540276d5a03590d 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
@@ -31,7 +31,7 @@ class RTPSenderAudio : public DTMFqueue { |
uint32_t frequency, |
uint8_t channels, |
uint32_t rate, |
- RtpUtility::Payload*& payload); |
+ RtpUtility::Payload** payload); |
int32_t SendAudio(FrameType frameType, |
int8_t payloadType, |
@@ -58,7 +58,7 @@ class RTPSenderAudio : public DTMFqueue { |
int32_t SetRED(int8_t payloadType); |
// Get payload type for Redundant Audio Data RFC 2198 |
- int32_t RED(int8_t& payloadType) const; |
+ int32_t RED(int8_t* payloadType) const; |
protected: |
int32_t SendTelephoneEventPacket( |