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 762668a4e4dbac39bee3f74a2abcdbda8571ede9..dd16fe51b402ae40cd60e9ec7c0f45de87ddb557 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h |
@@ -22,10 +22,9 @@ namespace webrtc { |
class RTPSenderAudio: public DTMFqueue |
{ |
public: |
- RTPSenderAudio(const int32_t id, |
- Clock* clock, |
- RTPSender* rtpSender, |
- RtpAudioFeedback* audio_feedback); |
+ RTPSenderAudio(Clock* clock, |
+ RTPSender* rtpSender, |
+ RtpAudioFeedback* audio_feedback); |
virtual ~RTPSenderAudio(); |
int32_t RegisterAudioPayload(const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
@@ -73,7 +72,6 @@ protected: |
const int8_t payloadType); |
private: |
- const int32_t _id; |
Clock* const _clock; |
RTPSender* const _rtpSender; |
RtpAudioFeedback* const _audioFeedback; |