Chromium Code Reviews| Index: talk/media/base/mediachannel.h |
| diff --git a/talk/media/base/mediachannel.h b/talk/media/base/mediachannel.h |
| index 9c9e8c4b294f6d060e8c35a3d38b79c456c52fa3..64cc80fd58541a2e4f8daf6cde6d6eb77acd244c 100644 |
| --- a/talk/media/base/mediachannel.h |
| +++ b/talk/media/base/mediachannel.h |
| @@ -549,6 +549,7 @@ class MediaChannel : public sigslot::has_slots<> { |
| // Called when a RTCP packet is received. |
| virtual void OnRtcpReceived(rtc::Buffer* packet, |
| const rtc::PacketTime& packet_time) = 0; |
| + virtual void OnPacketSent(const rtc::SentPacket& packet_sent) = 0; |
|
pthatcher1
2015/09/25 23:24:57
Can you just make this default to {}, so we don't
stefan-webrtc
2015/09/28 12:10:49
Great suggestion. :)
|
| // Called when the socket's ability to send has changed. |
| virtual void OnReadyToSend(bool ready) = 0; |
| // Creates a new outgoing media stream with SSRCs and CNAME as described |