| 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,
|
|
|