Chromium Code Reviews| 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 bfd8e657432b2dc6197e9aab0d7f130a47d8f232..9c2b231795294573d3c8dddba683fa1e6eae7082 100644 |
| --- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
| +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h |
| @@ -294,25 +294,42 @@ class RtpRtcp : public Module { |
| * Used by the codec module to deliver a video or audio frame for |
| * packetization. |
| * |
| - * frameType - type of frame to send |
| - * payloadType - payload type of frame to send |
| + * frame_type - type of frame to send |
| + * payload_type - payload type of frame to send |
| * timestamp - timestamp of frame to send |
| - * payloadData - payload buffer of frame to send |
| - * payloadSize - size of payload buffer to send |
| + * payload_data - payload buffer of frame to send |
| + * payload_size - size of payload buffer to send |
| * fragmentation - fragmentation offset data for fragmented frames such |
| * as layers or RED |
| * |
| * return -1 on failure else 0 |
| */ |
| virtual int32_t SendOutgoingData( |
| - FrameType frameType, |
| - int8_t payloadType, |
| - uint32_t timeStamp, |
| + FrameType frame_type, |
| + int8_t payload_type, |
| + uint32_t timestamp, |
| int64_t capture_time_ms, |
| - const uint8_t* payloadData, |
| - size_t payloadSize, |
| - const RTPFragmentationHeader* fragmentation = NULL, |
| - const RTPVideoHeader* rtpVideoHdr = NULL) = 0; |
| + const uint8_t* payload_data, |
| + size_t payload_size, |
| + const RTPFragmentationHeader* fragmentation, |
| + const RTPVideoHeader* rtp_video_header, |
| + uint32_t* frame_id_out) = 0; |
| + |
| + // Deprecated version of the methood above. |
|
stefan-webrtc
2016/07/28 08:51:17
method
Sergey Ulanov
2016/07/28 20:52:46
Done.
|
| + 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); |
| + } |
| virtual bool TimeToSendPacket(uint32_t ssrc, |
| uint16_t sequence_number, |