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 f0d23425bca70542525befb74952c8dee40ca971..85d14bd09c150f4aec4543c340e5c6d3a400c8fd 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
@@ -225,8 +225,21 @@ class RtpRtcp : public Module { |
// |payload_size| - size of payload buffer to send |
// |fragmentation| - fragmentation offset data for fragmented frames such |
// as layers or RED |
- // Returns -1 on failure else 0. |
- virtual int32_t SendOutgoingData( |
+ // |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, |
+ 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) = 0; |
+ |
+ // Deprecated version of the method above. |
+ int32_t SendOutgoingData( |
FrameType frame_type, |
int8_t payload_type, |
uint32_t timestamp, |
@@ -234,7 +247,14 @@ class RtpRtcp : public Module { |
const uint8_t* payload_data, |
size_t payload_size, |
const RTPFragmentationHeader* fragmentation = nullptr, |
- const RTPVideoHeader* rtp_video_header = nullptr) = 0; |
+ 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, |