Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
index e8fc545ec52e415784a7c3fedb16ecca5679de92..fd44a597261593be17cf99f32a8f28a41be569e4 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h |
@@ -112,15 +112,15 @@ class ModuleRtpRtcpImpl : public RtpRtcp { |
// Used by the codec module to deliver a video or audio frame for |
// packetization. |
- int32_t SendOutgoingData( |
- FrameType frame_type, |
- int8_t payload_type, |
- uint32_t time_stamp, |
- int64_t capture_time_ms, |
- const uint8_t* payload_data, |
- size_t payload_size, |
- const RTPFragmentationHeader* fragmentation = NULL, |
- const RTPVideoHeader* rtp_video_header = NULL) override; |
+ bool SendOutgoingData(FrameType frame_type, |
+ int8_t payload_type, |
+ uint32_t time_stamp, |
+ int64_t capture_time_ms, |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
+ const RTPFragmentationHeader* fragmentation, |
+ const RTPVideoHeader* rtp_video_header, |
+ uint32_t* transport_frame_id_out) override; |
bool TimeToSendPacket(uint32_t ssrc, |
uint16_t sequence_number, |