| 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(
|
|
|