Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
index 0c36117cea64d482a4044c89ea2237e521f4b702..9fa3959c906128a81596a7fc8f97e9a269167bbc 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
@@ -227,6 +227,7 @@ class RtpRtcp : public Module { |
// as layers or RED |
// |transport_frame_id_out| - set to RTP timestamp. |
// Returns true on success. |
+ |
virtual bool SendOutgoingData(FrameType frame_type, |
int8_t payload_type, |
uint32_t timestamp, |
@@ -237,6 +238,24 @@ class RtpRtcp : public Module { |
const RTPVideoHeader* rtp_video_header, |
uint32_t* transport_frame_id_out) = 0; |
+ // Deprecated version of the method above. |
+ int32_t SendOutgoingData( |
+ FrameType frame_type, |
+ int8_t payload_type, |
+ uint32_t timestamp, |
+ int64_t capture_time_ms, |
+ const uint8_t* payload_data, |
+ size_t payload_size, |
+ const RTPFragmentationHeader* fragmentation = nullptr, |
+ const RTPVideoHeader* rtp_video_header = nullptr) { |
+ return SendOutgoingData(frame_type, payload_type, timestamp, |
+ capture_time_ms, payload_data, payload_size, |
+ fragmentation, rtp_video_header, |
+ /*frame_id_out=*/nullptr) |
+ ? 0 |
+ : -1; |
+ } |
+ |
virtual bool TimeToSendPacket(uint32_t ssrc, |
uint16_t sequence_number, |
int64_t capture_time_ms, |