Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h |
| diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h |
| index 73fb96cc02a51c893a18a0c60715d74cc0816dd4..031af9558cece8cad8f3ac7d4dc515edbea6b3d5 100644 |
| --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h |
| +++ b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h |
| @@ -395,5 +395,36 @@ struct RtpPacketLossStats { |
| uint64_t multiple_packet_loss_packet_count; |
| }; |
| +class RtpPacketSender { |
| + public: |
| + RtpPacketSender() {} |
| + virtual ~RtpPacketSender() {} |
| + |
| + enum Priority { |
| + kHighPriority = 0, // Pass through; will be sent immediately. |
| + kNormalPriority = 2, // Put in back of the line. |
| + kLowPriority = 3, // Put in back of the low priority line. |
| + }; |
| + // Low priority packets are mixed with the normal priority packets |
| + // while we are paused. |
| + |
| + // Returns true if we send the packet now, else it will add the packet |
| + // information to the queue and call TimeToSendPacket when it's time to send. |
| + virtual bool SendPacket(Priority priority, |
| + uint32_t ssrc, |
| + uint16_t sequence_number, |
| + int64_t capture_time_ms, |
| + size_t bytes, |
| + bool retransmission) = 0; |
| +}; |
| + |
| +class SequenceNumberAllocator { |
|
stefan-webrtc
2015/09/21 13:48:14
This allocates a transport sequence number, right?
sprang_webrtc
2015/09/21 14:02:34
Done.
|
| + public: |
| + SequenceNumberAllocator() {} |
| + virtual ~SequenceNumberAllocator() {} |
| + |
| + virtual uint16_t AllocateSequenceNumber() = 0; |
| +}; |
| + |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ |