| 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,
|
|
|