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 6d11e8044dcbd17783ac0d166f409369c2cdf47d..f10cb75100947fd8559efbfb28a062c33ec932da 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -86,8 +86,7 @@ class RTPSenderInterface { |
class RTPSender : public RTPSenderInterface { |
public: |
- RTPSender(int32_t id, |
- bool audio, |
+ RTPSender(bool audio, |
Clock* clock, |
Transport* transport, |
RtpAudioFeedback* audio_feedback, |
@@ -388,8 +387,6 @@ class RTPSender : public RTPSenderInterface { |
rtc::scoped_ptr<BitrateAggregator> bitrates_; |
Bitrate total_bitrate_sent_; |
- int32_t id_; |
- |
const bool audio_configured_; |
rtc::scoped_ptr<RTPSenderAudio> audio_; |
rtc::scoped_ptr<RTPSenderVideo> video_; |