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Unified Diff: webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h

Issue 1350163005: Avoid circular dependency rtp_rtcp <-> paced_sender (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years, 3 months ago
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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_
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