| 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..9ff7406b4f6120c7cafc1c67eb2325e74b57d1c9 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 TransportSequenceNumberAllocator {
|
| + public:
|
| + TransportSequenceNumberAllocator() {}
|
| + virtual ~TransportSequenceNumberAllocator() {}
|
| +
|
| + virtual uint16_t AllocateSequenceNumber() = 0;
|
| +};
|
| +
|
| } // namespace webrtc
|
| #endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
|
|
|