| 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_;
|
|
|